2007-08-29 21:43:08 +00:00
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2007-08-29 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
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(gst_rtspsrc_dup_printf):
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Use new basesink async property to make sparse RTCP packet not wait for
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preroll.
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2007-08-27 14:44:19 +00:00
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2007-08-27 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/audiofx/Makefile.am:
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Dist the right file.
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2007-08-23 16:27:36 +00:00
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2007-08-23 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
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(gst_rtspsrc_get_float), (gst_rtspsrc_play):
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Make sure we generate and parse floating point values in the POSIX
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locale instead of the current locale.
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2007-08-22 15:01:29 +00:00
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2007-08-22 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
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(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
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(gst_rtspsrc_play):
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* gst/rtsp/gstrtspsrc.h:
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Fix method detection again.
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Keep track of when we must send a Range header.
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Use segment values for Range, Speed and Scale headers.
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Parse Speed and Scale headers to update the segment values.
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2007-08-22 08:22:50 +00:00
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2007-08-22 Stefan Kost <ensonic@users.sf.net>
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patch by: Mark Nauwelaerts <manauw@skynet.be>
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* sys/v4l2/v4l2src_calls.c:
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Handle optional v4l2 ioctls gracefully.
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gst/rtp/: Added an H263 depayloader. Fixes #369392.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
(gst_rtp_h263_depay_get_property),
(gst_rtp_h263_depay_change_state),
(gst_rtp_h263_depay_plugin_init):
* gst/rtp/gstrtph263depay.h:
Added an H263 depayloader. Fixes #369392.
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
Make the H263+ pay/depayloader support H263-1998 and H263-2000
payloads.
Also alow plain H263 on the h263p payloaders. Fixes #465040.
2007-08-20 16:52:03 +00:00
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2007-08-20 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtph263depay.c: (gst_rtp_h263_depay_base_init),
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(gst_rtp_h263_depay_class_init), (gst_rtp_h263_depay_init),
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(gst_rtp_h263_depay_finalize), (gst_rtp_h263_depay_setcaps),
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(gst_rtp_h263_depay_process), (gst_rtp_h263_depay_set_property),
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(gst_rtp_h263_depay_get_property),
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(gst_rtp_h263_depay_change_state),
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(gst_rtp_h263_depay_plugin_init):
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* gst/rtp/gstrtph263depay.h:
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Added an H263 depayloader. Fixes #369392.
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* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
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(gst_rtp_h263p_depay_process):
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* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
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(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_flush):
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Make the H263+ pay/depayloader support H263-1998 and H263-2000
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payloads.
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Also alow plain H263 on the h263p payloaders. Fixes #465040.
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2007-08-19 19:11:04 +00:00
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2007-08-19 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/audiofx/audiochebyshevfreqband.c:
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* gst/audiofx/audiochebyshevfreqlimit.c:
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Add small comparision with the windowed sinc filters in the docs.
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2007-08-19 18:47:19 +00:00
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2007-08-19 Sebastian Dröge <slomo@circular-chaos.org>
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* tests/check/elements/audiochebyshevfreqband.c: (GST_START_TEST),
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(audiochebyshevfreqband_suite):
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* tests/check/elements/audiochebyshevfreqlimit.c: (GST_START_TEST),
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(audiochebyshevfreqlimit_suite):
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Also test 32 bit float mode and the type 2 variants of the filters.
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2007-08-18 19:44:55 +00:00
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2007-08-18 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
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(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
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(gst_rtspsrc_loop):
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Refactor the udp and interleaved loop function a bit.
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gst/rtsp/gstrtspsrc.*: Protect connection activity with a new lock, avoids deadlocks when going to PAUSED. Fixes #455...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Protect connection activity with a new lock, avoids deadlocks when going
to PAUSED. Fixes #455808.
2007-08-17 17:08:11 +00:00
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize), (gst_rtspsrc_connection_send),
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(gst_rtspsrc_connection_receive), (gst_rtspsrc_sink_chain),
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(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_try_send), (gst_rtspsrc_pause):
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* gst/rtsp/gstrtspsrc.h:
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Protect connection activity with a new lock, avoids deadlocks when going
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to PAUSED. Fixes #455808.
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2007-08-17 15:30:39 +00:00
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
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Fix debug statement.
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2007-08-17 15:28:40 +00:00
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
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Fix stray %u in debug line as spotted by Saur on IRC.
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2007-08-17 14:43:33 +00:00
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2007-08-17 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/audiofx/audiochebyshevfreqband.c:
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(gst_audio_chebyshev_freq_band_class_init):
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* gst/audiofx/audiochebyshevfreqlimit.c:
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(gst_audio_chebyshev_freq_limit_class_init):
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Use generator macros for the process functions for the different
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sample types, add lower upper boundaries for the GObject properties
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so automatically generated UIs can use sliders and add a note about
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the number of poles as a too high number of poles combined with
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very low or very high frequencies will produce only noise.
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* docs/plugins/gst-plugins-good-plugins.args:
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Regenerated for the property changes.
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gst/rtsp/gstrtspsrc.*: Improve timeout handling.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Improve timeout handling.
Use the same socket for sending and receiving RTCP packets so that some
servers can track clients better.
Improve connection closed handling. Try to reconnect.
Don't overwrite our content base with NULL.
Improve debugging.
Improve range parsing and handling.
Remove flushing hack now that core does the right thing.
2007-08-17 14:15:19 +00:00
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_property),
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(gst_rtspsrc_flush), (gst_rtspsrc_sink_chain),
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(gst_rtspsrc_stream_configure_udp_sink),
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(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_interleaved),
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(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
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(gst_rtspsrc_try_send), (gst_rtspsrc_send),
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(gst_rtspsrc_parse_methods), (gst_rtspsrc_parse_range),
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(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_pause),
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Improve timeout handling.
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Use the same socket for sending and receiving RTCP packets so that some
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servers can track clients better.
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Improve connection closed handling. Try to reconnect.
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Don't overwrite our content base with NULL.
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Improve debugging.
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Improve range parsing and handling.
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Remove flushing hack now that core does the right thing.
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2007-08-17 13:59:15 +00:00
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2007-08-17 Wim Taymans <wim.taymans@gmail.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
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(gst_multiudpsink_init), (gst_multiudpsink_set_property),
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(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
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(gst_multiudpsink_close), (gst_multiudpsink_add):
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* gst/udp/gstmultiudpsink.h:
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Add support for getting and setting the socket to use.
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
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(gst_udpsrc_create), (gst_udpsrc_get_property):
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Add support for getting the currently used socket.
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gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_mode_get_type),
(gst_audio_chebyshev_freq_band_base_init),
(gst_audio_chebyshev_freq_band_dispose),
(gst_audio_chebyshev_freq_band_class_init),
(gst_audio_chebyshev_freq_band_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_band_set_property),
(gst_audio_chebyshev_freq_band_get_property),
(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
(gst_audio_chebyshev_freq_band_start):
* gst/audiofx/audiochebyshevfreqband.h:
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_mode_get_type),
(gst_audio_chebyshev_freq_limit_base_init),
(gst_audio_chebyshev_freq_limit_dispose),
(gst_audio_chebyshev_freq_limit_class_init),
(gst_audio_chebyshev_freq_limit_init),
(generate_biquad_coefficients), (calculate_gain),
(generate_coefficients),
(gst_audio_chebyshev_freq_limit_set_property),
(gst_audio_chebyshev_freq_limit_get_property),
(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
(gst_audio_chebyshev_freq_limit_start):
* gst/audiofx/audiochebyshevfreqlimit.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
Fixes #464800.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/audiochebyshevfreqband.c:
(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
* tests/check/elements/audiochebyshevfreqlimit.c:
(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
Add unit tests for the chebyshev filters.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-1394.xml:
* docs/plugins/inspect/plugin-audiofx.xml:
* docs/plugins/inspect/plugin-dv.xml:
* docs/plugins/inspect/plugin-flac.xml:
* docs/plugins/inspect/plugin-jpeg.xml:
* docs/plugins/inspect/plugin-png.xml:
* docs/plugins/inspect/plugin-rtp.xml:
* docs/plugins/inspect/plugin-shout2send.xml:
* docs/plugins/inspect/plugin-wavpack.xml:
And add docs for the chebyshev filters. While doing
that also run make update in docs/plugins.
2007-08-16 17:02:07 +00:00
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2007-08-16 Sebastian Dröge <slomo@circular-chaos.org>
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reviewed by: Stefan Kost <ensonic@users.sf.net>
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* gst/audiofx/Makefile.am:
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* gst/audiofx/audiochebyshevfreqband.c:
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(gst_audio_chebyshev_freq_band_mode_get_type),
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(gst_audio_chebyshev_freq_band_base_init),
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(gst_audio_chebyshev_freq_band_dispose),
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(gst_audio_chebyshev_freq_band_class_init),
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(gst_audio_chebyshev_freq_band_init),
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(generate_biquad_coefficients), (calculate_gain),
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(generate_coefficients),
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(gst_audio_chebyshev_freq_band_set_property),
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(gst_audio_chebyshev_freq_band_get_property),
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(gst_audio_chebyshev_freq_band_setup), (process), (process_64),
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(process_32), (gst_audio_chebyshev_freq_band_transform_ip),
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(gst_audio_chebyshev_freq_band_start):
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* gst/audiofx/audiochebyshevfreqband.h:
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* gst/audiofx/audiochebyshevfreqlimit.c:
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(gst_audio_chebyshev_freq_limit_mode_get_type),
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(gst_audio_chebyshev_freq_limit_base_init),
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(gst_audio_chebyshev_freq_limit_dispose),
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(gst_audio_chebyshev_freq_limit_class_init),
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(gst_audio_chebyshev_freq_limit_init),
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(generate_biquad_coefficients), (calculate_gain),
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(generate_coefficients),
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(gst_audio_chebyshev_freq_limit_set_property),
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(gst_audio_chebyshev_freq_limit_get_property),
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(gst_audio_chebyshev_freq_limit_setup), (process), (process_64),
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(process_32), (gst_audio_chebyshev_freq_limit_transform_ip),
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(gst_audio_chebyshev_freq_limit_start):
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* gst/audiofx/audiochebyshevfreqlimit.h:
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* gst/audiofx/audiofx.c: (plugin_init):
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Add Chebyshev lowpass/highpass and bandpass/bandreject elements.
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Fixes #464800.
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* tests/check/Makefile.am:
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* tests/check/elements/.cvsignore:
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* tests/check/elements/audiochebyshevfreqband.c:
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(setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband),
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(GST_START_TEST), (audiochebyshevfreqband_suite), (main):
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* tests/check/elements/audiochebyshevfreqlimit.c:
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(setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit),
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(GST_START_TEST), (audiochebyshevfreqlimit_suite), (main):
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Add unit tests for the chebyshev filters.
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/inspect/plugin-1394.xml:
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* docs/plugins/inspect/plugin-audiofx.xml:
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* docs/plugins/inspect/plugin-dv.xml:
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* docs/plugins/inspect/plugin-flac.xml:
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* docs/plugins/inspect/plugin-jpeg.xml:
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* docs/plugins/inspect/plugin-png.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-shout2send.xml:
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* docs/plugins/inspect/plugin-wavpack.xml:
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And add docs for the chebyshev filters. While doing
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that also run make update in docs/plugins.
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2007-08-16 12:15:06 +00:00
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2007-08-16 Stefan Kost <ensonic@users.sf.net>
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* ext/annodex/gstcmmltag.c:
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* gst/rtp/gstrtpvorbispay.c:
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Make ro memory to share.
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2007-08-16 11:49:01 +00:00
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2007-08-16 Wim Taymans <wim.taymans@gmail.com>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
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Improve UDP performance by avoiding a select() when we have data
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available immediatly.
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gst/rtsp/gstrtpdec.*: Add (dummy) SSRC management signals.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
(gst_rtp_dec_class_init):
* gst/rtsp/gstrtpdec.h:
Add (dummy) SSRC management signals.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
(on_timeout), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add connection-speed property.
Add find_stream helper functions.
Handle stream EOS based on BYE messages or SSRC timeout.
Returns SUCCESS from the state change function as we hide our async
elements from the parent.
2007-08-16 11:47:19 +00:00
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2007-08-16 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_VOID__UINT_UINT),
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(gst_rtp_dec_class_init):
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* gst/rtsp/gstrtpdec.h:
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Add (dummy) SSRC management signals.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
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(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
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(find_stream), (gst_rtspsrc_create_stream), (new_session_pad),
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(request_pt_map), (gst_rtspsrc_do_stream_eos), (on_bye_ssrc),
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(on_timeout), (gst_rtspsrc_stream_configure_manager),
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(gst_rtspsrc_stream_push_event), (gst_rtspsrc_push_event),
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Add connection-speed property.
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Add find_stream helper functions.
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Handle stream EOS based on BYE messages or SSRC timeout.
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Returns SUCCESS from the state change function as we hide our async
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elements from the parent.
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2007-08-16 07:40:48 +00:00
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2007-08-16 Stefan Kost <ensonic@users.sf.net>
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* gst/debug/rndbuffersize.c:
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Fix da leak.
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2007-08-14 13:50:43 +00:00
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2007-08-14 Stefan Kost <ensonic@users.sf.net>
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* gst/debug/Makefile.am:
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* gst/debug/breakmydata.c:
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* gst/debug/gstdebug.c:
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* gst/debug/negotiation.c:
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* gst/debug/progressreport.c:
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* gst/debug/rndbuffersize.c:
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* gst/debug/testplugin.c:
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Add new test element and clean-up the others a little.
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2007-08-12 14:35:41 +00:00
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2007-08-12 Wim Taymans <wim.taymans@gmail.com>
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* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
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Fix parsing of mp4a version 0 atoms. Fixes #465774.
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2007-08-10 17:08:01 +00:00
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2007-08-10 Stefan Kost <ensonic@users.sf.net>
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* gst/rtp/gstrtpilbcdepay.c:
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Include stdlib.
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2007-08-10 16:10:47 +00:00
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2007-08-10 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtp/gstrtpmpvdepay.c:
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Set the mpegversion in the caps so that autoplugging does not get
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confused.
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2007-08-09 10:54:05 +00:00
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2007-08-09 Thomas Vander Stichele <thomas at apestaart dot org>
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* po/hu.po:
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* po/uk.po:
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* po/vi.po:
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Updated translations.
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2007-08-08 17:47:05 +00:00
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2007-08-08 Michael Smith <msmith@fluendo.com>
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* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
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Render right border in the correct location.
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2007-08-08 10:54:50 +00:00
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2007-08-08 Wim Taymans <wim.taymans@gmail.com>
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Patch by: Olivier Crete <tester at tester dot ca>
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* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
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* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
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Make mode property a string. Fixes #464475.
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2007-08-05 14:58:20 +00:00
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2007-08-05 Stefan Kost <ensonic@users.sf.net>
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* ext/flac/gstflacenc.c:
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Widen caps to match decoder a bit and add more FIXMEs.
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2007-08-05 14:53:36 +00:00
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2007-08-05 Stefan Kost <ensonic@users.sf.net>
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patch by: Mark Nauwelaerts <manauw@skynet.be>
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* gst/avi/gstavimux.c:
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Fix ODML index tag numbering. Fixes #463624.
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2007-08-03 16:08:56 +00:00
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2007-08-03 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (get_default_rate_for_pt),
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(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
|
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(gst_rtspsrc_stream_configure_tcp),
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(gst_rtspsrc_stream_configure_udp_sink):
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Fix default clock-rate for realmedia.
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Fix parsing of transport.
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Don't try to link NULL pads.
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2007-07-30 17:17:04 +00:00
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2007-07-30 Tim-Philipp Müller <tim at centricular dot net>
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* po/POTFILES.skip:
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Add POTFILES.skip with list of source files that aren't disted at the
|
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moment but contain translatable strings. Should hopefully pacify
|
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broken tools and make it clearer that these files are left out
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intentionally (#461600).
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2007-07-30 12:41:58 +00:00
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2007-07-30 Edward Hervey <bilboed@bilboed.com>
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* gst/qtdemux/qtdemux.c: (gst_qtdemux_loop_state_movie):
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If the buffer was entirely clipped ... don't try sending it :)
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2007-07-27 16:56:45 +00:00
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2007-07-27 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams),
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_methods),
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(gst_rtspsrc_create_transports_string),
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(gst_rtspsrc_prepare_transports):
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If we don't hav a session manager, set the caps on outgoing buffers
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ourselves.
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Force PAUSE/PLAY methods for now until the extensions can overwrite.
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Append final bit of the transport string even when it does not contain a
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placeholder.
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2007-07-27 11:21:20 +00:00
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2007-07-27 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_free),
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(gst_rtsp_ext_list_connect):
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* gst/rtsp/gstrtspext.h:
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize), (gst_rtspsrc_send_cb):
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Clean up the interface list.
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Allow connecting to interface signals for the extensions.
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Remove old extension code.
|
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Free list on cleanup.
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Allow extensions to send additional RTSP messages.
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2007-07-27 10:38:34 +00:00
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2007-07-27 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
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Handle a NULL gconf key gracefully by rendering the default element.
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2007-07-27 10:11:18 +00:00
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2007-07-27 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/gstrtspext.h:
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Fix include path for extension interface.
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2007-07-26 19:45:30 +00:00
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2007-07-26 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/audiofx/audioamplify.h:
|
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|
Also remove a now unecessary variable here.
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2007-07-26 19:41:07 +00:00
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|
2007-07-26 Sebastian Dröge <slomo@circular-chaos.org>
|
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* gst/audiofx/audioamplify.c: (gst_audio_amplify_init),
|
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(gst_audio_amplify_setup), (gst_audio_amplify_transform_ip):
|
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|
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* gst/audiofx/audiodynamic.c:
|
|
|
|
(gst_audio_dynamic_set_process_function), (gst_audio_dynamic_init),
|
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|
|
(gst_audio_dynamic_setup), (gst_audio_dynamic_transform_ip):
|
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|
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* gst/audiofx/audiodynamic.h:
|
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|
* gst/audiofx/audioinvert.c: (gst_audio_invert_init),
|
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|
|
(gst_audio_invert_setup), (gst_audio_invert_transform_ip):
|
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|
|
* gst/audiofx/audioinvert.h:
|
|
|
|
Don't save format information ourselves, this is already saved in
|
|
|
|
GstAudioFilter.
|
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|
|
2007-07-26 15:48:47 +00:00
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|
2007-07-26 Wim Taymans <wim.taymans@gmail.com>
|
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* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
|
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(gst_rtsp_ext_list_stream_select):
|
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|
* gst/rtsp/gstrtspext.h:
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
|
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|
Use rank to filter out extensions.
|
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Add url to stream_select interface call.
|
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gst/rtsp/: Use shiny new RTSP and SDP library.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/base64.c:
* gst/rtsp/base64.h:
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
(gst_rtsp_ext_list_setup_media),
(gst_rtsp_ext_list_configure_stream),
(gst_rtsp_ext_list_get_transports),
(gst_rtsp_ext_list_stream_select):
* gst/rtsp/gstrtspext.h:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspconnection.c:
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c:
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtspmessage.c:
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsprange.c:
* gst/rtsp/rtsprange.h:
* gst/rtsp/rtsptransport.c:
* gst/rtsp/rtsptransport.h:
* gst/rtsp/rtspurl.c:
* gst/rtsp/rtspurl.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.c:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/test.c:
Use shiny new RTSP and SDP library.
Implement RTSP extensions using the new interface.
Remove a lot of old code.
2007-07-25 18:50:08 +00:00
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2007-07-25 Wim Taymans <wim.taymans@gmail.com>
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* gst/rtsp/Makefile.am:
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* gst/rtsp/base64.c:
|
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* gst/rtsp/base64.h:
|
|
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|
* gst/rtsp/gstrtspext.c: (gst_rtsp_ext_list_filter),
|
|
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(gst_rtsp_ext_list_init), (gst_rtsp_ext_list_get),
|
|
|
|
(gst_rtsp_ext_list_detect_server), (gst_rtsp_ext_list_before_send),
|
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(gst_rtsp_ext_list_after_send), (gst_rtsp_ext_list_parse_sdp),
|
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(gst_rtsp_ext_list_setup_media),
|
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(gst_rtsp_ext_list_configure_stream),
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(gst_rtsp_ext_list_get_transports),
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(gst_rtsp_ext_list_stream_select):
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|
* gst/rtsp/gstrtspext.h:
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
|
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|
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
|
|
|
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(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
|
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(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_flush), (gst_rtspsrc_do_seek),
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(gst_rtspsrc_sink_chain), (gst_rtspsrc_stream_configure_manager),
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(gst_rtspsrc_stream_configure_tcp),
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(gst_rtspsrc_stream_configure_mcast),
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(gst_rtspsrc_stream_configure_udp),
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(gst_rtspsrc_stream_configure_udp_sink),
|
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|
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(gst_rtspsrc_stream_configure_transport),
|
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(gst_rtspsrc_handle_request), (gst_rtspsrc_send_keep_alive),
|
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
|
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|
(gst_rtspsrc_loop_send_cmd), (gst_rtsp_auth_method_to_string),
|
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(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
|
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|
|
(gst_rtspsrc_try_send), (gst_rtspsrc_send),
|
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(gst_rtspsrc_parse_methods),
|
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(gst_rtspsrc_create_transports_string),
|
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(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
|
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(gst_rtspsrc_parse_range), (gst_rtspsrc_open), (gst_rtspsrc_close),
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|
|
(gst_rtspsrc_play), (gst_rtspsrc_pause),
|
|
|
|
(gst_rtspsrc_change_state), (gst_rtspsrc_uri_set_uri):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
* gst/rtsp/rtsp.h:
|
|
|
|
* gst/rtsp/rtspconnection.c:
|
|
|
|
* gst/rtsp/rtspconnection.h:
|
|
|
|
* gst/rtsp/rtspdefs.c:
|
|
|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
* gst/rtsp/rtspext.h:
|
|
|
|
* gst/rtsp/rtspextwms.c:
|
|
|
|
* gst/rtsp/rtspextwms.h:
|
|
|
|
* gst/rtsp/rtspmessage.c:
|
|
|
|
* gst/rtsp/rtspmessage.h:
|
|
|
|
* gst/rtsp/rtsprange.c:
|
|
|
|
* gst/rtsp/rtsprange.h:
|
|
|
|
* gst/rtsp/rtsptransport.c:
|
|
|
|
* gst/rtsp/rtsptransport.h:
|
|
|
|
* gst/rtsp/rtspurl.c:
|
|
|
|
* gst/rtsp/rtspurl.h:
|
|
|
|
* gst/rtsp/sdp.h:
|
|
|
|
* gst/rtsp/sdpmessage.c:
|
|
|
|
* gst/rtsp/sdpmessage.h:
|
|
|
|
* gst/rtsp/test.c:
|
|
|
|
Use shiny new RTSP and SDP library.
|
|
|
|
Implement RTSP extensions using the new interface.
|
|
|
|
Remove a lot of old code.
|
|
|
|
|
2007-07-24 14:31:56 +00:00
|
|
|
2007-07-24 Edward Hervey <bilboed@bilboed.com>
|
|
|
|
|
|
|
|
* gst/qtdemux/qtdemux.c: (qtdemux_video_caps):
|
|
|
|
Add codec mapping for '2vuy' (Raw YUV produced by FCP) and 'divx'.
|
|
|
|
|
2007-07-24 05:07:59 +00:00
|
|
|
2007-07-24 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
|
|
|
|
Don't unref the outgoing buffer twice when dropping it because it's
|
|
|
|
outside of the segment.
|
|
|
|
|
2007-07-24 04:57:20 +00:00
|
|
|
2007-07-24 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
|
|
|
|
(gst_wavpack_dec_chain), (gst_wavpack_dec_sink_event):
|
|
|
|
Use the new buffer clipping function from gstaudio here and
|
|
|
|
require gst-plugins-base CVS.
|
|
|
|
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
|
|
|
|
For framed Wavpack buffers we require a valid timestamp.
|
|
|
|
|
2007-07-23 18:03:54 +00:00
|
|
|
2007-07-23 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
|
|
|
|
* gst/qtdemux/qtdemux.c: (gst_qtdemux_activate_segment),
|
|
|
|
(gst_qtdemux_clip_buffer), (gst_qtdemux_loop_state_movie),
|
|
|
|
(qtdemux_parse_trak), (qtdemux_video_caps), (qtdemux_audio_caps):
|
|
|
|
Clip raw audio and video when we can, keep track of current output
|
|
|
|
segment.
|
|
|
|
Don't leak buffers and events when there is no output pad.
|
|
|
|
Improve debugging here and there.
|
|
|
|
|
2007-07-23 09:02:07 +00:00
|
|
|
2007-07-23 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Sync liboil check with plugins-base.
|
|
|
|
|
2007-07-20 07:41:58 +00:00
|
|
|
2007-07-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* ext/annodex/Makefile.am:
|
|
|
|
Fix CFLAGS/LIBS.
|
|
|
|
|
|
|
|
* ext/cdio/gstcdiocddasrc.c:
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_task):
|
|
|
|
Include stdlib
|
|
|
|
|
|
|
|
* ext/cairo/Makefile.am:
|
|
|
|
* gst/videofilter/Makefile.am:
|
|
|
|
* tests/examples/level/Makefile.am:
|
|
|
|
Use $(LIBM) instead of -lm
|
|
|
|
|
2007-07-18 11:55:13 +00:00
|
|
|
2007-07-18 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* sys/v4l2/gstv4l2src.c:
|
|
|
|
Add another example pipeline.
|
|
|
|
|
2007-07-18 11:42:33 +00:00
|
|
|
2007-07-18 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
Patch by: Alexander Eichner <alexeichi@yahoo.de>
|
|
|
|
|
|
|
|
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
|
|
|
|
Use define here.
|
|
|
|
|
|
|
|
* sys/v4l2/gstv4l2tuner.c:
|
|
|
|
(gst_v4l2_tuner_set_frequency_and_notify):
|
|
|
|
Don't touch the property - its still disabled.
|
|
|
|
|
|
|
|
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_probe_caps_for_format),
|
|
|
|
(gst_v4l2src_grab_frame), (gst_v4l2src_get_size_limits):
|
|
|
|
* sys/v4l2/v4l2src_calls.h:
|
|
|
|
Improve fallback format negotionation. Fixes #451388
|
|
|
|
|
2007-07-18 10:33:39 +00:00
|
|
|
2007-07-18 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* tests/check/elements/videocrop.c: (GST_START_TEST):
|
|
|
|
Fix the test.
|
|
|
|
|
2007-07-18 09:21:23 +00:00
|
|
|
2007-07-18 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-jpeg.xml:
|
|
|
|
* docs/plugins/inspect/plugin-png.xml:
|
|
|
|
* ext/jpeg/gstjpegdec.c:
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_task),
|
|
|
|
(gst_pngdec_sink_setcaps):
|
|
|
|
More docs. More logs in pngdec.
|
|
|
|
|
2007-07-17 11:35:29 +00:00
|
|
|
2007-07-17 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_init):
|
|
|
|
Initialize num_buffers with minimum value.
|
|
|
|
|
|
|
|
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
|
|
|
|
(gst_v4l2src_probe_caps_for_format), (gst_v4l2src_grab_frame):
|
|
|
|
Handle frame-size query failure gracefully.
|
|
|
|
|
2007-07-16 12:11:36 +00:00
|
|
|
2007-07-16 Wim Taymans <wim.taymans@gmail.com>
|
|
|
|
|
|
|
|
* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
|
|
|
|
Fix parsing of esds atoms inside mp4a atoms so that we can set correct
|
|
|
|
codec_info for AAC audio. Fixes #457097 along with a whole other bunch
|
|
|
|
of qt/aac files.
|
|
|
|
|
2007-07-16 09:16:03 +00:00
|
|
|
2007-07-16 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* ext/wavpack/gstwavpackdec.c:
|
|
|
|
(gst_wavpack_dec_clip_outgoing_buffer):
|
|
|
|
Fix buffer clipping to correctly clip to the segment stop.
|
|
|
|
|
2007-07-13 16:31:27 +00:00
|
|
|
2007-07-13 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* tests/Makefile.am:
|
|
|
|
Remove bogus check for libcheck, since we check for
|
|
|
|
gstreamer-check and it pulls in the required info from there,
|
|
|
|
and we weren't actually _using_ the information for libcheck
|
|
|
|
ourselves anyway.
|
|
|
|
|
2007-07-12 11:21:01 +00:00
|
|
|
2007-07-12 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Use pkg-config to locate check.
|
|
|
|
|
2007-07-11 22:31:06 +00:00
|
|
|
2007-07-11 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
|
|
|
|
* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_chain):
|
|
|
|
* gst/debug/gstnavigationtest.c: (gst_navigationtest_transform):
|
|
|
|
* gst/effectv/gstaging.c: (gst_agingtv_transform):
|
|
|
|
* gst/effectv/gstdice.c: (gst_dicetv_transform):
|
|
|
|
* gst/effectv/gstedge.c: (gst_edgetv_transform):
|
|
|
|
* gst/effectv/gstquark.c: (gst_quarktv_transform):
|
|
|
|
* gst/effectv/gstrev.c: (gst_revtv_transform):
|
|
|
|
* gst/effectv/gstshagadelic.c: (gst_shagadelictv_transform):
|
|
|
|
* gst/effectv/gstvertigo.c: (gst_vertigotv_transform):
|
|
|
|
* gst/effectv/gstwarp.c: (gst_warptv_transform):
|
|
|
|
* gst/matroska/matroska-demux.c:
|
|
|
|
(gst_matroska_demux_add_wvpk_header),
|
|
|
|
(gst_matroska_demux_check_subtitle_buffer),
|
|
|
|
(gst_matroska_decode_buffer):
|
|
|
|
* gst/videofilter/gstvideoflip.c: (gst_video_flip_transform):
|
|
|
|
Fix build against core CVS.
|
|
|
|
|
2007-07-10 10:16:38 +00:00
|
|
|
2007-07-10 Edward Hervey <bilboed@gmail.com>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
|
|
|
|
Don't return GST_FLOW_ERROR when pushing an event returns FALSE. We
|
|
|
|
don't have enough granularity to convert that boolean into a
|
|
|
|
GstFlowReturn.
|
|
|
|
|
2007-07-06 15:00:47 +00:00
|
|
|
2007-07-06 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/law/alaw-decode.c: (alawdec_sink_setcaps),
|
|
|
|
(gst_alawdec_class_init), (gst_alawdec_init), (gst_alawdec_chain),
|
|
|
|
(gst_alawdec_change_state):
|
|
|
|
* gst/law/alaw-decode.h:
|
|
|
|
* gst/law/mulaw-decode.c: (mulawdec_sink_setcaps),
|
|
|
|
(gst_mulawdec_class_init), (gst_mulawdec_init),
|
|
|
|
(gst_mulawdec_chain), (gst_mulawdec_change_state):
|
|
|
|
* gst/law/mulaw-decode.h:
|
|
|
|
Fix capsnego bogosity in *law decoders.
|
|
|
|
|
2007-07-06 14:35:59 +00:00
|
|
|
2007-07-06 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init),
|
|
|
|
(gst_smokeenc_setcaps), (gst_smokeenc_chain),
|
|
|
|
(gst_smokeenc_change_state):
|
|
|
|
* ext/jpeg/gstsmokeenc.h:
|
|
|
|
Remove stupidity in get/set caps functions.
|
|
|
|
Fix some refcounting problems.
|
|
|
|
|
2007-07-06 11:42:53 +00:00
|
|
|
2007-07-06 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
|
|
|
|
Remove endianness-flipping hack that seems to have been required
|
|
|
|
only because of a bug in ffmpegcolorspace.
|
|
|
|
Partially Fixes: #451908
|
|
|
|
|
2007-07-05 08:44:11 +00:00
|
|
|
2007-07-05 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
Simplify --extra-dir as gtkdoc scans recursively.
|
|
|
|
|
2007-07-03 09:59:46 +00:00
|
|
|
2007-07-03 Wim Taymans,,, <set EMAIL_ADDRESS environment variable>
|
|
|
|
|
|
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_setcaps):
|
|
|
|
Set the encoding-name in the rtp caps to all uppercase, as required by
|
|
|
|
the caps spec.
|
|
|
|
Some small cleanups in the error paths. Fixes #453037.
|
|
|
|
|
2007-06-28 19:00:43 +00:00
|
|
|
2007-06-28 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* ext/wavpack/gstwavpackparse.c:
|
|
|
|
(gst_wavpack_parse_index_get_last_entry),
|
|
|
|
(gst_wavpack_parse_index_get_entry_from_sample),
|
|
|
|
(gst_wavpack_parse_index_append_entry), (gst_wavpack_parse_reset),
|
|
|
|
(gst_wavpack_parse_scan_to_find_sample):
|
|
|
|
* ext/wavpack/gstwavpackparse.h:
|
|
|
|
Use a GSList for the GArray that is used like a list anyway.
|
|
|
|
|
2007-06-28 13:25:05 +00:00
|
|
|
2007-06-28 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
|
|
|
|
(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_flush),
|
|
|
|
(gst_gdk_pixbuf_sink_event), (gst_gdk_pixbuf_change_state):
|
|
|
|
Add state change function where we set 0/1 as default framerate in
|
|
|
|
case our setcaps function isn't called, like it might not in a
|
|
|
|
filesrc ! gdkpixbufdec scenario. Fixes assertion triggered by
|
|
|
|
gdkpixbufdec trying to create caps with a 0/0 framerate.
|
|
|
|
Also post an error message on the bus if gst_pad_push() fails when
|
|
|
|
called from our sink event handler (+1 for flow returns for event
|
|
|
|
functions in 0.11) instead of failing silently.
|
|
|
|
|
2007-06-27 11:36:24 +00:00
|
|
|
2007-06-27 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps):
|
|
|
|
Cast stack args to the proper types. Fixes #451249.
|
|
|
|
|
2007-06-27 11:04:47 +00:00
|
|
|
2007-06-27 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
|
|
|
|
(new_session_pad), (gst_rtspsrc_setup_streams):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
For container formats we only need to activate one of the streams so
|
|
|
|
that we correctly signal no-more-pads. Fixes #451015.
|
|
|
|
|
2007-06-25 12:46:08 +00:00
|
|
|
2007-06-25 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
* docs/plugins/inspect/plugin-aasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alpha.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alphacolor.xml:
|
|
|
|
* docs/plugins/inspect/plugin-annodex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-apetag.xml:
|
|
|
|
* docs/plugins/inspect/plugin-audiofx.xml:
|
|
|
|
* docs/plugins/inspect/plugin-auparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-autodetect.xml:
|
|
|
|
* docs/plugins/inspect/plugin-avi.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cacasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cairo.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cdio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cutter.xml:
|
|
|
|
* docs/plugins/inspect/plugin-debug.xml:
|
|
|
|
* docs/plugins/inspect/plugin-efence.xml:
|
|
|
|
* docs/plugins/inspect/plugin-effectv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-esdsink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flac.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flxdec.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gconfelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
|
|
|
|
* docs/plugins/inspect/plugin-goom.xml:
|
|
|
|
* docs/plugins/inspect/plugin-halelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-icydemux.xml:
|
|
|
|
* docs/plugins/inspect/plugin-id3demux.xml:
|
|
|
|
* docs/plugins/inspect/plugin-jpeg.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ladspa.xml:
|
|
|
|
* docs/plugins/inspect/plugin-level.xml:
|
|
|
|
* docs/plugins/inspect/plugin-matroska.xml:
|
|
|
|
* docs/plugins/inspect/plugin-mulaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-multipart.xml:
|
|
|
|
* docs/plugins/inspect/plugin-navigationtest.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ossaudio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-png.xml:
|
|
|
|
* docs/plugins/inspect/plugin-quicktime.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtsp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-smpte.xml:
|
|
|
|
* docs/plugins/inspect/plugin-speex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-taglib.xml:
|
|
|
|
* docs/plugins/inspect/plugin-udp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videobalance.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videobox.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videocrop.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videoflip.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videomixer.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavenc.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ximagesrc.xml:
|
|
|
|
Update docs with caps info.
|
|
|
|
|
2007-06-25 12:13:09 +00:00
|
|
|
2007-06-25 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* po/POTFILES.in:
|
|
|
|
Add more files with translatable strings (#450878).
|
|
|
|
|
2007-06-22 14:26:36 +00:00
|
|
|
2007-06-22 Jan Schmidt <thaytan@noraisin.net>
|
|
|
|
|
|
|
|
* MAINTAINERS:
|
|
|
|
Updating all the maintainers files
|
|
|
|
|
2007-06-22 10:12:15 +00:00
|
|
|
2007-06-22 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/flac/gstflactag.c: (gst_flac_tag_init):
|
|
|
|
* gst/interleave/deinterleave.c: (deinterleave_init),
|
|
|
|
(deinterleave_sink_link):
|
|
|
|
* gst/interleave/interleave.c: (interleave_init):
|
|
|
|
* gst/median/gstmedian.c: (gst_median_init):
|
|
|
|
* gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init):
|
|
|
|
Fix memory leaks.
|
|
|
|
* tests/check/elements/id3demux.c: (pad_added_cb):
|
|
|
|
Remove unused variable.
|
|
|
|
|
2007-06-21 10:48:10 +00:00
|
|
|
2007-06-21 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/gconf/gconf.h:
|
|
|
|
Make the prototype of gst_gconf_get_key_for_sink_profile
|
|
|
|
match the implementation.
|
|
|
|
Patch by: Damien Carbery <damien dot carbery at sun dot com>
|
|
|
|
Fixes: #449747
|
|
|
|
|
2007-06-20 12:56:12 +00:00
|
|
|
2007-06-20 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpdepay.c:
|
|
|
|
Fix description - rtpdepay is not a payloader.
|
|
|
|
|
2007-06-20 08:26:21 +00:00
|
|
|
2007-06-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/qtdemux/qtdemux.c: (qtdemux_parse_samples),
|
|
|
|
(qtdemux_video_caps):
|
|
|
|
* gst/qtdemux/qtdemux_fourcc.h:
|
|
|
|
Add MJPG to the variants of motion jpeg.
|
|
|
|
|
2007-06-19 16:40:40 +00:00
|
|
|
2007-06-19 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
* tests/check/elements/audiopanorama.c: (GST_START_TEST):
|
|
|
|
* tests/check/elements/videocrop.c: (GST_START_TEST):
|
|
|
|
* tests/check/elements/videofilter.c:
|
|
|
|
* tests/check/elements/wavpackdec.c: (GST_START_TEST):
|
|
|
|
* tests/check/elements/wavpackparse.c: (GST_START_TEST):
|
|
|
|
Add GST_OPTION_CFLAGS to CFLAGS when building unit tests, so the
|
|
|
|
error flags are included and it errors out on compiler warnings
|
|
|
|
for CVS builds; remove unused variables in various unit tests.
|
|
|
|
|
2007-06-19 14:48:03 +00:00
|
|
|
2007-06-19 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
|
|
|
|
(rtsp_connection_close), (rtsp_connection_free):
|
|
|
|
Use threadsafe inet_ntop to convert an ip number to a string.
|
|
|
|
Fixes #447961.
|
|
|
|
Don't leak fd (and ip) when freeing a connection without first closing
|
|
|
|
it.
|
|
|
|
|
2007-06-19 10:41:49 +00:00
|
|
|
2007-06-19 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Back to CVS
|
|
|
|
|
|
|
|
* gst-plugins-good.doap:
|
|
|
|
Add 0.10.6 to the doap file.
|
|
|
|
|
2007-06-19 10:24:55 +00:00
|
|
|
=== release 0.10.6 ===
|
|
|
|
|
|
|
|
2007-06-18 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.10.6, "Wobble Board"
|
|
|
|
|
2007-06-17 12:35:03 +00:00
|
|
|
2007-06-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
|
|
|
|
(rtsp_connection_free):
|
|
|
|
Revert previous commit again, since we are frozen (sorry).
|
|
|
|
|
2007-06-17 12:24:58 +00:00
|
|
|
2007-06-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Peter Kjellerstedt <pkj at axis com>
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
|
|
|
|
(rtsp_connection_free):
|
|
|
|
inet_ntoa() uses a static buffer internally, so we need to copy the
|
|
|
|
returned string if we want to store it for later (#447961).
|
|
|
|
|
2007-06-15 09:13:55 +00:00
|
|
|
2007-06-15 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* win32/vs6/autogen.dsp:
|
|
|
|
* win32/vs6/gst_plugins_good.dsw:
|
|
|
|
* win32/vs6/libgstalaw.dsp:
|
|
|
|
* win32/vs6/libgstalpha.dsp:
|
|
|
|
* win32/vs6/libgstalphacolor.dsp:
|
|
|
|
* win32/vs6/libgstapetag.dsp:
|
|
|
|
* win32/vs6/libgstaudiofx.dsp:
|
|
|
|
* win32/vs6/libgstauparse.dsp:
|
|
|
|
* win32/vs6/libgstautodetect.dsp:
|
|
|
|
* win32/vs6/libgstavi.dsp:
|
|
|
|
* win32/vs6/libgstcutter.dsp:
|
|
|
|
* win32/vs6/libgstdirectdraw.dsp:
|
|
|
|
* win32/vs6/libgstdirectsound.dsp:
|
|
|
|
* win32/vs6/libgsteffectv.dsp:
|
|
|
|
* win32/vs6/libgstflx.dsp:
|
|
|
|
* win32/vs6/libgstgoom.dsp:
|
|
|
|
* win32/vs6/libgsticydemux.dsp:
|
|
|
|
* win32/vs6/libgstid3demux.dsp:
|
|
|
|
* win32/vs6/libgstinterleave.dsp:
|
|
|
|
* win32/vs6/libgstjpeg.dsp:
|
|
|
|
* win32/vs6/libgstlevel.dsp:
|
|
|
|
* win32/vs6/libgstmatroska.dsp:
|
|
|
|
* win32/vs6/libgstmedian.dsp:
|
|
|
|
* win32/vs6/libgstmonoscope.dsp:
|
|
|
|
* win32/vs6/libgstmulaw.dsp:
|
|
|
|
* win32/vs6/libgstmultipart.dsp:
|
|
|
|
* win32/vs6/libgstqtdemux.dsp:
|
|
|
|
* win32/vs6/libgstrtp.dsp:
|
|
|
|
* win32/vs6/libgstrtsp.dsp:
|
|
|
|
* win32/vs6/libgstsmpte.dsp:
|
|
|
|
* win32/vs6/libgstspeex.dsp:
|
|
|
|
* win32/vs6/libgstudp.dsp:
|
|
|
|
* win32/vs6/libgstvideobalance.dsp:
|
|
|
|
* win32/vs6/libgstvideobox.dsp:
|
|
|
|
* win32/vs6/libgstvideocrop.dsp:
|
|
|
|
* win32/vs6/libgstvideoflip.dsp:
|
|
|
|
* win32/vs6/libgstvideomixer.dsp:
|
|
|
|
* win32/vs6/libgstwaveform.dsp:
|
|
|
|
* win32/vs6/libgstwavenc.dsp:
|
|
|
|
* win32/vs6/libgstwavparse.dsp:
|
|
|
|
Mark *.dsp & *.dsw as binary files and convert to DOS line
|
|
|
|
endings, as they don't load into VS6 correctly otherwise.
|
|
|
|
|
2007-06-15 08:32:52 +00:00
|
|
|
2007-06-15 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
|
|
|
|
(rtsp_connection_connect):
|
|
|
|
Fix the MingW build.
|
|
|
|
Patch By: Vincent Torri <vtorri at univ-evry dot fr>
|
|
|
|
Fixes: #446981
|
|
|
|
|
2007-06-14 14:03:41 +00:00
|
|
|
2007-06-14 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* tests/check/elements/.cvsignore:
|
|
|
|
* tests/icles/.cvsignore:
|
|
|
|
Hush the buildbots up
|
|
|
|
|
2007-06-14 12:14:24 +00:00
|
|
|
2007-06-14 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* sys/Makefile.am:
|
|
|
|
* sys/directdraw/Makefile.am:
|
|
|
|
* sys/directsound/Makefile.am:
|
|
|
|
* sys/waveform/Makefile.am:
|
|
|
|
Make sure to dist everything needed for win32 builds.
|
|
|
|
|
2007-06-14 10:23:20 +00:00
|
|
|
2007-06-14 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
|
|
|
|
For AMR-NB streams, export the AMRSpecificBox as codec_data on the
|
|
|
|
caps.
|
|
|
|
Fixes #447458
|
|
|
|
|
2007-06-13 17:11:24 +00:00
|
|
|
2007-06-13 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
|
|
|
|
Make sure we allocate enough memory for the codec_data.
|
|
|
|
Fixes #447210.
|
|
|
|
|
2007-06-12 21:05:22 +00:00
|
|
|
2007-06-12 Sebastien Moutte <sebastien@moutte.net>
|
|
|
|
|
|
|
|
* win32/MANIFEST:
|
|
|
|
Add videocrop project file to the win32 manifest.
|
|
|
|
* win32/vs6/gst_plugins_good.dsw:
|
|
|
|
Add qtdemux,videocrop and waveform projects to the workspace.
|
|
|
|
* win32/vs6/libgstqtdemux.dsp:
|
|
|
|
Add zlib to the link list of qtdemux.
|
|
|
|
* win32/vs6/libgstvideocrop.dsp:
|
|
|
|
Add a project file for videocrop.
|
|
|
|
|
2007-06-12 20:22:26 +00:00
|
|
|
2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* po/POTFILES.in:
|
|
|
|
Add qtdemux for translation
|
|
|
|
|
2007-06-12 20:15:29 +00:00
|
|
|
2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-videocrop.xml:
|
|
|
|
* gst-plugins-good.spec.in:
|
|
|
|
* sys/Makefile.am:
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
* tests/icles/Makefile.am:
|
|
|
|
* tests/icles/videocrop-test.c:
|
|
|
|
Move videocrop and osxvideo from -bad.
|
|
|
|
|
2007-06-12 19:35:08 +00:00
|
|
|
2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
* docs/plugins/inspect/plugin-qtdemux.xml:
|
|
|
|
* docs/plugins/inspect/plugin-quicktime.xml:
|
|
|
|
* win32/MANIFEST:
|
|
|
|
Move qtdemux from -bad.
|
|
|
|
|
|
|
|
* gst-plugins-good.spec.in:
|
|
|
|
Update spec file to reflect moving of qtdemux and wavpack
|
|
|
|
|
2007-06-12 18:52:33 +00:00
|
|
|
2007-06-12 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
2007-06-12 19:01:41 +00:00
|
|
|
* win32/MANIFEST:
|
2007-06-12 18:52:33 +00:00
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-directdraw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-directsound.xml:
|
|
|
|
* docs/plugins/inspect/plugin-waveform.xml:
|
|
|
|
Move the waveform plugin from -bad too. Update the inspect xml
|
|
|
|
files to mention Plugins Good instead of Plugins Bad.
|
|
|
|
|
sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize) (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
Original commit message from CVS:
2007-06-12 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
finalization and resuscitation. No longer public.
(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
(gst_v4l2_buffer_pool_destroy): Make the pool follow common
miniobject semantics, and be threadsafe.
(gst_v4l2src_queue_frame): Remove this function, as we just call
the ioctls directly in the two places where we queue buffers.
(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
directly.
(gst_v4l2src_capture_init): Use the new buffer_pool_new function
to allocate the pool, which also preallocates the GstBuffers.
(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
queueing the frames directly.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
real MiniObject instead of rolling our own refcounting and
finalizing. Give it a lock.
(struct _GstV4l2Buffer): Remove one intermediary object, having
the buffers hold the struct v4l2_buffer directly.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
capture_init so that it can set them on the buffers that it will
create.
(gst_v4l2src_get_read): For better or for worse, include the
timestamping and offsetting code here; really we should be using
bufferalloc though.
(gst_v4l2src_get_mmap): Just make grab_frame return one of our
preallocated, mmap'd buffers.
2007-06-12 11:23:01 +00:00
|
|
|
2007-06-12 Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
|
|
|
|
(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
|
|
|
|
(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
|
|
|
|
finalization and resuscitation. No longer public.
|
|
|
|
(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
|
|
|
|
(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
|
|
|
|
(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
|
|
|
|
(gst_v4l2_buffer_pool_destroy): Make the pool follow common
|
|
|
|
miniobject semantics, and be threadsafe.
|
|
|
|
(gst_v4l2src_queue_frame): Remove this function, as we just call
|
|
|
|
the ioctls directly in the two places where we queue buffers.
|
|
|
|
(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
|
|
|
|
directly.
|
|
|
|
(gst_v4l2src_capture_init): Use the new buffer_pool_new function
|
|
|
|
to allocate the pool, which also preallocates the GstBuffers.
|
|
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|
(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
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queueing the frames directly.
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2007-06-12 13:33:56 +00:00
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(gst_v4l2src_grab_frame): Return a copy of the pool buffer if all
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mmap buffers have been dequeued.
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sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize) (gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
Original commit message from CVS:
2007-06-12 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.c (gst_v4l2_buffer_finalize)
(gst_v4l2_buffer_class_init, gst_v4l2_buffer_get_type)
(gst_v4l2_buffer_new): Behave more like ximagesink's buffers, with
finalization and resuscitation. No longer public.
(gst_v4l2_buffer_pool_finalize, gst_v4l2_buffer_pool_init)
(gst_v4l2_buffer_pool_class_init, gst_v4l2_buffer_pool_get_type)
(gst_v4l2_buffer_pool_new, gst_v4l2_buffer_pool_activate)
(gst_v4l2_buffer_pool_destroy): Make the pool follow common
miniobject semantics, and be threadsafe.
(gst_v4l2src_queue_frame): Remove this function, as we just call
the ioctls directly in the two places where we queue buffers.
(gst_v4l2src_grab_frame): Return a flowreturn and fill the buffer
directly.
(gst_v4l2src_capture_init): Use the new buffer_pool_new function
to allocate the pool, which also preallocates the GstBuffers.
(gst_v4l2src_capture_start): Call buffer_pool_activate instead of
queueing the frames directly.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
real MiniObject instead of rolling our own refcounting and
finalizing. Give it a lock.
(struct _GstV4l2Buffer): Remove one intermediary object, having
the buffers hold the struct v4l2_buffer directly.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
capture_init so that it can set them on the buffers that it will
create.
(gst_v4l2src_get_read): For better or for worse, include the
timestamping and offsetting code here; really we should be using
bufferalloc though.
(gst_v4l2src_get_mmap): Just make grab_frame return one of our
preallocated, mmap'd buffers.
2007-06-12 11:23:01 +00:00
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* sys/v4l2/gstv4l2src.h (struct _GstV4l2BufferPool): Make this a
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real MiniObject instead of rolling our own refcounting and
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finalizing. Give it a lock.
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(struct _GstV4l2Buffer): Remove one intermediary object, having
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the buffers hold the struct v4l2_buffer directly.
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* sys/v4l2/gstv4l2src.c (gst_v4l2src_set_caps): Pass the caps to
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capture_init so that it can set them on the buffers that it will
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create.
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(gst_v4l2src_get_read): For better or for worse, include the
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timestamping and offsetting code here; really we should be using
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bufferalloc though.
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(gst_v4l2src_get_mmap): Just make grab_frame return one of our
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preallocated, mmap'd buffers.
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2007-06-11 11:41:56 +00:00
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2007-06-11 Wim Taymans <wim@fluendo.com>
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Patch by: daniel fischer <dan at f3c dot com>
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_start),
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(gst_ximage_src_get_caps):
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Actually use the display_name property so that we can dump any
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available X display. Fixes #445905.
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2007-06-11 10:21:13 +00:00
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2007-06-11 Wim Taymans <wim@fluendo.com>
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Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
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* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_setcaps):
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* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_setcaps):
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Add missing rate fields to caps. Fixes #441118.
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2007-06-10 21:14:11 +00:00
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2007-06-10 Sebastien Moutte <sebastien@moutte.net>
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* win32/vs6/gst_plugins_good.dsw:
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* win32/vs8/gst-plugins-good.sln:
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Add DirectSound and DirectDraw sinks project files to
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workspace and solution files.
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Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
Original commit message from CVS:
Patch by: Josh Coalson <xflac at yahoo dot com>,
updated by Alexis Ballier <aballier at gentoo dot org>:
* configure.ac:
* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
(gst_flac_dec_setup_seekable_decoder),
(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
* ext/flac/gstflacdec.h:
* ext/flac/gstflacenc.c: (gst_flac_enc_init),
(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
(gst_flac_enc_chain), (gst_flac_enc_set_property),
(gst_flac_enc_get_property), (gst_flac_enc_change_state):
* ext/flac/gstflacenc.h:
Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
2007-06-10 10:53:26 +00:00
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2007-06-10 Sebastian Dröge <slomo@circular-chaos.org>
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Patch by: Josh Coalson <xflac at yahoo dot com>,
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updated by Alexis Ballier <aballier at gentoo dot org>:
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* configure.ac:
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* ext/flac/gstflacdec.c: (gst_flac_dec_reset_decoders),
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(gst_flac_dec_setup_seekable_decoder),
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(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_seek),
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(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
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(gst_flac_dec_read_seekable), (gst_flac_dec_read_stream):
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* ext/flac/gstflacdec.h:
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* ext/flac/gstflacenc.c: (gst_flac_enc_init),
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(gst_flac_enc_finalize), (gst_flac_enc_set_metadata),
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(gst_flac_enc_sink_setcaps), (gst_flac_enc_update_quality),
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(gst_flac_enc_seek_callback), (gst_flac_enc_write_callback),
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(gst_flac_enc_tell_callback), (gst_flac_enc_sink_event),
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(gst_flac_enc_chain), (gst_flac_enc_set_property),
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(gst_flac_enc_get_property), (gst_flac_enc_change_state):
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* ext/flac/gstflacenc.h:
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Add support for flac >= 1.1.3 which changed the API. Fixes bug #385887.
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2007-06-09 15:41:52 +00:00
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2007-06-09 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/wavpack/gstwavpackenc.c: (gst_wavpack_enc_sink_set_caps):
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Remove workaround for bug #421543. This is fixed in core 0.10.13 and
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not necessary anymore as we need at least that core version.
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2007-06-09 15:33:32 +00:00
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2007-06-09 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/wavpack/gstwavpackdec.c: (gst_wavpack_dec_reset),
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(gst_wavpack_dec_chain):
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* ext/wavpack/gstwavpackdec.h:
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* ext/wavpack/gstwavpackparse.c: (gst_wavpack_parse_reset),
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(gst_wavpack_parse_push_buffer):
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* ext/wavpack/gstwavpackparse.h:
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Improve discont handling by checking if the next Wavpack block has
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the expected, following block index.
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2007-06-08 20:20:56 +00:00
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2007-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/rtp/gstrtpmp4vpay.c (gst_rtp_mp4vpay_details):
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Fix element description.
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2007-06-08 20:19:55 +00:00
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2007-06-08 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* docs/plugins/gst-plugins-good-plugins.signals:
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* docs/plugins/inspect/plugin-autodetect.xml:
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* docs/plugins/inspect/plugin-gconfelements.xml:
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* docs/plugins/inspect/plugin-ladspa.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-wavpack.xml:
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* ext/Makefile.am:
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* tests/check/Makefile.am:
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move wavpack plugin. See #352605.
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2007-06-08 17:37:02 +00:00
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2007-06-08 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* sys/Makefile.am:
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* win32/MANIFEST:
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Add DirectDraw & DirectSound plugins to the build and docs.
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2007-06-08 10:43:26 +00:00
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2007-06-08 Tim-Philipp Müller <tim at centricular dot net>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
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* ext/libpng/gstpngdec.c: (user_read_data), (gst_pngdec_task):
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When operating in pull mode, error out correct on not-linked.
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2007-06-06 10:19:17 +00:00
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2007-06-06 Andy Wingo <wingo@pobox.com>
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* sys/v4l2/v4l2src_calls.c (gst_v4l2src_probe_caps_for_format)
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(gst_v4l2src_probe_caps_for_format_and_size): Only probe for
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format and size if the ioctls are defined; should fix compilation
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on Linux < 2.16.19.
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2007-06-06 08:53:12 +00:00
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2007-06-06 Tim-Philipp Müller <tim at centricular dot net>
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* gst/videobox/gstvideobox.c: (gst_video_box_ayuv_i420):
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Printf fixes in debug statements; use LOG level for debug statements
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that are printed for each and every frame; convert c++ comments to
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C-style comments; not much point using g_try_malloc() if we then not
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even check the return value.
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2007-06-05 16:32:19 +00:00
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2007-06-05 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Bump requirements to released versions (core and base 0.10.13).
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* gst/icydemux/gsticydemux.c: (gst_icydemux_unicodify):
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Use gst_tag_utf8_from_freeform_string() from libgsttag instead of
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own implementation.
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2007-06-05 14:17:25 +00:00
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2007-06-05 Andy Wingo <wingo@pobox.com>
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* sys/v4l2/gstv4l2src.c (gst_v4l2src_start, gst_v4l2src_stop): Add
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some useless comments.
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* sys/v4l2/v4l2src_calls.c (gst_v4l2src_capture_init): Don't queue
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frames before calling STREAMON, that might leave them in a state
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where they can't be dequeued if we go back to NULL without calling
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STREAMON, according to the docs.
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(gst_v4l2src_capture_start): Enqueue buffers here instead, right
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before we call STREAMON.
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(gst_v4l2src_capture_deinit): Remove crack to work around dequeue
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failures. (For me this code hung.) The pool refcounting is still
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crack; added a note to that effect.
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2007-06-05 09:11:41 +00:00
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2007-06-05 Wim Taymans <wim@fluendo.com>
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* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
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(gst_multipart_mux_get_mime), (gst_multipart_mux_collected):
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Add support for mapping gst structure names to the MIME type equivalent.
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Implemented for audio/x-mulaw->audio/basic. Fixes #442874.
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2007-06-03 11:21:44 +00:00
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2007-06-03 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
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(gst_wavenc_sink_setcaps), (gst_wavenc_format_samples),
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(gst_wavenc_chain), (gst_wavenc_change_state):
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* gst/wavenc/gstwavenc.h:
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Properly write wav files with width!=depth by having the depth most
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significant bytes set and all others zero. Fixes #442535.
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2007-06-01 13:52:17 +00:00
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2007-06-01 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtspconnection.c:
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Add include to make buildbot happy.
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gst/rtsp/: Improves version checking, allowing an RTSP server to reply with "505
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (add_date_header),
(rtsp_connection_send), (parse_response_status),
(parse_request_line), (parse_line), (rtsp_connection_receive):
* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspmessage.c: (key_value_foreach),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_remove_header), (rtsp_message_append_headers),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Improves version checking, allowing an RTSP server to reply with "505
RTSP Version not supported.
Adds a Date header to all messages.
Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
want to be able to send a response even if something in the request was
invalid. EINVAL is only used when passing wrong arguments to functions.
Do not handle an invalid method in parse_request_line(). Defer this to
the caller so it can respond with "405 Method Not Allowed".
Improves parsing of the timeout parameter to the Session header,
allowing whitespace after the semicolon.
Avoids a compiler warning due to variables shadowing a function argument.
2007-06-01 13:07:11 +00:00
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2007-06-01 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
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(rtsp_connection_connect), (add_date_header),
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(rtsp_connection_send), (parse_response_status),
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(parse_request_line), (parse_line), (rtsp_connection_receive):
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* gst/rtsp/rtspdefs.c: (rtsp_version_as_text):
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* gst/rtsp/rtspdefs.h:
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* gst/rtsp/rtspmessage.c: (key_value_foreach),
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(rtsp_message_init_request), (rtsp_message_init_response),
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(rtsp_message_remove_header), (rtsp_message_append_headers),
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(rtsp_message_dump):
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* gst/rtsp/rtspmessage.h:
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Improves version checking, allowing an RTSP server to reply with "505
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RTSP Version not supported.
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Adds a Date header to all messages.
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Replies with RTSP_EPARSE rather than RTSP_EINVALID in cases where we
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|
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want to be able to send a response even if something in the request was
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invalid. EINVAL is only used when passing wrong arguments to functions.
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Do not handle an invalid method in parse_request_line(). Defer this to
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the caller so it can respond with "405 Method Not Allowed".
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Improves parsing of the timeout parameter to the Session header,
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allowing whitespace after the semicolon.
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Avoids a compiler warning due to variables shadowing a function argument.
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2007-06-01 11:16:17 +00:00
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2007-06-01 Wim Taymans <wim@fluendo.com>
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Based on Patch by: Daniel Charles <dcharles at ti dot com>
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
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(gst_rtp_amr_depay_process):
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* gst/rtp/gstrtpamrdepay.h:
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_base_init),
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(gst_rtp_amr_pay_class_init), (gst_rtp_amr_pay_init),
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(gst_rtp_amr_pay_setcaps), (gst_rtp_amr_pay_handle_buffer):
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* gst/rtp/gstrtpamrpay.h:
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Add support for AMR-WB.
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Small cleanups such as using BOILERPLATE.
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2007-05-31 15:57:07 +00:00
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2007-05-31 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream):
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Fix compile warning when debug is disabled as spotted bu Saur on IRC.
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sys/v4l2/v4l2src_calls.*: Store the format list in the order that the driver gives it to us.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.h:
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
the format list in the order that the driver gives it to us.
(gst_v4l2src_probe_caps_for_format_and_size)
(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
based on the capabilities of the device.
(gst_v4l2src_grab_frame): Update for object variable renaming.
(gst_v4l2src_set_capture): Update to be strict in its parameters,
as in the set_caps below.
(gst_v4l2src_capture_init): Update for object variable renaming,
and reflow.
(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
(gst_v4l2src_capture_deinit): Update for object variable renaming.
(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
(gst_v4l2src_get_fps): Remove; these functions don't have much
meaning outside of an atomic set_caps method.
(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
known.
* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
call to update_fps; not sure about this change.
(gst_v4l2_tuner_set_norm): Work around the fact that for the
moment we don't have an update_fps_func.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
structures in the object, just store what we need. Do store the
probed caps of the device. Don't store the current frame rate.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
update_fps_function, for now. Update for new object variable
naming.
(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
new object variable naming.
(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
(gst_v4l2src_get_caps): Rework to probe the device for supported
frame sizes and frame rates.
(gst_v4l2src_set_caps): Rework to be strict in the given
parameters: if someone asks us to have a certain size and rate,
that is what we configure.
(gst_v4l2src_get_read): Update for object variable naming. Don't
leak buffers on short reads.
(gst_v4l2src_get_mmap): Update for object variable naming, and add
comments.
(gst_v4l2src_create): Update for object variable naming.
2007-05-30 14:40:53 +00:00
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2007-05-30 Andy Wingo <wingo@pobox.com>
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2007-05-30 14:57:44 +00:00
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* sys/v4l2/gstv4l2object.h:
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* sys/v4l2/gstv4l2object.c (gst_v4l2_object_new): Revert some
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unintended changes.
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sys/v4l2/v4l2src_calls.*: Store the format list in the order that the driver gives it to us.
Original commit message from CVS:
2007-05-30 Andy Wingo <wingo@pobox.com>
* sys/v4l2/v4l2src_calls.h:
* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
the format list in the order that the driver gives it to us.
(gst_v4l2src_probe_caps_for_format_and_size)
(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
based on the capabilities of the device.
(gst_v4l2src_grab_frame): Update for object variable renaming.
(gst_v4l2src_set_capture): Update to be strict in its parameters,
as in the set_caps below.
(gst_v4l2src_capture_init): Update for object variable renaming,
and reflow.
(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
(gst_v4l2src_capture_deinit): Update for object variable renaming.
(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
(gst_v4l2src_get_fps): Remove; these functions don't have much
meaning outside of an atomic set_caps method.
(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
known.
* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
call to update_fps; not sure about this change.
(gst_v4l2_tuner_set_norm): Work around the fact that for the
moment we don't have an update_fps_func.
* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
structures in the object, just store what we need. Do store the
probed caps of the device. Don't store the current frame rate.
* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
update_fps_function, for now. Update for new object variable
naming.
(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
new object variable naming.
(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
(gst_v4l2src_get_caps): Rework to probe the device for supported
frame sizes and frame rates.
(gst_v4l2src_set_caps): Rework to be strict in the given
parameters: if someone asks us to have a certain size and rate,
that is what we configure.
(gst_v4l2src_get_read): Update for object variable naming. Don't
leak buffers on short reads.
(gst_v4l2src_get_mmap): Update for object variable naming, and add
comments.
(gst_v4l2src_create): Update for object variable naming.
2007-05-30 14:40:53 +00:00
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* sys/v4l2/v4l2src_calls.h:
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* sys/v4l2/v4l2src_calls.c (gst_v4l2src_fill_format_list): Store
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the format list in the order that the driver gives it to us.
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(gst_v4l2src_probe_caps_for_format_and_size)
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(gst_v4l2src_probe_caps_for_format): New functions, fill GstCaps
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based on the capabilities of the device.
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(gst_v4l2src_grab_frame): Update for object variable renaming.
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(gst_v4l2src_set_capture): Update to be strict in its parameters,
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as in the set_caps below.
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(gst_v4l2src_capture_init): Update for object variable renaming,
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and reflow.
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(gst_v4l2src_capture_start, gst_v4l2src_capture_stop)
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(gst_v4l2src_capture_deinit): Update for object variable renaming.
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(gst_v4l2src_update_fps, gst_v4l2src_set_fps)
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(gst_v4l2src_get_fps): Remove; these functions don't have much
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meaning outside of an atomic set_caps method.
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(gst_v4l2src_buffer_new): Don't set buffer duration, it is not
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known.
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* sys/v4l2/gstv4l2tuner.c (gst_v4l2_tuner_set_channel): Remove
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call to update_fps; not sure about this change.
|
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(gst_v4l2_tuner_set_norm): Work around the fact that for the
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moment we don't have an update_fps_func.
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* sys/v4l2/gstv4l2src.h (struct _GstV4l2Src): Don't put v4l2
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structures in the object, just store what we need. Do store the
|
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probed caps of the device. Don't store the current frame rate.
|
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* sys/v4l2/gstv4l2src.c (gst_v4l2src_init): Remove the
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update_fps_function, for now. Update for new object variable
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naming.
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(gst_v4l2src_set_property, gst_v4l2src_get_property): Update for
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new object variable naming.
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(gst_v4l2src_v4l2fourcc_to_structure): Rename from ..._to_caps.
|
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(gst_v4l2_structure_to_v4l2fourcc): Rename from ...caps_to_....
|
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(gst_v4l2src_get_caps): Rework to probe the device for supported
|
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frame sizes and frame rates.
|
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(gst_v4l2src_set_caps): Rework to be strict in the given
|
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parameters: if someone asks us to have a certain size and rate,
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that is what we configure.
|
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(gst_v4l2src_get_read): Update for object variable naming. Don't
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leak buffers on short reads.
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(gst_v4l2src_get_mmap): Update for object variable naming, and add
|
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comments.
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(gst_v4l2src_create): Update for object variable naming.
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2007-05-30 14:38:59 +00:00
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2007-05-30 Tim-Philipp Müller <tim at centricular dot net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
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(gst_avi_demux_reset), (gst_avi_demux_parse_stream):
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* gst/avi/gstavidemux.h:
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Parse subtitle text streams instead of erroring out (#442034). Still
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needs a parser for the subtitles to actually show up.
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2007-05-30 12:46:32 +00:00
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2007-05-30 Tim-Philipp Müller <tim at centricular dot net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_push_event),
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(gst_avi_demux_loop):
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Make _push_event() return TRUE if the event could be pushed on at
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least one pad and not only if it could be pushed on all pads,
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otherwise we'll end up posting an error message on EOS if one or
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more source pads are not connected.
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2007-05-28 16:39:09 +00:00
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2007-05-28 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtsptransport.c:
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Use renamed RTP bin.
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gst/videobox/gstvideobox.c: Add AYUV->AYUV and AYUV->I420 formats.
Original commit message from CVS:
Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
(gst_video_box_set_property), (gst_video_box_transform_caps),
(video_box_recalc_transform), (gst_video_box_set_caps),
(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
(gst_video_box_i420_i420), (gst_video_box_transform),
(plugin_init):
Add AYUV->AYUV and AYUV->I420 formats.
Fix negotiation and I420->AYUV conversion.
Fixes #429329.
2007-05-28 15:01:33 +00:00
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2007-05-28 Wim Taymans <wim@fluendo.com>
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Based on patch by: Dejan Sakelšak <sakdean at gmail dot com>
|
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
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(gst_video_box_set_property), (gst_video_box_transform_caps),
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(video_box_recalc_transform), (gst_video_box_set_caps),
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(gst_video_box_get_unit_size), (gst_video_box_apply_alpha),
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(gst_video_box_ayuv_ayuv), (gst_video_box_clear), (UVfloor),
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(UVceil), (gst_video_box_ayuv_i420), (gst_video_box_i420_ayuv),
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(gst_video_box_i420_i420), (gst_video_box_transform),
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(plugin_init):
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Add AYUV->AYUV and AYUV->I420 formats.
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Fix negotiation and I420->AYUV conversion.
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Fixes #429329.
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2007-05-26 15:25:18 +00:00
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2007-05-26 Wim Taymans <wim@fluendo.com>
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* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_data):
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Use different variables for nested for loops so that the outer loop
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functions properly and speex files with multiple frames per buffer work
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properly.
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Fixes #441408.
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2007-05-25 20:51:36 +00:00
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2007-05-25 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/gstid3demux.c: (gst_id3demux_sink_event):
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Don't leak newsegment events.
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2007-05-25 20:33:10 +00:00
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2007-05-25 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavparse/Makefile.am:
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Add '-lm' to LIBS for ceil(), don't assume one of our dependencies
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drags it in.
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2007-05-25 16:02:51 +00:00
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2007-05-25 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacenc.c: (gst_flac_enc_init),
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(notgst_value_array_append_buffer),
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(gst_flac_enc_process_stream_headers),
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(gst_flac_enc_write_callback), (gst_flac_enc_chain),
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(gst_flac_enc_change_state):
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* ext/flac/gstflacenc.h:
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Collect headers, add "streamheader" field to output caps and set
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BUFFER_IN_CAPS flag on pushed header buffers. That way oggmux
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produces output according to the official FLAC-to-Ogg mapping
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instead of completely broken files. Fixes #426044.
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2007-05-25 10:44:12 +00:00
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2007-05-25 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/gstid3demux.c: (gst_id3demux_reset),
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(gst_id3demux_send_new_segment), (gst_id3demux_chain),
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(gst_id3demux_sink_event):
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* gst/id3demux/gstid3demux.h:
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset),
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(gst_tag_demux_chain), (gst_tag_demux_sink_event),
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(gst_tag_demux_send_new_segment):
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Handle and adjust new-segment events so that downstream really
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sees a stream with the tag pieces stripped off the front and back.
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Fixes strangeness in seeking when mp3 decoders use the new-segment
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byte position to estimate their current playback position timestamp
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and then the arriving buffers don't match up.
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2007-05-25 10:23:49 +00:00
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2007-05-25 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_detect):
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Don't unnecessarily perform a READY->NULL->READY transition on the
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detected audio sink when starting up. Fixes: #440127
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2007-05-24 17:00:21 +00:00
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2007-05-24 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacenc.c: (gst_flac_enc_sink_setcaps),
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(gst_flac_enc_chain):
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Don't crash in chain function if setcaps hasn't been called.
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2007-05-24 08:14:00 +00:00
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2007-05-24 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_methods):
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Init value to avoid infinte loops.
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gst/rtsp/: Fix for new API.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
(gst_rtspsrc_play):
(rtsp_connection_send), (rtsp_connection_receive):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
Fix for new API.
* gst/rtsp/rtspconnection.c: (add_auth_header),
Only add authorisation and session headers when sending messages.
* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
(rtsp_message_init_request), (rtsp_message_init_response),
(rtsp_message_unset), (rtsp_message_add_header),
(rtsp_message_remove_header), (rtsp_message_get_header),
(rtsp_message_append_headers), (dump_key_value),
(rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Add support for multiple headers of the same type by storing the parsed
headers in a GArray instaed of a hashtable.
2007-05-24 08:10:42 +00:00
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2007-05-24 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_auth),
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(gst_rtspsrc_try_send), (gst_rtspsrc_parse_methods),
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(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
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(gst_rtspsrc_play):
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(rtsp_connection_send), (rtsp_connection_receive):
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send):
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Fix for new API.
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* gst/rtsp/rtspconnection.c: (add_auth_header),
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Only add authorisation and session headers when sending messages.
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* gst/rtsp/rtspmessage.c: (key_value_foreach), (rtsp_message_init),
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(rtsp_message_init_request), (rtsp_message_init_response),
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(rtsp_message_unset), (rtsp_message_add_header),
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(rtsp_message_remove_header), (rtsp_message_get_header),
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(rtsp_message_append_headers), (dump_key_value),
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(rtsp_message_dump):
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* gst/rtsp/rtspmessage.h:
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Add support for multiple headers of the same type by storing the parsed
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headers in a GArray instaed of a hashtable.
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2007-05-21 10:07:05 +00:00
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2007-05-21 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
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(gst_udpsrc_create), (gst_udpsrc_unlock), (gst_udpsrc_unlock_stop):
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Since we depend on 0.10.13 -core, override the unlock_stop vmethod for
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safer shutdown.
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2007-05-21 10:03:42 +00:00
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2007-05-21 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init):
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* gst/rtsp/gstrtpdec.h:
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Added signal for backwards compat.
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2007-05-21 09:32:26 +00:00
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2007-05-21 Sebastian Dröge <slomo@circular-chaos.org>
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Patch by: René Stadler <mail at renestadler dot de>
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* configure.ac:
|
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* gst/auparse/gstauparse.c: (gst_au_parse_reset),
|
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(gst_au_parse_parse_header), (gst_au_parse_chain):
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* gst/auparse/gstauparse.h:
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Use audioconvert for converting from non-native endianness floats
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in auparse instead of doing it ourself. Fixes #424527.
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This needs the audioconvert from plugins-base CVS.
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2007-05-21 09:29:30 +00:00
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2007-05-21 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
|
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(gst_rtp_h263p_pay_flush):
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Fix enum registration.
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2007-05-21 08:57:18 +00:00
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2007-05-21 Wim Taymans <wim@fluendo.com>
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Patch by: Antoine Tremblay <hexa00 at gmail dot com>
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* gst/rtp/gstrtph263ppay.c: (gst_fragmentation_mode_get_type),
|
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(gst_rtp_h263p_pay_class_init), (gst_rtp_h263p_pay_init),
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(gst_rtp_h263p_pay_set_property), (gst_rtp_h263p_pay_get_property),
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(gst_rtp_h263p_pay_flush):
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* gst/rtp/gstrtph263ppay.h:
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Add new fragmentation mode base on GOB headers. Fixes #438940.
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2007-05-20 14:05:42 +00:00
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2007-05-20 Tim-Philipp Müller <tim at centricular dot net>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp):
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Printf format fix.
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2007-05-18 13:27:39 +00:00
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2007-05-18 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams):
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Don't crash when an unsupported transport error was returned by the
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server, just try to configure the next stream. Fixes #439255.
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gst/rtsp/gstrtspsrc.*: Add TCP timeout property and use it for all TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
Add TCP timeout property and use it for all TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_write), (rtsp_connection_next_timeout),
(rtsp_connection_reset_timeout):
Make connect and writes cancelable and make them use the timeout.
2007-05-18 11:39:12 +00:00
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2007-05-18 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
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(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
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(gst_rtspsrc_stream_configure_udp), (gst_rtspsrc_loop_interleaved),
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(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
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(gst_rtspsrc_setup_streams), (gst_rtspsrc_open):
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* gst/rtsp/gstrtspsrc.h:
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Add TCP timeout property and use it for all TCP connection.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
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(rtsp_connection_write), (rtsp_connection_next_timeout),
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(rtsp_connection_reset_timeout):
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Make connect and writes cancelable and make them use the timeout.
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2007-05-18 10:36:12 +00:00
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2007-05-18 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
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(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_try_send), (gst_rtspsrc_send),
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(gst_rtspsrc_setup_streams):
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Refactor timeout handling.
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Also send keep-alive when dealing with TCP transport.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
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(rtsp_connection_free), (rtsp_connection_next_timeout),
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(rtsp_connection_reset_timeout):
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* gst/rtsp/rtspconnection.h:
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Use a timer to handle the session timeouts, add some methods to deal
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with timeouts.
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2007-05-17 14:56:39 +00:00
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2007-05-17 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
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(gst_rtspsrc_setup_streams):
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Ignore streams that fail the setup command, we will retry with a
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different transport later on.
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
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(rtsp_ext_wms_configure_stream):
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Fix encoding name case.
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2007-05-16 16:50:23 +00:00
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2007-05-16 Edward Hervey <edward@fluendo.com>
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* ext/libpng/gstpngdec.c: (user_endrow_callback), (user_read_data):
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Fix build on macosx.
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2007-05-16 16:30:03 +00:00
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2007-05-16 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/raw1394/gstdv1394src.c: (gst_dv1394src_uri_set_uri):
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Replace direct comparison of a string with the string literal "" with
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a comparison of the first character with '\0'. Fixes #438926.
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2007-05-15 11:18:33 +00:00
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2007-05-15 Stefan Kost <ensonic@users.sf.net>
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* gst/debug/breakmydata.c (gst_break_my_data_init):
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One more try. This should be the proper fix now.
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2007-05-15 06:41:58 +00:00
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2007-05-15 Stefan Kost <ensonic@users.sf.net>
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* gst/debug/breakmydata.c:
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Ooops, no // comments please.
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2007-05-15 06:34:48 +00:00
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2007-05-15 Stefan Kost <ensonic@users.sf.net>
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* gst/debug/breakmydata.c: (gst_break_my_data_class_init),
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(gst_break_my_data_init):
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Fix gst_buffer_is_writable() assertion.
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2007-05-15 02:56:23 +00:00
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2007-05-14 David Schleef <ds@schleef.org>
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* sys/v4l2/gstv4l2src.c: Add support for Bayer images as
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video/x-raw-bayer. Fixes #314160.
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2007-05-14 17:10:12 +00:00
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2007-05-14 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtptheoradepay.c: (decode_base64),
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(gst_rtp_theora_depay_parse_configuration):
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* gst/rtp/gstrtptheorapay.c: (encode_base64),
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(gst_rtp_theora_pay_finish_headers),
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(gst_rtp_theora_pay_handle_buffer):
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Update theora pay/depayloader in a similar to vorbis.
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* gst/rtp/gstrtpvorbisdepay.c:
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(gst_rtp_vorbis_depay_parse_configuration):
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Update docs.
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2007-05-14 16:19:58 +00:00
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2007-05-14 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send):
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When we try to execute a method that is not supported by the server,
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don't error out but remove the method from the accepted methods so that
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we never try to perform this method again.
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2007-05-14 14:47:26 +00:00
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2007-05-14 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
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Remove annoying _dump_mem.
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2007-05-14 11:11:42 +00:00
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2007-05-14 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_range):
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Parse range correctly.
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* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
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The baseurl now always has a '/' at the start.
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2007-05-14 09:01:05 +00:00
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2007-05-14 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_configure_caps),
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(gst_rtspsrc_parse_range), (gst_rtspsrc_open),
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(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
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Factor out caps configuration and configure more stuff such as the time
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ranges and speed/scale values.
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* gst/rtsp/rtsptransport.c:
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Add Copyright after non-trival fixes.
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2007-05-12 16:37:50 +00:00
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2007-05-12 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/rtsp/gstrtspsrc.h:
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|
* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
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* gst/rtsp/rtspmessage.c: (rtsp_message_init_data),
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(rtsp_message_get_header):
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* gst/rtsp/rtspmessage.h:
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|
Make channel guint8 where possible.
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Make rtsp_message_init_data() take the channel as a guint8.
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* gst/rtsp/rtspdefs.c:
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Fixed a typo: Timout -> Timeout
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* gst/rtsp/rtspdefs.h:
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Make RTSP_CHECK() behave as a statement.
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* gst/rtsp/sdpmessage.c:
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Avoid a compiler warning in INIT_ARRAY().
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Fixes #437692.
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2007-05-12 16:27:51 +00:00
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2007-05-12 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free),
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(rtsp_url_get_request_uri):
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|
* gst/rtsp/rtspurl.h:
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|
|
Add support for query parameters to RTSP URLs.
|
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|
|
gst/rtsp/rtsptransport.*: Add validation to rtsp_transport_parse().
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
(rtsp_transport_parse), (rtsp_transport_as_text):
* gst/rtsp/rtsptransport.h:
Add validation to rtsp_transport_parse().
Add rtsp_transport_as_text() to generate an RTSP header from an
RTSPTransport.
Change ssrc to guint (was a string) since that is what it is, even
though it is sent as a hex string.
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
incorrect, which can be seen when looking at the examples in the RFC).
Fixes #437670.
2007-05-12 16:26:06 +00:00
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2007-05-12 Wim Taymans <wim@fluendo.com>
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|
Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
|
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|
(parse_range), (range_as_text), (rtsp_transport_mode_as_text),
|
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|
|
(rtsp_transport_profile_as_text), (rtsp_transport_ltrans_as_text),
|
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|
|
(rtsp_transport_parse), (rtsp_transport_as_text):
|
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|
|
* gst/rtsp/rtsptransport.h:
|
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|
Add validation to rtsp_transport_parse().
|
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|
Add rtsp_transport_as_text() to generate an RTSP header from an
|
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|
RTSPTransport.
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Change ssrc to guint (was a string) since that is what it is, even
|
|
|
|
though it is sent as a hex string.
|
|
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|
Correctly identify PLAY|RECORD mode parameters (the syntax in the RFC is
|
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|
|
incorrect, which can be seen when looking at the examples in the RFC).
|
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Fixes #437670.
|
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2007-05-11 16:11:04 +00:00
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2007-05-11 Zaheer Abbas Merali <<zaheerabbas at merali dot org>>
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Patch by: Eric Anholt
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* sys/ximage/gstximagesrc.c (gst_ximage_src_open_display,
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gst_ximage_src_ximage_get):
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Use union of all damage between frames to make it faster.
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Fixes bug #342463.
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Also fix crasher when cursor is at bottom right of window.
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2007-05-11 16:01:45 +00:00
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2007-05-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
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Skip LIST chunks before the fmt chunk (fixes #437499). Also fix
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streaming mode regression for file from #343837 with 'bext' chunk
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before the 'fmt' chunk.
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gst/rtsp/: Preliminary seek support.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
(gst_rtspsrc_handle_src_event),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.h:
Preliminary seek support.
Activate internal pads so that we can receive events on them.
Don't try to parse a range string when it's NULL.
2007-05-11 15:09:39 +00:00
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2007-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_flush),
|
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|
(gst_rtspsrc_do_seek), (gst_rtspsrc_perform_seek),
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|
(gst_rtspsrc_handle_src_event),
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(gst_rtspsrc_stream_configure_manager),
|
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|
(gst_rtspsrc_stream_configure_tcp), (gst_rtspsrc_loop_interleaved),
|
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|
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open),
|
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|
|
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
|
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|
* gst/rtsp/gstrtspsrc.h:
|
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|
* gst/rtsp/rtspdefs.h:
|
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|
Preliminary seek support.
|
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|
Activate internal pads so that we can receive events on them.
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Don't try to parse a range string when it's NULL.
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2007-05-11 15:04:38 +00:00
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2007-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtp/README:
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Update README with new RTP variables that will be used for
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|
synchronisation.
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* gst/rtp/gstrtpvorbisdepay.c: (decode_base64),
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(gst_rtp_vorbis_depay_parse_configuration),
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(gst_rtp_vorbis_depay_process):
|
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|
* gst/rtp/gstrtpvorbispay.c: (encode_base64),
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|
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(gst_rtp_vorbis_pay_finish_headers),
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|
(gst_rtp_vorbis_pay_handle_buffer):
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|
Update vorbis pay and depayloader to draft-04.
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|
2007-05-11 11:24:13 +00:00
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2007-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtsptransport.c:
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UDP MCAST is actually the default for RTP/AVP.
|
2007-05-13 15:47:13 +00:00
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|
2007-05-13 Sebastien Moutte <sebastien@moutte.net>
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* gst/level/gstlevel.c: (gst_level_transform_ip):
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Use guint8 * instead of gpointer then vs6 can build
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|
in_data += (filter->width / 8).
|
2007-05-11 11:24:13 +00:00
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2007-05-11 10:31:27 +00:00
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2007-05-11 Zaheer Abbas Merali <<zaheerabbas at merali dot org>>
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* sys/ximage/gstximagesrc.c (gst_ximage_src_start,
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gst_ximage_src_ximage_get):
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* sys/ximage/gstximagesrc.h (last_ximage):
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When using Damage actually keep the last frame, and not assume
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that the buffer we get already has the last frame on it.
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Copy the cursor over if we specify a non-zero start x and
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start y.
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2007-05-11 09:12:55 +00:00
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2007-05-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtsptransport.c:
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Make UDP the default transport when not specified.
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2007-05-10 01:21:19 +00:00
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2007-05-09 David Schleef <ds@schleef.org>
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* gst/level/gstlevel.c:
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Revert last change.
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2007-05-09 21:30:53 +00:00
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2007-05-09 Sebastien Moutte <sebastien@moutte.net>
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* gst/level/gstlevel.c: (gst_level_calculate_##TYPE),
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(gst_level_transform_ip):
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Use guint8 * instead of gpointer then vs6 know the size of data
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pointed when moving the pointer.
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* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_handle_buffer):
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Move instructions after variables declaration.
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* win32/vs6/autogen.dsp:
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* win32/vs6/libgstrtp.dsp:
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* win32/vs6/libgstrtsp.dsp:
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Update vs6 project files.
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2007-05-09 11:23:39 +00:00
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2007-05-09 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/Makefile.am:
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_query),
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(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_open):
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* gst/rtsp/rtsprange.c: (parse_npt_time), (parse_npt_range),
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(parse_clock_range), (parse_smpte_range), (rtsp_range_parse),
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(rtsp_range_free):
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* gst/rtsp/rtsprange.h:
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Add code to parse time ranges.
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Report DURATION on the stream when possible.
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2007-05-08 15:49:01 +00:00
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2007-05-08 Tim-Philipp Müller <tim at centricular dot net>
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* gst/videomixer/videomixer.c: (gst_videomixer_blend_ayuv_ayuv),
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(gst_videomixer_fill_checker), (gst_videomixer_fill_color),
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(gst_videomixer_collected):
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Fix strides calculation for AYUV (it's just width*4) (#436910).
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2007-05-06 21:32:40 +00:00
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2007-05-06 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/audiofx/audioamplify.c: (gst_audio_amplify_transform_ip):
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* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_transform_ip):
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* gst/audiofx/audioinvert.c: (gst_audio_invert_transform_ip):
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Sync the GObject properties before each processing step to properly
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work with the controller.
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2007-05-04 15:17:14 +00:00
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2007-05-04 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive),
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(gst_rtspsrc_loop_udp), (gst_rtspsrc_try_send), (gst_rtspsrc_send),
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(gst_rtspsrc_setup_streams), (gst_rtspsrc_open),
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(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
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(gst_rtspsrc_change_state):
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Let more error state trickle down so that we can catch more error
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cases.
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Handle keep-alive a little smarter by selecting a method the server
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actually supports.
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Fix a race in UDP streaming shutdown.
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2007-05-04 13:04:31 +00:00
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2007-05-04 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send_keep_alive):
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Ignore errors when trying to use the keep-alive messages.
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gst/rtsp/gstrtspsrc.c: Send RTCP messages back to the server over the TCP connection.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport):
Send RTCP messages back to the server over the TCP connection.
* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Factor out and expose lowlevel _write and _read methods.
Implement sending data messages to the server.
2007-05-04 12:31:32 +00:00
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2007-05-04 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_handle_src_event),
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(gst_rtspsrc_handle_src_query), (gst_rtspsrc_sink_chain),
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(gst_rtspsrc_stream_configure_manager),
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(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
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(gst_rtspsrc_stream_configure_mcast),
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(gst_rtspsrc_stream_configure_udp),
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(gst_rtspsrc_stream_configure_udp_sink),
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(gst_rtspsrc_stream_configure_transport):
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Send RTCP messages back to the server over the TCP connection.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_write),
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(rtsp_connection_send), (rtsp_connection_read), (read_body),
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(rtsp_connection_receive):
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* gst/rtsp/rtspconnection.h:
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Factor out and expose lowlevel _write and _read methods.
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Implement sending data messages to the server.
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2007-05-03 15:55:06 +00:00
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2007-05-03 Wim Taymans <wim@fluendo.com>
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* gst/multipart/multipartmux.c: (gst_multipart_mux_queue_pads),
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(gst_multipart_mux_collected):
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Fix timestamps on outgoing buffers.
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2007-05-03 14:39:09 +00:00
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2007-05-03 Wim Taymans <wim@fluendo.com>
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* gst/multipart/multipartmux.c:
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(gst_multipart_mux_request_new_pad), (gst_multipart_mux_collected),
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(gst_multipart_mux_change_state):
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Emit NEWSEGMENT events before pushing the first buffer.
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gst/rtsp/gstrtspsrc.c: Refactor transport configuration code.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
(gst_rtspsrc_handle_src_query),
(gst_rtspsrc_stream_configure_manager),
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
(gst_rtspsrc_stream_configure_mcast),
(gst_rtspsrc_stream_configure_udp),
(gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
Refactor transport configuration code.
Create internal pads for TCP transport so that we can implement events
and queries.
Handle events and queries.
Parse range from the SDP.
Fix race in pause handler where the connection could still be flushing.
2007-05-03 13:48:54 +00:00
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2007-05-03 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
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|
(gst_rtspsrc_alloc_udp_ports), (gst_rtspsrc_handle_src_event),
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|
(gst_rtspsrc_handle_src_query),
|
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|
(gst_rtspsrc_stream_configure_manager),
|
|
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|
(gst_rtspsrc_stream_free_udp), (gst_rtspsrc_stream_configure_tcp),
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|
(gst_rtspsrc_stream_configure_mcast),
|
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(gst_rtspsrc_stream_configure_udp),
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|
(gst_rtspsrc_stream_configure_udp_sink),
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|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
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|
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|
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
|
|
|
|
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
|
|
|
|
(gst_rtspsrc_pause):
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|
|
Refactor transport configuration code.
|
|
|
|
Create internal pads for TCP transport so that we can implement events
|
|
|
|
and queries.
|
|
|
|
Handle events and queries.
|
|
|
|
Parse range from the SDP.
|
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|
|
Fix race in pause handler where the connection could still be flushing.
|
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|
|
|
gst/rtsp/gstrtspsrc.*: Fix race when multiple udp sources post timeouts, just act on the first received timeout.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
(gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Fix race when multiple udp sources post timeouts, just act on the first
received timeout.
Protect stream list with a recursive lock to fix some races.
Flush connection when we need to do a reconnect or stop.
Make state lock recursive.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_close):
Some small cleanups.
2007-05-02 19:32:58 +00:00
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|
|
2007-05-02 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
|
|
|
|
(gst_rtspsrc_finalize), (new_session_pad), (request_pt_map),
|
|
|
|
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
|
|
|
|
(gst_rtspsrc_send), (gst_rtspsrc_async_open), (gst_rtspsrc_close),
|
|
|
|
(gst_rtspsrc_play), (gst_rtspsrc_handle_message),
|
|
|
|
(gst_rtspsrc_change_state):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Fix race when multiple udp sources post timeouts, just act on the first
|
|
|
|
received timeout.
|
|
|
|
Protect stream list with a recursive lock to fix some races.
|
|
|
|
Flush connection when we need to do a reconnect or stop.
|
|
|
|
Make state lock recursive.
|
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|
|
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|
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
|
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|
(rtsp_connection_close):
|
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|
Some small cleanups.
|
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|
|
|
2007-05-02 18:25:09 +00:00
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|
2007-05-02 Wim Taymans <wim@fluendo.com>
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|
|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
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|
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
|
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|
Only set DISCONT when there actually is a discont or when we just
|
|
|
|
started.
|
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|
|
|
2007-05-02 18:01:52 +00:00
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|
|
2007-05-02 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* ext/flac/gstflac.c: (plugin_init):
|
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|
Call bindtextdomain() to get localized strings.
|
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|
2007-05-02 17:19:36 +00:00
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2007-05-02 Wim Taymans <wim@fluendo.com>
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|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
|
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|
(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
|
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|
|
(gst_wavparse_stream_data):
|
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|
|
* gst/wavparse/gstwavparse.h:
|
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|
Be a bit more clever when dealing with VBR files with FACT tags, we
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|
don't want to timestamp buffers in that case but the estimated BPS can
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|
be used for seeking.
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Only send close segment in the streaming thread.
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|
2007-05-02 17:08:09 +00:00
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|
2007-05-02 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
|
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|
Correctly post an error on the bus if something went wrong in the loop
|
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|
|
function. This fixes a few cases where the task was paused and nothing
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|
happened anymore.
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|
2007-05-02 14:27:28 +00:00
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2007-05-02 Wim Taymans <wim@fluendo.com>
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|
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* gst/rtsp/test.c: (main):
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|
Fix compilation of deprecated test just because I'm too lazy to delete
|
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|
it.
|
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|
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|
gst/rtsp/gstrtspsrc.*: Fix sending RTCP to the right place.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
Fix sending RTCP to the right place.
Fix bug in reffing the wrong UDP element.
Use new pad names for the session manager.
Implement handling server requests in interleaved and UDP modes.
Handle session keep-alive in UDP modes.
Remove GCond for handling UDP timeouts.
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
(rtsp_connection_send), (rtsp_connection_read), (read_body),
(rtsp_connection_receive), (rtsp_connection_close):
* gst/rtsp/rtspconnection.h:
Store connection IP address for later.
Add timeout args to all operations that might block forever.
Parse session timeout.
Only close sockets when not already closed.
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Add timeout return value and error string.
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
Add small comment.
2007-05-02 13:32:57 +00:00
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|
|
2007-05-02 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
|
|
|
|
(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
|
|
|
|
(gst_rtspsrc_handle_request), (gst_rtspsrc_loop_interleaved),
|
|
|
|
(gst_rtspsrc_send_keep_alive), (gst_rtspsrc_loop_udp),
|
|
|
|
(gst_rtspsrc_loop_send_cmd), (gst_rtspsrc_try_send),
|
|
|
|
(gst_rtspsrc_open), (gst_rtspsrc_handle_message):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Fix sending RTCP to the right place.
|
|
|
|
Fix bug in reffing the wrong UDP element.
|
|
|
|
Use new pad names for the session manager.
|
|
|
|
Implement handling server requests in interleaved and UDP modes.
|
|
|
|
Handle session keep-alive in UDP modes.
|
|
|
|
Remove GCond for handling UDP timeouts.
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_connect),
|
|
|
|
(rtsp_connection_send), (rtsp_connection_read), (read_body),
|
|
|
|
(rtsp_connection_receive), (rtsp_connection_close):
|
|
|
|
* gst/rtsp/rtspconnection.h:
|
|
|
|
Store connection IP address for later.
|
|
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|
Add timeout args to all operations that might block forever.
|
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|
Parse session timeout.
|
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|
Only close sockets when not already closed.
|
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|
|
|
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* gst/rtsp/rtspdefs.c:
|
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|
* gst/rtsp/rtspdefs.h:
|
|
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|
Add timeout return value and error string.
|
|
|
|
|
|
|
|
* gst/rtsp/rtspmessage.c: (rtsp_message_init_response):
|
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Add small comment.
|
|
|
|
|
2007-05-01 16:13:58 +00:00
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|
2007-05-01 Wim Taymans <wim@fluendo.com>
|
|
|
|
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|
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|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_init),
|
|
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|
(gst_rtp_mp4v_pay_empty), (gst_rtp_mp4v_pay_event):
|
|
|
|
* gst/rtp/gstrtpmp4vpay.h:
|
|
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|
Handle NEWSEGMENT and FLUSH events. Fixes #434824.
|
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|
|
|
2007-04-30 11:15:58 +00:00
|
|
|
2007-04-30 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
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|
Remove v4l2src from docs, since it breaks the docs build, and the
|
|
|
|
plugin is only built if --enable-experimental is used anyway.
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
Spaces => tab.
|
|
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|
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2007-04-29 14:43:37 +00:00
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2007-04-29 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (leave_multicast),
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(gst_multiudpsink_add), (gst_multiudpsink_remove):
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Add code to drop membership of a multicast group.
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* gst/udp/gstudpsink.c: (gst_udpsink_update_uri),
|
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(gst_udpsink_set_uri):
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Implement URI handler.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
|
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(gst_rtspsrc_parse_rtpinfo):
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Use URI handler to make udpsink instace.
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Improve code to configure port and destination.
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|
2007-04-29 12:19:21 +00:00
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2007-04-29 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
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Fix multicast detection.
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Don't try to join a multicast group if the address is not multicast.
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* gst/udp/gstudpsrc.c: (gst_udpsrc_update_uri):
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Small debug improvement.
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2007-04-27 16:44:17 +00:00
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2007-04-27 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
|
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(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
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(gst_rtspsrc_handle_message):
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Ignore ASYNC state messages from the udpsink, it's irrelevant for the
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parent.
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2007-04-27 15:30:39 +00:00
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2007-04-27 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpilbcdepay.h:
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Fix mode property when specified as an arg.
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|
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|
2007-04-26 15:08:20 +00:00
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2007-04-26 Edward Hervey <edward@fluendo.com>
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|
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|
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
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|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
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|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
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* docs/plugins/inspect/plugin-osxaudio.xml:
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Add documentation for osxaudio plugin.
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2007-04-26 10:08:27 +00:00
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2007-04-26 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize), (gst_rtspsrc_stream_configure_transport),
|
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(gst_rtspsrc_open), (gst_rtspsrc_close),
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(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play),
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(gst_rtspsrc_pause):
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* gst/rtsp/gstrtspsrc.h:
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Protect state changes with a lock.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
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(parse_line):
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* gst/rtsp/rtspconnection.h:
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Remove some unused stuff.
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2007-04-26 08:48:30 +00:00
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2007-04-26 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
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Handle the case where there are exactly 0 bytes to read and the ioctl
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did not report an error. Fixes #433530.
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2007-04-26 08:39:49 +00:00
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2007-04-26 Wim Taymans <wim@fluendo.com>
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|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
|
|
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|
* gst/wavparse/gstwavparse.h:
|
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|
Apply DISCONT to buffers.
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Only apply timestamp to the first sample after a DISCONT, too many VBR
|
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files cause random jitter in the timestamps. Fixes #433119.
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|
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|
2007-04-25 15:55:32 +00:00
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2007-04-25 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_class_init),
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(gst_rtp_dec_init), (gst_rtp_dec_set_property),
|
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(gst_rtp_dec_get_property):
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* gst/rtsp/gstrtpdec.h:
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Add dummy latency property to be backwards compat with rtpbin.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
|
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(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
|
|
|
|
(gst_rtspsrc_stream_configure_transport),
|
|
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|
(gst_rtspsrc_parse_rtpinfo):
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* gst/rtsp/gstrtspsrc.h:
|
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|
Add latency property and configure in the session manager.
|
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Don't set invalid clock-base and seqnum-base on caps, some servers
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sometimes don't send them.
|
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|
gst/alpha/gstalphacolor.c: Double-check that RGB input caps are really RGBA caps (apparently the core doesn't always ...
Original commit message from CVS:
* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
Double-check that RGB input caps are really RGBA caps (apparently
the core doesn't always catch it if those caps aren't a subset of
our template caps, also see #421543). Fixes #429319 in a way.
Also, don't leak the pad template in the transform_caps function.
* tests/check/Makefile.am:
* tests/check/elements/.cvsignore:
* tests/check/elements/alphacolor.c: (setup_alphacolor),
(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
(GST_START_TEST), (alphacolor_suite):
Add some basic unit tests for alphacolor.
2007-04-25 15:31:53 +00:00
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2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
|
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|
|
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps):
|
|
|
|
Double-check that RGB input caps are really RGBA caps (apparently
|
|
|
|
the core doesn't always catch it if those caps aren't a subset of
|
|
|
|
our template caps, also see #421543). Fixes #429319 in a way.
|
|
|
|
Also, don't leak the pad template in the transform_caps function.
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
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|
|
* tests/check/elements/.cvsignore:
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|
|
* tests/check/elements/alphacolor.c: (setup_alphacolor),
|
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|
|
(cleanup_alphacolor), (create_caps_rgb24), (create_caps_rgba32),
|
|
|
|
(create_buffer_rgb24_3x4), (create_buffer_rgba32_3x4),
|
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|
(GST_START_TEST), (alphacolor_suite):
|
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|
|
Add some basic unit tests for alphacolor.
|
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|
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|
2007-04-25 15:08:22 +00:00
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2007-04-25 Tim-Philipp Müller <tim at centricular dot net>
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* ext/libpng/gstpngdec.c: (gst_pngdec_task):
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If we get a fatal flow return in the loop function, first post the
|
|
|
|
error message and only then send the EOS event downstream, otherwise
|
|
|
|
applications might get an eos message before the error message and
|
|
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|
think everything was ok (related to #429319).
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2007-04-25 10:07:12 +00:00
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2007-04-25 Wim Taymans <wim@fluendo.com>
|
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|
|
|
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* gst/rtsp/rtspconnection.c: (rtsp_connection_receive):
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|
Read the channel byte as an unsigned byte.
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gst/rtp/: Make sure we configure the clock_rate in the baseclass in the setcaps function. Fixes #431282.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
(gst_rtp_gsm_depay_setcaps):
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
(gst_ilbc_depay_get_property):
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
* gst/rtp/gstrtpmp4adepay.c:
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
(gst_rtp_pcma_depay_setcaps):
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
(gst_rtp_pcmu_depay_setcaps):
Make sure we configure the clock_rate in the baseclass in the setcaps
function. Fixes #431282.
2007-04-25 09:47:48 +00:00
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2007-04-25 Wim Taymans <wim@fluendo.com>
|
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* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_set_property):
|
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init),
|
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(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
|
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|
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_init),
|
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(gst_rtp_gsm_depay_setcaps):
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|
|
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps):
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|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_setcaps):
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|
|
|
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_class_init),
|
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|
|
(gst_rtp_ilbc_depay_init), (gst_rtp_ilbc_depay_setcaps),
|
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|
|
(gst_rtp_ilbc_depay_process), (gst_ilbc_depay_set_property),
|
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|
(gst_ilbc_depay_get_property):
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|
|
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
|
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|
|
* gst/rtp/gstrtpmp4adepay.c:
|
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|
|
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_init),
|
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(gst_rtp_pcma_depay_setcaps):
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|
|
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_init),
|
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|
|
(gst_rtp_pcmu_depay_setcaps):
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|
|
Make sure we configure the clock_rate in the baseclass in the setcaps
|
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|
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function. Fixes #431282.
|
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|
|
|
2007-04-25 08:36:46 +00:00
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2007-04-25 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
|
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|
|
(gst_rtspsrc_stream_free), (request_pt_map),
|
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|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open):
|
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|
|
* gst/rtsp/gstrtspsrc.h:
|
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|
|
Parse server address from SDP.
|
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|
|
Hook up a udpsink to send RTCP back to the server.
|
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|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* gst/rtsp/rtsptransport.h:
|
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|
Add some docs.
|
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|
|
|
2007-04-25 06:52:09 +00:00
|
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|
2007-04-25 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
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|
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
|
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Make header field check conditional. Fixes #433135
|
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|
|
|
2007-04-24 09:12:42 +00:00
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|
2007-04-24 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
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|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-alphacolor.xml:
|
|
|
|
* gst/alpha/Makefile.am:
|
|
|
|
* gst/alpha/gstalphacolor.c:
|
|
|
|
* gst/alpha/gstalphacolor.h:
|
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|
Add minimal docs blurb to alphacolor; split out headers into
|
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|
|
separate header file for gtk-doc.
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2007-04-20 17:25:50 +00:00
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2007-04-20 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
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* gst/debug/progressreport.c: (gst_progress_report_report):
|
|
|
|
Don't try to post NULL message (in case we can't query upstream
|
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|
|
position or duration).
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|
2007-04-18 12:36:37 +00:00
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|
2007-04-18 Michael Smith <msmith@fluendo.com>
|
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|
|
|
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* gst/cutter/gstcutter.c: (gst_cutter_init), (gst_cutter_chain),
|
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|
|
(gst_cutter_get_caps):
|
|
|
|
* gst/cutter/gstcutter.h:
|
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|
|
Fix some of the most obvious bugs in cutter. Now doesn't leak
|
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|
|
everything if input is silent.
|
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|
2007-04-18 09:48:25 +00:00
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2007-04-18 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
|
|
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* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
|
|
|
|
(gst_wavenc_sink_setcaps), (gst_wavenc_change_state):
|
|
|
|
* gst/wavenc/gstwavenc.h:
|
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|
|
Wav apparently only supports width==GST_ROUND_UP(depth), everything
|
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|
|
else results in a invalid block align and invalid files.
|
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|
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|
2007-04-17 16:39:02 +00:00
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2007-04-17 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
|
Patch by: Snaik <snaik32 gmail com>
|
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|
|
|
|
|
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* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw):
|
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|
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Add missing break statement for BOX_HORIZONTAL case.
|
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|
|
|
2007-04-17 10:14:43 +00:00
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|
2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
|
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|
|
|
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|
Patch by: Vincent Torri <vtorri at univ-evry dot fr>
|
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|
|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
|
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Use correct format strings for integer types.
|
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|
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|
2007-04-17 02:51:02 +00:00
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|
|
2007-04-17 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
|
|
|
|
(gst_wavparse_create_sourcepad):
|
|
|
|
Use gst_riff_create_audio_template_caps () instead of the local caps.
|
|
|
|
This makes updates of the local caps unecessary whenever libgstriff
|
|
|
|
gets support for new formats.
|
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|
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|
2007-04-16 21:29:40 +00:00
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|
2007-04-16 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
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|
Patch by: Brian Cameron <brian.cameron at sun dot com>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudio.c:
|
|
|
|
* sys/sunaudio/gstsunaudiomixer.c:
|
|
|
|
* sys/sunaudio/gstsunaudiomixer.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.h:
|
|
|
|
* sys/sunaudio/gstsunaudiosink.c:
|
|
|
|
* sys/sunaudio/gstsunaudiosink.h:
|
|
|
|
* sys/sunaudio/gstsunaudiosrc.c:
|
|
|
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* sys/sunaudio/gstsunaudiosrc.h:
|
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|
|
Fix and/or update copyright attributions (#430228).
|
|
|
|
|
2007-04-13 09:32:21 +00:00
|
|
|
2007-04-13 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
Fix docs.
|
|
|
|
|
|
|
|
* gst/rtsp/URLS:
|
|
|
|
Add some more example urls.
|
|
|
|
|
|
|
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* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
|
|
|
|
(gst_rtp_dec_chain_rtp):
|
|
|
|
Better debugging.
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (request_pt_map),
|
|
|
|
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
|
|
|
|
(gst_rtspsrc_parse_rtpinfo):
|
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|
|
Remove unused code.
|
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2007-04-13 08:19:35 +00:00
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|
2007-04-13 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
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* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
|
|
|
|
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
|
|
|
|
(gst_wavparse_stream_data):
|
|
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|
Relax the audio/mpeg caps again and add FIXME: comment.
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2007-04-13 06:20:28 +00:00
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2007-04-13 Stefan Kost <ensonic@users.sf.net>
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* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
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(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
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(gst_wavparse_stream_data):
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* gst/wavparse/gstwavparse.h:
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More sanity check for the header fields. Fix type for 'rate' header
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field.
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2007-04-12 16:06:31 +00:00
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2007-04-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/icydemux/gsticydemux.c: (notgst_tag_freeform_string_to_utf8),
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(gst_icydemux_unicodify):
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If the metadata strings we get in the stream are not UTF-8, try to
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interpret them according to the character encodings specified in the
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GST_ICY_TAG_ENCODING and GST_TAG_ENCODING environment variables, and
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only fall back to locale/ISO-8859-1 if those aren't set or don't
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work. Should fix #428901.
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2007-04-12 14:20:56 +00:00
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2007-04-12 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtph264depay.c:
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Use the proper sync word for SPS and PPS.
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gst/rtp/Makefile.am: gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME, fnv1_hash_32_new, fnv1_hash_...
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
Add a simple hashing implementation that we can use to generate
a 24-bit ident value based on the codebooks for vorbis and theora.
* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
gst_rtp_theora_pay_handle_buffer):
* gst/rtp/gstrtpvorbisdepay.c
(gst_rtp_vorbis_depay_parse_configuration,
gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
Use the hashing function, ensuring that the same codebooks result
in the same ident and thus the same SDP description.
Various log fixes/changes.
2007-04-12 11:41:11 +00:00
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2007-04-12 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/rtp/Makefile.am:
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* gst/rtp/fnv1hash.c (MASK_24, FNV1_HASH_32_INIT, FNV1_HASH_32_PRIME,
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fnv1_hash_32_new, fnv1_hash_32_update, fnv1_hash_32_to_24):
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* gst/rtp/fnv1hash.h (__GST_FNV1_HASH_H__):
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Add a simple hashing implementation that we can use to generate
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a 24-bit ident value based on the codebooks for vorbis and theora.
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* gst/rtp/gstrtptheorapay.c (gst_rtp_theora_pay_finish_headers,
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gst_rtp_theora_pay_handle_buffer):
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* gst/rtp/gstrtpvorbisdepay.c
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(gst_rtp_vorbis_depay_parse_configuration,
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gst_rtp_vorbis_depay_switch_codebook, gst_rtp_vorbis_depay_process):
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* gst/rtp/gstrtpvorbispay.c (gst_rtp_vorbis_pay_reset_packet,
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gst_rtp_vorbis_pay_init_packet, gst_rtp_vorbis_pay_flush_packet,
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gst_rtp_vorbis_pay_finish_headers, gst_rtp_vorbis_pay_handle_buffer):
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Use the hashing function, ensuring that the same codebooks result
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in the same ident and thus the same SDP description.
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Various log fixes/changes.
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2007-04-12 11:37:50 +00:00
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2007-04-12 Wim Taymans <wim@fluendo.com>
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Patch by: jerry tan <jerry dot tan at sun dot com>
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* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
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remove the call of ioctl (fd, AUDIO_MIXER_MULTIPLE_OPEN), it is the
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application's responsibility to make sure it open the device once.
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Remove a careless error if AUDIODEV is set. Fixes #392620.
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2007-04-12 08:21:28 +00:00
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2007-04-12 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtp_dec_marshal_BOXED__UINT_UINT),
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(gst_rtp_dec_class_init), (gst_rtp_dec_chain_rtp):
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* gst/rtsp/gstrtpdec.h:
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Make backward compat with rtpbin by adding the request-pt-map signals.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
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(new_session_pad), (request_pt_map),
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(gst_rtspsrc_stream_configure_transport),
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(gst_rtspsrc_stream_configure_caps),
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(gst_rtspsrc_activate_streams):
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* gst/rtsp/gstrtspsrc.h:
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Implement request-pt-map signals instead of setting caps on the buffers
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for the session manager.
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2007-04-11 10:25:25 +00:00
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2007-04-11 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstudp.c: (plugin_init):
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Register GstNetBuffer in plugin_init so that the type can be used from
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multiple threads without races.
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2007-04-10 17:06:05 +00:00
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2007-04-10 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_setcaps),
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(gst_rtp_amr_depay_process):
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Fix depayloader clock_rate and some cleanups.
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* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_finalize),
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(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
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* gst/rtp/gstrtph264depay.h:
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Don't push codec_data in the adapter because it might get flushed when
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we get a discont.
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
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Handle multiple AU per packet.
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* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process),
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(gst_rtp_sv3v_depay_plugin_init):
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Disable rank, this one does not work.
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Remove timestamping, base class does that.
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2007-04-10 12:01:33 +00:00
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2007-04-10 Stefan Kost <ensonic@users.sf.net>
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* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
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limit caps to the formats we announce in the template
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* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
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(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
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(gst_wavparse_add_src_pad), (gst_wavparse_stream_data):
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fix some crashers/asserts when dealing with broken files
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2007-04-10 10:01:14 +00:00
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2007-04-10 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
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* gst/rtp/gstrtpL16depay.c:
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
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* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_get_mode),
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(gst_rtp_speex_depay_setcaps):
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_loop_udp):
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init_send):
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Fix some compiler warnings. Fixes #428182.
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gst/rtsp/: Morph RTPDec into something compatible with RTPBin as a fallback.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
(gst_rtp_dec_init), (gst_rtp_dec_finalize),
(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
(create_rtcp), (gst_rtp_dec_request_new_pad),
(gst_rtp_dec_release_pad):
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/gstrtsp.c: (plugin_init):
Morph RTPDec into something compatible with RTPBin as a fallback.
Various other style fixes.
* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
(new_session_pad), (gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Implement RTPBin session manager handling.
Don't try to add empty properties to caps.
Implement fallback session manager, handling.
Don't combine errors from RTCP streams, just ignore them.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
* gst/rtsp/rtsptransport.h:
Implement fallback session manager.
Make RTPBin the default one when available.
2007-04-06 12:54:16 +00:00
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2007-04-06 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/Makefile.am:
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* gst/rtsp/gstrtpdec.c: (find_session_by_id), (create_session),
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(free_session), (gst_rtp_dec_base_init), (gst_rtp_dec_class_init),
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(gst_rtp_dec_init), (gst_rtp_dec_finalize),
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(gst_rtp_dec_query_src), (gst_rtp_dec_chain_rtp),
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(gst_rtp_dec_chain_rtcp), (gst_rtp_dec_set_property),
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(gst_rtp_dec_get_property), (gst_rtp_dec_provide_clock),
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(gst_rtp_dec_change_state), (create_recv_rtp), (create_recv_rtcp),
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(create_rtcp), (gst_rtp_dec_request_new_pad),
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(gst_rtp_dec_release_pad):
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* gst/rtsp/gstrtpdec.h:
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* gst/rtsp/gstrtsp.c: (plugin_init):
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Morph RTPDec into something compatible with RTPBin as a fallback.
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Various other style fixes.
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* gst/rtsp/gstrtspsrc.c: (find_stream_by_id),
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(find_stream_by_udpsrc), (gst_rtspsrc_stream_free),
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(gst_rtspsrc_cleanup), (gst_rtspsrc_media_to_caps),
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(new_session_pad), (gst_rtspsrc_stream_configure_transport),
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(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
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(gst_rtspsrc_loop_udp), (gst_rtspsrc_setup_auth),
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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|
* gst/rtsp/gstrtspsrc.h:
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Implement RTPBin session manager handling.
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Don't try to add empty properties to caps.
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Implement fallback session manager, handling.
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Don't combine errors from RTCP streams, just ignore them.
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* gst/rtsp/rtsptransport.c: (rtsp_transport_get_manager):
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* gst/rtsp/rtsptransport.h:
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Implement fallback session manager.
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Make RTPBin the default one when available.
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2007-04-05 13:56:44 +00:00
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2007-04-05 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
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(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_plugin_init):
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This element is ready to be autoplugged.
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2007-04-05 11:26:25 +00:00
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2007-04-05 Julien MOUTTE <julien@moutte.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
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Don't leave the offsets defined by upstream element on the
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compressed data buffer we are pushing downstream. Make them
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GST_BUFFER_OFFSET_NONE.
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gst/avi/: Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
Original commit message from CVS:
* gst/avi/README:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
2007-04-04 12:39:41 +00:00
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2007-04-04 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/README:
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
|
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(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
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(gst_avi_demux_stream_index), (gst_avi_demux_sync),
|
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(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_stream_header_push),
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(gst_avi_demux_stream_header_pull), (gst_avi_demux_combine_flows),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
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Don't abort on out-of-memory. Use stream-nr as unsigned integer only.
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2007-04-03 09:55:45 +00:00
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2007-04-03 Wim Taymans <wim@fluendo.com>
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* gst/smpte/barboxwipes.c:
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Fix error as spotted by Snaik <snaik32 at gmail dot com>
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2007-03-30 17:19:34 +00:00
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2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/wavparse/gstwavparse.c:
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Support audio/x-raw-float in wav files. This only works with
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plugins-base CVS, using an older version doesn't have any
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disadvantages though.
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2007-03-30 15:59:27 +00:00
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2007-03-30 Sebastian Dröge <slomo@circular-chaos.org>
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* configure.ac:
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* gst/auparse/gstauparse.c: (gst_au_parse_reset),
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(gst_au_parse_parse_header), (gst_au_parse_chain):
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* gst/auparse/gstauparse.h:
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Revert last change as we don't want plugins-good to depend on
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plugins-base CVS now.
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2007-03-29 18:51:33 +00:00
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* configure.ac:
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Require gst-plugins-base CVS for audioconvert with non-native
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float support and width/depth fix in libgstriff.
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Patch by: René Stadler <mail at renestadler dot de>
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* gst/auparse/gstauparse.c: (gst_au_parse_reset),
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(gst_au_parse_parse_header), (gst_au_parse_chain):
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* gst/auparse/gstauparse.h:
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Don't swap the floats ourself if they're not in native endianness.
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Instead let audioconvert handle this. Fixes #339838.
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2007-03-29 14:40:35 +00:00
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2007-03-29 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstasteriskh263.h:
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* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_process),
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(gst_rtp_h263p_depay_change_state):
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* gst/rtp/gstrtph263pdepay.h:
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* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
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(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
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(gst_rtp_h264_depay_change_state):
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* gst/rtp/gstrtph264depay.h:
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* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_class_init),
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(gst_rtp_mp4a_depay_setcaps), (gst_rtp_mp4a_depay_process):
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
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Flush adapter on disconts.
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2007-03-29 14:03:21 +00:00
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2007-03-29 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_process):
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* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_process):
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* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_process):
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* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_process):
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_flush):
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* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
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(gst_rtp_mp4v_depay_process):
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* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_flush):
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* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_process):
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* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_flush):
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* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
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* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_process):
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* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_process):
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* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_process):
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Use more efficient adapter and rtpbuffer methods when possible.
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2007-03-29 12:14:22 +00:00
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/wavenc/gstwavenc.c: (gst_wavenc_create_header_buf),
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(gst_wavenc_sink_setcaps):
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Correctly handle width!=depth input.
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* gst/wavparse/gstwavparse.c:
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Already export in the caps that width==8 uses unsigned samples and
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everything else uses signed samples.
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gst/udp/: Rework the socket allocation a bit based on the sockfd argument so that it becomes usable.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
(gst_dynudpsink_init), (gst_dynudpsink_set_property),
(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
(gst_dynudpsink_close):
* gst/udp/gstdynudpsink.h:
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_set_property),
(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
* gst/udp/gstudpsrc.h:
Rework the socket allocation a bit based on the sockfd argument so that
it becomes usable.
Add a closefd property to instruct the udp elements to close the custom
file descriptors when going to READY. Fixes #423304.
API:GstUDPSrc::closefd property
API:GstDynUDPSink::closefd property
2007-03-29 09:59:23 +00:00
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2007-03-29 Wim Taymans <wim@fluendo.com>
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Patch by: Laurent Glayal <spglegle at yahoo dot fr>
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* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init),
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(gst_dynudpsink_init), (gst_dynudpsink_set_property),
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(gst_dynudpsink_get_property), (gst_dynudpsink_init_send),
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(gst_dynudpsink_close):
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* gst/udp/gstdynudpsink.h:
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
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(gst_udpsrc_create), (gst_udpsrc_set_property),
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(gst_udpsrc_get_property), (gst_udpsrc_start), (gst_udpsrc_stop):
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* gst/udp/gstudpsrc.h:
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Rework the socket allocation a bit based on the sockfd argument so that
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it becomes usable.
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Add a closefd property to instruct the udp elements to close the custom
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file descriptors when going to READY. Fixes #423304.
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API:GstUDPSrc::closefd property
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API:GstDynUDPSink::closefd property
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gst/rtp/: Added H264 payloader. Fixes #423782.
Original commit message from CVS:
Patch by: Laurent Glayal <spglegle at yahoo dot fr>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
(gst_rtp_h264_pay_plugin_init):
* gst/rtp/gstrtph264pay.h:
Added H264 payloader. Fixes #423782.
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
Small fixes.
2007-03-29 08:08:49 +00:00
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2007-03-29 Wim Taymans <wim@fluendo.com>
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Patch by: Laurent Glayal <spglegle at yahoo dot fr>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtph264pay.c: (gst_rtp_h264_pay_base_init),
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(gst_rtp_h264_pay_class_init), (gst_rtp_h264_pay_init),
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(gst_rtp_h264_pay_finalize), (gst_rtp_h264_pay_setcaps),
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(gst_rtp_h264_pay_handle_buffer), (gst_rtp_h264_pay_set_property),
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(gst_rtp_h264_pay_get_property), (gst_rtp_h264_pay_change_state),
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(gst_rtp_h264_pay_plugin_init):
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* gst/rtp/gstrtph264pay.h:
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Added H264 payloader. Fixes #423782.
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* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
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(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
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Small fixes.
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2007-03-28 22:27:36 +00:00
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/wavparse/gstwavparse.c:
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Actually support depths from 1 to 32, not only 8 to 32.
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2007-03-28 22:23:43 +00:00
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2007-03-29 Sebastian Dröge <slomo@circular-chaos.org>
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* gst/wavparse/gstwavparse.c:
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Add support for wav files containing audio/x-raw-int with random
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depths between 1 and 32 bits.
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gst/rtp/: Added MP4A-LATM depayloader. Fixes #417792.
Original commit message from CVS:
Based on patch by: Stefan Kost <ensonic@users.sf.net>
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
(gst_rtp_mp4a_depay_get_property),
(gst_rtp_mp4a_depay_change_state),
(gst_rtp_mp4a_depay_plugin_init):
* gst/rtp/gstrtpmp4adepay.h:
Added MP4A-LATM depayloader. Fixes #417792.
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
(gst_rtp_mp4v_depay_process):
Fixup depayloader, setting codec_data, using more efficient adaptor and
rtpbuffer handling.
* gst/rtsp/URLS:
Add url to test above.
2007-03-28 18:40:12 +00:00
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2007-03-28 Wim Taymans <wim@fluendo.com>
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Based on patch by: Stefan Kost <ensonic@users.sf.net>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
|
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* gst/rtp/gstrtpmp4adepay.c: (gst_rtp_mp4a_depay_base_init),
|
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(gst_rtp_mp4a_depay_class_init), (gst_rtp_mp4a_depay_init),
|
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(gst_rtp_mp4a_depay_finalize), (gst_rtp_mp4a_depay_setcaps),
|
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(gst_rtp_mp4a_depay_process), (gst_rtp_mp4a_depay_set_property),
|
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(gst_rtp_mp4a_depay_get_property),
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(gst_rtp_mp4a_depay_change_state),
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(gst_rtp_mp4a_depay_plugin_init):
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* gst/rtp/gstrtpmp4adepay.h:
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Added MP4A-LATM depayloader. Fixes #417792.
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* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps),
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(gst_rtp_mp4v_depay_process):
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Fixup depayloader, setting codec_data, using more efficient adaptor and
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rtpbuffer handling.
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* gst/rtsp/URLS:
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Add url to test above.
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gst/rtsp/gstrtspsrc.*: Handle default clock-rates for static payload types, rearrange stuff so that the rtpmap field ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_stream_configure_caps),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
* gst/rtsp/gstrtspsrc.h:
Handle default clock-rates for static payload types, rearrange stuff so
that the rtpmap field in the sdp can override the defaults.
Parse RTP-Info field to get the seqnum and timebase fields that should
go in the caps.
Delay configuring caps after we got the RTP-Info from the PLAY reply from
the server.
2007-03-25 15:34:42 +00:00
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2007-03-25 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (find_stream_by_setup),
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(gst_rtspsrc_create_stream), (gst_rtspsrc_stream_free),
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(get_default_rate_for_pt), (gst_rtspsrc_parse_rtpmap),
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(gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_stream_configure_transport),
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(gst_rtspsrc_stream_configure_caps),
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(gst_rtspsrc_activate_streams), (gst_rtspsrc_parse_rtpinfo):
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* gst/rtsp/gstrtspsrc.h:
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|
Handle default clock-rates for static payload types, rearrange stuff so
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that the rtpmap field in the sdp can override the defaults.
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Parse RTP-Info field to get the seqnum and timebase fields that should
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go in the caps.
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Delay configuring caps after we got the RTP-Info from the PLAY reply from
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the server.
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2007-03-22 09:44:17 +00:00
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2007-03-22 Wim Taymans <wim@fluendo.com>
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Patch by: Christophe Dehais <christophe dot dehais at gmail dot com>
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* ext/gconf/gconf.c: (gst_gconf_render_bin_with_default):
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Accept complex pipeline descriptions as an audio profile instead of just
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a single element. Fixes #420658.
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2007-03-21 11:49:32 +00:00
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2007-03-21 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_type):
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Rename registered type in preparation of GstTagDemux moving to
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-base at some point in the future.
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2007-03-19 10:29:19 +00:00
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2007-03-19 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
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Streaming mode fixes: don't unref buffer we don't own any longer;
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remove bogus adapter flush. Fixes #419338.
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2007-03-18 04:21:28 +00:00
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2007-03-17 David Schleef <ds@schleef.org>
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* REQUIREMENTS: Change the format to key/value, add a bunch of
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|
information, remove a bunch of requirements that are for
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|
other GStreamer packages.
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2007-03-18 02:00:54 +00:00
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2007-03-17 David Schleef <ds@schleef.org>
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* REQUIREMENTS: Fix a few things. This file really needs a
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good once-over.
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2007-03-15 12:05:01 +00:00
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2007-03-15 Edward Hervey <edward@fluendo.com>
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* sys/Makefile.am:
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Don't forget to distribute the sys/osxaudio/ directory.
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Activate osxaudio in gst-plugins-good with proper build setup.
Original commit message from CVS:
* configure.ac:
* sys/Makefile.am:
* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudio.c:
* sys/osxaudio/gstosxaudiosink.c:
(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
(gst_osx_audio_sink_getcaps),
(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
* sys/osxaudio/gstosxaudiosrc.c:
(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
(gst_osx_audio_src_create_ringbuffer):
* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
* sys/osxaudio/gstosxringbuffer.h:
Activate osxaudio in gst-plugins-good with proper build setup.
Add inlined documentation.
Fix debug statements
Fix ringbuffer when pausing.
Fixes #323471
2007-03-15 11:39:53 +00:00
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2007-03-15 Edward Hervey <edward@fluendo.com>
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* configure.ac:
|
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* sys/Makefile.am:
|
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* sys/osxaudio/Makefile.am:
|
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* sys/osxaudio/gstosxaudio.c:
|
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* sys/osxaudio/gstosxaudiosink.c:
|
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(gst_osx_audio_sink_osxelement_do_init), (gst_osx_audio_sink_init),
|
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|
(gst_osx_audio_sink_getcaps),
|
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(gst_osx_audio_sink_create_ringbuffer), (plugin_init):
|
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* sys/osxaudio/gstosxaudiosrc.c:
|
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(gst_osx_audio_src_osxelement_do_init), (gst_osx_audio_src_init),
|
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(gst_osx_audio_src_create_ringbuffer):
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* sys/osxaudio/gstosxringbuffer.c: (gst_osx_ring_buffer_get_type),
|
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(gst_osx_ring_buffer_class_init), (gst_osx_ring_buffer_init),
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(gst_osx_ring_buffer_acquire), (gst_osx_ring_buffer_start),
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(gst_osx_ring_buffer_pause), (gst_osx_ring_buffer_stop):
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* sys/osxaudio/gstosxringbuffer.h:
|
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Activate osxaudio in gst-plugins-good with proper build setup.
|
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Add inlined documentation.
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Fix debug statements
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Fix ringbuffer when pausing.
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Fixes #323471
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2007-03-14 22:21:26 +00:00
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2007-03-14 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
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* gst/rtp/gstrtppcmapay.c:
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* gst/rtp/gstrtppcmapay.h:
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* gst/rtp/gstrtppcmupay.c:
|
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* gst/rtp/gstrtppcmupay.h:
|
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Ported mulaw and alaw payloaders to use new base class
|
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2007-03-14 15:25:10 +00:00
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2007-03-14 Thomas Vander Stichele <thomas at apestaart dot org>
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* po/af.po:
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* po/az.po:
|
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* po/cs.po:
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* po/en_GB.po:
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* po/it.po:
|
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* po/nl.po:
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* po/or.po:
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* po/sq.po:
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* po/sr.po:
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* po/sv.po:
|
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* po/uk.po:
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* po/vi.po:
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Update translations.
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2007-03-14 14:49:45 +00:00
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|
2007-03-14 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Fix string replace error (AG_AG_GST_* => AG_GST_*).
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2007-03-12 17:56:54 +00:00
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|
2007-03-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_srcpad_event):
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Fix handling of -1 values for start and stop values when seeking,
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|
and SEEK_CUR+SEEK_END here as well.
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2007-03-12 17:24:23 +00:00
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2007-03-12 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/gstid3demux.c: (gst_id3demux_srcpad_event):
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Fix handling of -1 values for start and stop values when seeking,
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|
|
and SEEK_CUR+SEEK_END.
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2007-03-12 13:28:29 +00:00
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2007-03-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3v2frames.c: (parse_picture_frame):
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|
Fix parsing of ID3 v2.2.0 PIC frames. Only in version >= 2.3.0 is
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|
the image format a variable-length NUL-terminated string; in
|
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|
versions before that the image format is a fixed-length string of
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|
|
|
3 characters (see #348644 for a sample tag).
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Also make supplied mime type lower-case and fix up 'jpg' to 'jpeg'.
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|
2007-03-10 16:07:31 +00:00
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|
2007-03-10 Sebastien Moutte <sebastien@moutte.net>
|
|
|
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|
* win32/MANIFEST:
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|
Add new project files to MANIFEST.
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|
* win32/vs6/libgstaudiofx.dsp:
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|
|
* win32/vs6/libgstrtp.dsp:
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|
* win32/vs6/libgstrtsp.dsp:
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Update project files.
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|
2007-03-10 12:30:48 +00:00
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|
2007-03-10 Tim-Philipp Müller <tim at centricular dot net>
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|
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* gst/avi/gstavidemux.c: (gst_avi_demux_parse_subindex),
|
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|
(gst_avi_demux_parse_index):
|
|
|
|
* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
|
|
|
|
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
|
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|
Printf format fixes; also add some missing quotes in translated
|
|
|
|
strings. Fixes #416728 and #416727.
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|
2007-03-09 20:12:08 +00:00
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|
2007-03-09 Jan Schmidt <thaytan@mad.scientist.com>
|
|
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* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_find_best):
|
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|
Tim and I can't think of any reason the child audio sink needs to
|
|
|
|
be set back to NULL after successfully determining that it can
|
|
|
|
reach READY - it gets immediately set back to READY by the caller
|
|
|
|
anyway, causing an unnecessary close/open of any audio devices
|
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|
|
involved.
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|
2007-03-09 19:51:27 +00:00
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|
|
2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
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|
|
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|
* po/LINGUAS:
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* po/ja.po:
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|
Add ja.po file from #377306.
|
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|
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|
2007-03-09 19:44:30 +00:00
|
|
|
2007-03-09 Tim-Philipp Müller <tim at centricular dot net>
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|
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* sys/sunaudio/gstsunaudio.c: (plugin_init):
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* sys/sunaudio/gstsunaudiomixertrack.c:
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|
(gst_sunaudiomixer_track_new):
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|
|
Actually translate sunaudio mixer track labels instead of just
|
|
|
|
marking the strings as translatable (#377306); clean up weird
|
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|
label string mapping code that serves no apparent purpose. Also
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|
set the 'untranslated-label' property when creating mixer tracks
|
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|
if the GstMixerTrack base class supports this.
|
|
|
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|
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|
* tests/check/Makefile.am:
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|
* tests/check/elements/.cvsignore:
|
|
|
|
* tests/check/elements/sunaudio.c: (GST_START_TEST),
|
|
|
|
(sunaudio_suite):
|
|
|
|
Very minimalistic unit test for sunaudiomixer element (compiles, but not
|
|
|
|
actually tested on a system where sunaudiomixer is available).
|
|
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|
2007-03-09 18:49:37 +00:00
|
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|
2007-03-09 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
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|
* tests/check/Makefile.am:
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|
Re-enable the states test and see if it works on the buildbots.
|
|
|
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|
2007-03-09 17:32:32 +00:00
|
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|
2007-03-09 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
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|
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps),
|
|
|
|
(gst_dvdec_src_negotiate), (gst_dvdec_chain),
|
|
|
|
(gst_dvdec_change_state):
|
|
|
|
* ext/dv/gstdvdec.h:
|
|
|
|
Infer pixel-aspect-ratio from the video frame format if it isn't
|
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|
provided by the container, as happens when playing DV from AVI
|
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|
or Quicktime containers.
|
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|
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|
Patch by: Wim Taymans <wim@fluendo.com>
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Fixes #380944
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|
2007-03-09 17:05:17 +00:00
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2007-03-09 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
|
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|
When activated, remove the udpsrc timeout, we have dataflow and timeouts
|
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|
will later be handled by the jitterbuffer.
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2007-03-09 16:53:39 +00:00
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2007-03-09 Wim Taymans <wim@fluendo.com>
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* ext/taglib/gstid3v2mux.cc:
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|
Add write support for GST_TAG_MUSICBRAINZ_SORTNAME (TSOP) tag.
|
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|
Fixes #414496.
|
2007-03-12 15:49:02 +00:00
|
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|
Patch by: Alex Lancaster <alexl at users sourceforge net>
|
2007-03-09 16:53:39 +00:00
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|
2007-03-09 15:04:45 +00:00
|
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|
2007-03-09 Wim Taymans <wim@fluendo.com>
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|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_push_event), (gst_avi_demux_do_seek),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
|
|
|
|
(gst_avi_demux_chain):
|
|
|
|
Fix stream position reporting after a seek. Fixes #416445.
|
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|
2007-03-08 16:01:42 +00:00
|
|
|
2007-03-08 Wim Taymans <wim@fluendo.com>
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|
Patch by: René Stadler <mail at renestadler dot de>
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* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
|
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|
(gst_avi_demux_push_event), (gst_avi_demux_process_next_entry),
|
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|
|
(gst_avi_demux_stream_data), (gst_avi_demux_chain):
|
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|
Make avidemux accept optional header chunks in any order.
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Fixes #415446.
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|
2007-03-08 12:23:57 +00:00
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|
2007-03-08 Jan Schmidt <thaytan@mad.scientist.com>
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|
* tests/check/Makefile.am:
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|
Disable the states check until the remaining Valgrind errors
|
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|
|
are fixed or suppressed.
|
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|
2007-03-08 10:24:43 +00:00
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|
2007-03-08 Sebastian Dröge <slomo@circular-chaos.org>
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* tests/check/elements/.cvsignore:
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|
Add audiodynamic check to .cvsignore
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|
gst/audiofx/: Add new audiodynamic element which can act as a compressor or expander. Supported are hard-knee and sof...
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiodynamic.c:
(gst_audio_dynamic_characteristics_get_type),
(gst_audio_dynamic_mode_get_type),
(gst_audio_dynamic_set_process_function),
(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
(gst_audio_dynamic_transform_hard_knee_compressor_int),
(gst_audio_dynamic_transform_hard_knee_compressor_float),
(gst_audio_dynamic_transform_soft_knee_compressor_int),
(gst_audio_dynamic_transform_soft_knee_compressor_float),
(gst_audio_dynamic_transform_hard_knee_expander_int),
(gst_audio_dynamic_transform_hard_knee_expander_float),
(gst_audio_dynamic_transform_soft_knee_expander_int),
(gst_audio_dynamic_transform_soft_knee_expander_float),
(gst_audio_dynamic_transform_ip):
* gst/audiofx/audiodynamic.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new audiodynamic element which can act as a compressor or
expander. Supported are hard-knee and soft-knee operation modes with
user-specified ratio and threshold.
Attack and release parameters are not yet implemented but will follow.
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Integrate audiodynamic into the docs.
* tests/check/Makefile.am:
* tests/check/elements/audiodynamic.c: (setup_dynamic),
(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
Add unit test for audiodynamic.
2007-03-08 10:02:12 +00:00
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2007-03-08 Sebastian Dröge <slomo@circular-chaos.org>
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reviewed by: Stefan Kost <ensonic@users.sf.net>
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* gst/audiofx/Makefile.am:
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* gst/audiofx/audiodynamic.c:
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(gst_audio_dynamic_characteristics_get_type),
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(gst_audio_dynamic_mode_get_type),
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(gst_audio_dynamic_set_process_function),
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(gst_audio_dynamic_base_init), (gst_audio_dynamic_class_init),
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(gst_audio_dynamic_init), (gst_audio_dynamic_set_property),
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(gst_audio_dynamic_get_property), (gst_audio_dynamic_setup),
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(gst_audio_dynamic_transform_hard_knee_compressor_int),
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(gst_audio_dynamic_transform_hard_knee_compressor_float),
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(gst_audio_dynamic_transform_soft_knee_compressor_int),
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(gst_audio_dynamic_transform_soft_knee_compressor_float),
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(gst_audio_dynamic_transform_hard_knee_expander_int),
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(gst_audio_dynamic_transform_hard_knee_expander_float),
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(gst_audio_dynamic_transform_soft_knee_expander_int),
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(gst_audio_dynamic_transform_soft_knee_expander_float),
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(gst_audio_dynamic_transform_ip):
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* gst/audiofx/audiodynamic.h:
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* gst/audiofx/audiofx.c: (plugin_init):
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|
Add new audiodynamic element which can act as a compressor or
|
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|
|
expander. Supported are hard-knee and soft-knee operation modes with
|
|
|
|
user-specified ratio and threshold.
|
|
|
|
Attack and release parameters are not yet implemented but will follow.
|
|
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|
* docs/plugins/Makefile.am:
|
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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|
* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/inspect/plugin-audiofx.xml:
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Integrate audiodynamic into the docs.
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* tests/check/Makefile.am:
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* tests/check/elements/audiodynamic.c: (setup_dynamic),
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(cleanup_dynamic), (GST_START_TEST), (dynamic_suite), (main):
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Add unit test for audiodynamic.
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|
2007-03-07 19:48:03 +00:00
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2007-03-07 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/raw1394/gstdv1394src.c: (gst_dv1394src_start):
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Free handles that we allocated when exiting via the error paths.
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2007-03-07 11:24:05 +00:00
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2007-03-07 Stefan Kost <ensonic@users.sf.net>
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* gst/level/gstlevel.c: (gst_level_class_init),
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(gst_level_set_caps), (gst_level_start), (gst_level_event),
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(gst_level_transform_ip):
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* gst/level/gstlevel.h:
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Resolve message timestamps against the playback segment.
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2007-03-06 23:19:30 +00:00
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2007-03-06 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
|
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(gst_id3demux_sink_activate):
|
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Don't leak caps: make gst_id3demux_add_srcpad() not take ownership of the
|
2007-03-06 23:21:41 +00:00
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caps passed to it (previously one code path assumed it took ownership
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while another one assumed it didn't, while in fact it sometimes did and
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sometimes didn't ...).
|
2007-03-06 23:19:30 +00:00
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* configure.ac:
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* tests/files/Makefile.am:
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* tests/files/id3-407349-1.tag:
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* tests/files/id3-407349-2.tag:
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Add directory where data for unit tests can be stored.
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* tests/Makefile.am:
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* tests/check/Makefile.am:
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* tests/check/elements/.cvsignore:
|
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* tests/check/elements/id3demux.c: (pad_added_cb), (error_cb),
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(read_tags_from_file), (run_check_for_file),
|
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|
(check_date_1977_06_23), (GST_START_TEST), (id3demux_suite):
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Add unit test for id3demux, and in particular for bug #407349. Only
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testing pull-mode for now; push mode doesn't work yet because the test
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files are smaller than ID3_TYPE_FIND_MIN_SIZE.
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2007-03-06 22:14:59 +00:00
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2007-03-06 Tim-Philipp Müller <tim at centricular dot net>
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* tests/check/Makefile.am:
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Add missing backslash at end of line.
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2007-03-06 18:36:09 +00:00
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|
2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
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Trigger rebuild.
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2007-03-06 18:16:49 +00:00
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2007-03-06 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
|
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* gst/id3demux/id3tags.h:
|
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* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
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(parse_obsolete_tdat_frame):
|
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|
Do not convert obsolete TDA/TDAT frames to TDRC frames, otherwise
|
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|
|
the four-digit number will be interpreted as a year, whereas it is
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month and day in DDMM format. Instead, parse TDAT frames and fix up
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the date in the GST_TAG_DATE tag later if we also extracted a year.
|
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|
Fixes #407349.
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2007-03-06 14:53:04 +00:00
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2007-03-06 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
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(gst_switch_commit_new_kid):
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Fix up the dispose logic so it doesn't leak, and fix setting of
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the child state so that we don't set a child to our current state
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just as we are changing it to something else.
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2007-03-06 13:21:23 +00:00
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2007-03-06 Wim Taymans <wim@fluendo.com>
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* gst/goom/gstgoom.c: (gst_goom_src_setcaps), (get_buffer),
|
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(gst_goom_chain):
|
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* gst/goom/gstgoom.h:
|
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|
Document, fix and improve goom adapter behaviour.
|
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|
|
Fixes #407006.
|
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2007-03-05 18:43:29 +00:00
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|
2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/esd/esdsink.c: (gst_esdsink_open):
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Unref static pad template after using it.
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2007-03-05 17:17:04 +00:00
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2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_dispose),
|
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(gst_switch_commit_new_kid):
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|
Fix up the reference counting of the child elements.
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2007-03-05 17:08:32 +00:00
|
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|
2007-03-05 Wim Taymans <wim@fluendo.com>
|
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
|
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* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_finish_headers):
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_finish_headers):
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Fix encoding-name case.
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2007-03-05 16:39:29 +00:00
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2007-03-05 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init),
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(gst_rtp_speex_depay_get_mode), (gst_rtp_speex_depay_setcaps),
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(gst_rtp_speex_depay_process):
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* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_base_init),
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(gst_rtp_speex_pay_class_init), (gst_rtp_speex_pay_setcaps),
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(gst_rtp_speex_pay_parse_ident), (gst_rtp_speex_pay_handle_buffer),
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(gst_rtp_speex_pay_change_state):
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* gst/rtp/gstrtpspeexpay.h:
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Fix speex (de)payloader. Fixes #358040.
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2007-03-05 15:42:58 +00:00
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2007-03-05 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset),
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(gst_switch_commit_new_kid), (gst_switch_sink_set_child):
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Install fakesink in NULL by fixing some broken logic. This obviates
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the need to manually set _IS_SINK.
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Add some comments and remove a little cruft while I'm at it.
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2007-03-05 14:46:43 +00:00
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2007-03-05 Wim Taymans <wim@fluendo.com>
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_reset):
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Mark us as a sink when we have no fakesink in NULL. Fixes #414887.
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2007-03-04 18:52:12 +00:00
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2007-03-04 Tim-Philipp Müller <tim at centricular dot net>
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* po/POTFILES.in:
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Update.
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2007-03-04 17:33:34 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* tests/check/Makefile.am:
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Gah! Also disable gconfvideosink from the tests, otherwise
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it will instantiate autovideosink, and dfbvideosink and
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leak on the buildbots.
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2007-03-04 17:13:19 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open),
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(gst_cdio_cdda_src_finalize):
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Make sure we always destroy our libcdio handle.
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2007-03-04 17:05:58 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* tests/check/Makefile.am:
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Disable autovideosink so the buildbots don't barf over memory
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leaked in the directfb sink.
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2007-03-04 15:28:30 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_dispose):
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Chain up in dispose
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2007-03-04 15:07:15 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
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(gst_multipart_find_pad_by_mime):
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Use gst_pad_new_from_static_template instead of
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static_pad_template_get+pad_new.
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2007-03-04 14:56:53 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_create):
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Catch the case where no clock has been set.
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Fix a bunch of leaks shown by the newly-added states test.
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
(gst_gconf_audio_src_finalize), (do_toggle_element):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
(gst_gconf_video_src_finalize), (do_toggle_element):
* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
(gst_switch_sink_reset), (gst_switch_sink_set_child):
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
(gst_shout2send_init), (gst_shout2send_finalize):
* gst/debug/testplugin.c: (gst_test_class_init),
(gst_test_finalize):
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(gst_flxdec_dispose):
* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize):
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
* gst/rtsp/rtspextwms.h:
* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
(gst_smpte_finalize):
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
(gst_udpsink_finalize):
* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
(gst_wavparse_sink_activate):
* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
(gst_oss_src_finalize):
* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_finalize):
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
Fix a bunch of leaks shown by the newly-added states test.
2007-03-04 13:52:03 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/flac/gstflacenc.c: (gst_flac_enc_finalize):
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_class_init),
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(gst_gconf_audio_sink_dispose), (gst_gconf_audio_sink_finalize):
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* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
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(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_dispose),
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(gst_gconf_audio_src_finalize), (do_toggle_element):
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* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init),
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(gst_gconf_video_sink_class_init), (gst_gconf_video_sink_finalize),
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(do_toggle_element):
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* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
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(gst_gconf_video_src_class_init), (gst_gconf_video_src_dispose),
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(gst_gconf_video_src_finalize), (do_toggle_element):
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_class_init),
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(gst_switch_sink_reset), (gst_switch_sink_set_child):
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* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
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* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
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* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
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(gst_shout2send_init), (gst_shout2send_finalize):
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* gst/debug/testplugin.c: (gst_test_class_init),
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(gst_test_finalize):
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* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
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(gst_flxdec_dispose):
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* gst/multipart/multipartmux.c: (gst_multipart_mux_finalize):
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_finalize):
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
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(gst_rtspsrc_finalize):
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_free_context):
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* gst/rtsp/rtspextwms.h:
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* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
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(gst_smpte_finalize):
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_finalize):
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* gst/udp/gstudpsink.c: (gst_udpsink_class_init),
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(gst_udpsink_finalize):
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* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose),
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(gst_wavparse_sink_activate):
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* sys/oss/gstosssink.c: (gst_oss_sink_finalise):
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* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
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(gst_oss_src_finalize):
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* sys/v4l2/gstv4l2object.c: (gst_v4l2_object_destroy):
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* sys/v4l2/gstv4l2object.h:
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* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
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(gst_v4l2src_finalize):
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
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Fix a bunch of leaks shown by the newly-added states test.
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2007-03-04 13:41:00 +00:00
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2007-03-04 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/dv/gstdvdec.c: (gst_dvdec_init):
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Use gst_pad_new_from_static_template instead of
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static_pad_template_get+pad_new.
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2007-03-03 13:06:21 +00:00
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2007-03-03 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Loïc Minier <lool+gnome at via ecp fr>
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* ext/libcaca/Makefile.am:
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* gst/debug/Makefile.am:
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Don't mix tabs and spaces (#414168).
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2007-03-02 21:35:11 +00:00
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2007-03-02 Stefan Kost <ensonic@users.sf.net>
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* tests/check/generic/.cvsignore:
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Ignore files to please buildbot.
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2007-03-02 21:01:19 +00:00
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2007-03-02 Stefan Kost <ensonic@users.sf.net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers),
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(gst_wavparse_stream_data):
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Unbreak my previous commit (swapped nominator & denominator). Tim,
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thanks for spotting.
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2007-03-02 16:08:17 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_probe_devices),
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(gst_cdio_cdda_src_read_sector), (gst_cdio_cdda_src_open),
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(gst_cdio_cdda_src_finalize):
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Small code cleanups.
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Don't use pad_alloc as the base class cannot deal with the error codes.
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2007-03-02 13:40:06 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
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(gst_udpsrc_create):
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Fix doc.
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2007-03-02 13:29:25 +00:00
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2007-03-02 Stefan Kost <ensonic@users.sf.net>
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Patch by: René Stadler <mail@renestadler.de>
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* gst/wavparse/gstwavparse.c: (uint64_ceiling_scale_int),
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(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
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(gst_wavparse_stream_data):
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Handle rounding better to not drop last sample frame. Fixes #356692
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2007-03-02 13:19:57 +00:00
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2007-03-02 Jan Schmidt <thaytan@mad.scientist.com>
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* tests/check/Makefile.am:
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Disable cacasink from the states check too - it also calls exit(1)
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on us when it can't find a terminal to talk to.
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2007-03-02 12:56:13 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
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(gst_udpsrc_create), (gst_udpsrc_set_property),
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(gst_udpsrc_get_property):
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* gst/udp/gstudpsrc.h:
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Add support to strip proprietary headers. Fixes #350296.
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2007-03-02 12:52:56 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_process):
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Fix compilation.
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2007-03-02 12:16:16 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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Patch by: Thijs Vermeir <thijsvermeir at gmail dot com>
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* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_class_init),
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(gst_rtp_mp2t_depay_init), (gst_rtp_mp2t_depay_process),
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(gst_rtp_mp2t_depay_set_property),
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(gst_rtp_mp2t_depay_get_property):
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* gst/rtp/gstrtpmp2tdepay.h:
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Add support to strip off proprietary headers. Fixes #350278.
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2007-03-02 11:22:35 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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* ext/hal/hal.c:
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Fix compilation.
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2007-03-02 10:54:49 +00:00
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2007-03-02 Wim Taymans <wim@fluendo.com>
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* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_class_init),
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(gst_sunaudiosrc_init), (gst_sunaudiosrc_get_property),
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(gst_sunaudiosrc_open):
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* sys/sunaudio/gstsunaudiosrc.h:
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Remove device-name from GstSunAudioSrc. Fixes #412597.
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2007-03-01 21:50:36 +00:00
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2007-03-01 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/hal/gsthalaudiosink.c: (do_toggle_element):
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* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
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Having NULL as UDI previously selected the default sink/src. Change
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this back but mention it in the debug output.
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* ext/hal/hal.c: (gst_hal_get_alsa_element),
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(gst_hal_get_oss_element), (gst_hal_get_string),
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(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
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(gst_hal_get_audio_src):
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* ext/hal/hal.h:
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Refactor a bit, check all error conditions, greatly improve debugging
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and fix some possible memory leaks. Also implement OSS support
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and allow specifying an UDI that points to a real device. For this the
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child device which supports ALSA (preferred) or OSS is used.
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As a side effect this makes it impossible now to get a alsasink in
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halaudiosrc and a alsasrc in halaudiosink.
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2007-03-01 18:47:28 +00:00
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2007-03-01 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (find_stream_by_channel),
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(find_stream_by_udpsrc), (gst_rtspsrc_handle_message):
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Errors from the udp sources are not fatal unless all of them are in
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error.
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2007-03-01 18:14:42 +00:00
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2007-03-01 Jan Schmidt <thaytan@mad.scientist.com>
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* tests/check/Makefile.am:
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Disable aasink in the states test. I suspect this is the element that
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is calling exit(1) when it can't proceed.
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|
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2007-03-01 17:26:30 +00:00
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2007-03-01 Jan Schmidt <thaytan@mad.scientist.com>
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* tests/check/Makefile.am:
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Draw plugins in from the build tree sys/ dir, rather than picking
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|
up the already installed versions.
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2007-03-01 10:44:36 +00:00
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2007-03-01 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_open_display):
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Error out correctly when getting xcontext fails.
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2007-03-01 09:29:34 +00:00
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2007-03-01 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtpdec_change_state):
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Make state change to PAUSED NO_PREROLL because that's what it will be in
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the future and rtspsrc relies on it.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport),
|
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(gst_rtspsrc_change_state):
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Don't error out when we don't get an error from the state change
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function.
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2007-03-01 01:48:59 +00:00
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2007-03-01 Sebastian Dröge <slomo@circular-chaos.org>
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* ext/hal/gsthalaudiosink.c: (do_toggle_element):
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* ext/hal/gsthalaudiosrc.c: (do_toggle_element):
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Check if the device UDI is set before trying to query HAL
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about it and give a useful error message if it wasn't set.
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* ext/hal/hal.c: (gst_hal_get_string):
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Don't query HAL for NULL UDIs. Passing NULL as UDI to HAL
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gives an assertion failure in D-Bus when running with
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DBUS_FATAL_WARNINGS=1.
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2007-02-28 19:29:25 +00:00
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2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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Convert to new AG_GST style.
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2007-02-28 12:59:43 +00:00
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2007-02-28 Thomas Vander Stichele <thomas at apestaart dot org>
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* tests/check/Makefile.am:
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* tests/check/generic/states.c: (GST_START_TEST), (states_suite):
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add test for states
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2007-02-28 10:58:10 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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|
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* tests/check/elements/.cvsignore:
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Add new videofilter check to .cvsignore.
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2007-02-28 10:54:55 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_combine_flows),
|
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(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
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(gst_avi_demux_loop), (gst_avi_demux_chain):
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Fix combined flow return. Fixes #412608.
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2007-02-28 10:41:14 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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* gst/videofilter/Makefile.am:
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Dist header..
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2007-02-28 10:29:08 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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* gst/videofilter/gstgamma.h:
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Add header too.
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gst/videofilter/: Port gamma filter to 0.10. Fixes #412704.
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videofilter/Makefile.am:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
(gst_gamma_get_property), (gst_gamma_calculate_tables),
(oil_tablelookup_u8), (gst_gamma_set_caps),
(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
Port gamma filter to 0.10. Fixes #412704.
* tests/check/Makefile.am:
* tests/check/elements/videofilter.c: (setup_filter),
(cleanup_filter), (check_filter), (GST_START_TEST),
(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
Add unit tests for videofilters.
2007-02-28 10:17:15 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/videofilter/Makefile.am:
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* gst/videofilter/gstgamma.c: (gst_gamma_base_init),
|
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(gst_gamma_class_init), (gst_gamma_init), (gst_gamma_set_property),
|
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(gst_gamma_get_property), (gst_gamma_calculate_tables),
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(oil_tablelookup_u8), (gst_gamma_set_caps),
|
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(gst_gamma_planar411_ip), (gst_gamma_transform_ip), (plugin_init):
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Port gamma filter to 0.10. Fixes #412704.
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* tests/check/Makefile.am:
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* tests/check/elements/videofilter.c: (setup_filter),
|
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(cleanup_filter), (check_filter), (GST_START_TEST),
|
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(videobalance_suite), (videoflip_suite), (gamma_suite), (main):
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Add unit tests for videofilters.
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2007-02-28 10:06:27 +00:00
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2007-02-28 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/URLS:
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Add another interesting test url.
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* gst/rtsp/rtspmessage.c: (rtsp_message_get_header):
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Don't allow getting header fields from data packets.
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2007-02-27 23:43:08 +00:00
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2007-02-28 Michael Smith <msmith@fluendo.com>
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* ext/shout2/gstshout2.c: (gst_shout2send_class_init),
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(gst_shout2send_init), (gst_shout2send_start),
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(gst_shout2send_set_property), (gst_shout2send_get_property):
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* ext/shout2/gstshout2.h:
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Add a property for username.
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2007-02-27 11:59:21 +00:00
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2007-02-27 Christian Schallerr <christian@fluendo.com>
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* sys/osxaudio: Add Pioneers of the inevitable to the copyright list
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2007-02-26 12:07:14 +00:00
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2007-02-26 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/rtsp/Makefile.am:
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Fix make check too.
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2007-02-26 10:00:28 +00:00
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2007-02-26 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/rtsp/base64.c: (util_base64_encode):
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* gst/rtsp/base64.h:
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Commit missing files for base64 encoding.
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2007-02-24 22:57:49 +00:00
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2007-02-24 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Loïc Minier <lool+gnome at via ecp fr>
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* configure.ac:
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* ext/annodex/Makefile.am:
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* ext/jpeg/Makefile.am:
|
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* ext/speex/Makefile.am:
|
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* gst/alpha/Makefile.am:
|
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* gst/cutter/Makefile.am:
|
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* gst/debug/Makefile.am:
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* gst/effectv/Makefile.am:
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* gst/goom/Makefile.am:
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* gst/level/Makefile.am:
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* gst/smpte/Makefile.am:
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* gst/videofilter/Makefile.am:
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Fix build with LDFLAGS='-Wl,-z,defs' (#410997)
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2007-02-23 19:12:52 +00:00
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2007-02-23 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
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* gst/rtsp/Makefile.am:
|
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* gst/rtsp/rtspconnection.c: (append_auth_header),
|
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|
(rtsp_connection_send), (rtsp_connection_set_auth):
|
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|
|
g_base64_encode is a GLib 2.12 function. Use an equivalent taken
|
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|
from icecast to replace it. Relicensed from GPL courtesy of Mike
|
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|
Smith.
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gst/rtsp/: Implement simple Basic Authentication support so that urls like rtsp://user:pass@hostname/rtspstream work ...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(append_auth_header), (rtsp_connection_send),
(rtsp_connection_free), (rtsp_connection_set_auth):
* gst/rtsp/rtspconnection.h:
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
* gst/rtsp/rtspurl.h:
Implement simple Basic Authentication support so that urls like
rtsp://user:pass@hostname/rtspstream work on hosts that require
authentication.
2007-02-23 18:12:27 +00:00
|
|
|
2007-02-23 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_finalize),
|
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|
(gst_rtspsrc_create_stream), (rtsp_auth_method_to_string),
|
|
|
|
(gst_rtspsrc_parse_auth_hdr), (gst_rtspsrc_setup_auth),
|
|
|
|
(gst_rtspsrc_send), (gst_rtspsrc_try_send), (gst_rtspsrc_open),
|
|
|
|
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
|
|
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(gst_rtspsrc_uri_set_uri):
|
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|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
|
|
|
|
(append_auth_header), (rtsp_connection_send),
|
|
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|
(rtsp_connection_free), (rtsp_connection_set_auth):
|
|
|
|
* gst/rtsp/rtspconnection.h:
|
|
|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
* gst/rtsp/rtspurl.c: (rtsp_url_get_request_uri):
|
|
|
|
* gst/rtsp/rtspurl.h:
|
|
|
|
|
|
|
|
Implement simple Basic Authentication support so that urls like
|
|
|
|
rtsp://user:pass@hostname/rtspstream work on hosts that require
|
|
|
|
authentication.
|
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|
2007-02-22 17:53:26 +00:00
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|
2007-02-22 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
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* sys/v4l2/gstv4l2object.c:
|
|
|
|
* sys/v4l2/gstv4l2tuner.c:
|
|
|
|
* sys/v4l2/v4l2_calls.c:
|
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|
|
Fix segfault when oppening a radio device.
|
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|
2007-02-22 14:35:28 +00:00
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|
2007-02-22 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/level/gstlevel.c: (gst_level_set_caps),
|
|
|
|
(gst_level_transform_ip):
|
|
|
|
* sys/v4l2/README:
|
|
|
|
* tests/check/elements/level.c: (GST_START_TEST):
|
|
|
|
Fix level for multi-channel case.
|
|
|
|
|
2007-02-21 10:18:12 +00:00
|
|
|
2007-02-21 Stefan Kost <ensonic@users.sf.net>
|
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|
|
|
|
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|
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
|
|
|
|
(gst_level_transform_ip):
|
|
|
|
* gst/level/gstlevel.h:
|
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|
Use function pointer for process function and add process functions
|
|
|
|
for float audio.
|
|
|
|
|
2007-02-19 12:22:43 +00:00
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|
2007-02-19 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
|
|
|
|
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
|
|
|
|
(gst_v4l2src_capture_init):
|
|
|
|
Readd GST_ELEMENT_ERROR if we can't reenque buffers after EIO,
|
|
|
|
fixes #407369
|
|
|
|
|
2007-02-18 13:24:26 +00:00
|
|
|
2007-02-18 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_base_init),
|
|
|
|
(gst_rtp_mp2t_pay_class_init), (gst_rtp_mp2t_pay_init),
|
|
|
|
(gst_rtp_mp2t_pay_setcaps), (gst_rtp_mp2t_pay_handle_buffer),
|
|
|
|
(gst_rtp_mp2t_pay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmp2tpay.h:
|
|
|
|
Added simple mpeg transport stream payloader.
|
|
|
|
|
2007-02-16 12:32:01 +00:00
|
|
|
2007-02-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/URLS:
|
|
|
|
Add example H264 rtsp url.
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
|
|
|
|
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
|
|
|
|
Don't convert values to lowercase or we might mess up base64 encoded
|
|
|
|
properties.
|
|
|
|
|
2007-02-16 12:30:22 +00:00
|
|
|
2007-02-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/README:
|
|
|
|
Fix case of string params.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_class_init),
|
|
|
|
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process):
|
|
|
|
Fix depayloader, support more packet types.
|
|
|
|
Add sync codes to make sure the packetizer can do its job.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmp4gdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gpay.c:
|
|
|
|
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_process):
|
|
|
|
Fix caps case again.
|
|
|
|
|
2007-02-15 12:26:28 +00:00
|
|
|
2007-02-15 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
|
|
|
|
Set right caps on output buffers.
|
|
|
|
|
2007-02-14 17:04:47 +00:00
|
|
|
2007-02-14 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/sdpmessage.c: (sdp_parse_line):
|
|
|
|
As spotted by: Peter Kjellerstedt <pkj at axis com>:
|
|
|
|
Clear stack allocated SDPMedia struct before calling _init() on it.
|
|
|
|
Clarify this in the docs as well.
|
|
|
|
|
2007-02-14 17:01:25 +00:00
|
|
|
2007-02-14 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
|
|
|
|
(do_change_child):
|
|
|
|
Don't reset the profile when going switching states, as it makes
|
|
|
|
the element non-reusable.
|
|
|
|
|
gst/rtsp/sdpmessage.*: Fix memory management of SDP messages. Fixes #407793.
Original commit message from CVS:
* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
(sdp_parse_line):
* gst/rtsp/sdpmessage.h:
Based on patch by: jp.liu <jp_liu at astrocom dot cn>
Fix memory management of SDP messages. Fixes #407793.
2007-02-14 15:24:50 +00:00
|
|
|
2007-02-14 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/sdpmessage.c: (sdp_origin_init), (sdp_connection_init),
|
|
|
|
(sdp_bandwidth_init), (sdp_time_init), (sdp_zone_init),
|
|
|
|
(sdp_key_init), (sdp_attribute_init), (sdp_message_init),
|
|
|
|
(sdp_message_uninit), (sdp_message_free), (sdp_media_init),
|
|
|
|
(sdp_media_uninit), (sdp_media_free), (sdp_message_add_media),
|
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|
|
(sdp_parse_line):
|
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|
|
* gst/rtsp/sdpmessage.h:
|
|
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|
Based on patch by: jp.liu <jp_liu at astrocom dot cn>
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Fix memory management of SDP messages. Fixes #407793.
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2007-02-14 12:07:01 +00:00
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2007-02-14 Stefan Kost <ensonic@users.sf.net>
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Patch by: zhangfei gao <gaozhangfei@yahoo.com.cn>
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* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
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Allow muxing video/x-h264 (was already in the caps). Fixes #407780.
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2007-02-14 Wim Taymans <wim@fluendo.com>
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2007-02-14 10:09:12 +00:00
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Patch by: jp.liu <jp_liu at astrocom dot cn>
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* gst/rtsp/rtspurl.c: (rtsp_url_parse):
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Fix parsing of password field in url. Fixes #407797.
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2007-02-14 12:07:01 +00:00
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2007-02-14 Wim Taymans <wim@fluendo.com>
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gst/wavparse/gstwavparse.*: Update docs.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_chain),
(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
Update docs.
Use boilerplate.
Various code cleanups.
When the bitrate is not known (bps == 0 or compressed formats) let
downstream element guestimate the duration and position and don't
generate timestamps or durations. Fixes #405213.
Fix EOS and ERROR conditions in chain mode, we just need to forward the
error flowreturn upstream.
2007-02-14 09:55:47 +00:00
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* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
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(gst_wavparse_reset), (gst_wavparse_init),
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(gst_wavparse_destroy_sourcepad), (gst_wavparse_fmt),
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(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
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(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
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(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
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(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
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(gst_wavparse_loop), (gst_wavparse_chain),
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(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
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(gst_wavparse_srcpad_event), (gst_wavparse_change_state),
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(plugin_init):
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* gst/wavparse/gstwavparse.h:
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Update docs.
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Use boilerplate.
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Various code cleanups.
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When the bitrate is not known (bps == 0 or compressed formats) let
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downstream element guestimate the duration and position and don't
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generate timestamps or durations. Fixes #405213.
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Fix EOS and ERROR conditions in chain mode, we just need to forward the
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error flowreturn upstream.
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Re-factor the gconfaudiosink into a "GstSwitchSink" base class and a child that implements the GConf key monitoring. ...
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gconf.c: (gst_gconf_get_string),
(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
(gst_gconf_render_bin_with_default):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
(gst_gconf_audio_sink_dispose), (do_change_child),
(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
(cb_change_child), (gst_gconf_audio_sink_change_state):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
(gst_switch_sink_class_init), (gst_switch_sink_reset),
(gst_switch_sink_init), (gst_switch_sink_dispose),
(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
(gst_switch_sink_get_property), (gst_switch_sink_change_state):
* ext/gconf/gstswitchsink.h:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
(gst_auto_audio_sink_detect):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
(gst_auto_video_sink_detect):
Re-factor the gconfaudiosink into a "GstSwitchSink" base class
and a child that implements the GConf key monitoring. The end goal of
this is an audio sink that can be changed on the fly, but at the
moment it still only changes on the next READY transition.
2007-02-13 16:01:29 +00:00
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2007-02-13 Jan Schmidt <thaytan@mad.scientist.com>
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* ext/gconf/Makefile.am:
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* ext/gconf/gconf.c: (gst_gconf_get_string),
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(gst_gconf_get_key_for_sink_profile), (gst_gconf_set_string),
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(gst_gconf_render_bin_with_default):
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* ext/gconf/gconf.h:
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init),
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(gst_gconf_audio_sink_reset), (gst_gconf_audio_sink_init),
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(gst_gconf_audio_sink_dispose), (do_change_child),
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(gst_gconf_switch_profile), (gst_gconf_audio_sink_set_property),
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(cb_change_child), (gst_gconf_audio_sink_change_state):
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* ext/gconf/gstgconfaudiosink.h:
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* ext/gconf/gstswitchsink.c: (gst_switch_sink_base_init),
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(gst_switch_sink_class_init), (gst_switch_sink_reset),
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(gst_switch_sink_init), (gst_switch_sink_dispose),
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(gst_switch_commit_new_kid), (gst_switch_sink_set_child),
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(gst_switch_sink_set_property), (gst_switch_sink_handle_event),
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(gst_switch_sink_get_property), (gst_switch_sink_change_state):
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* ext/gconf/gstswitchsink.h:
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* gst/autodetect/gstautoaudiosink.c:
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(gst_auto_audio_sink_class_init), (gst_auto_audio_sink_dispose),
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(gst_auto_audio_sink_clear_kid), (gst_auto_audio_sink_reset),
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(gst_auto_audio_sink_detect):
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* gst/autodetect/gstautovideosink.c:
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(gst_auto_video_sink_class_init), (gst_auto_video_sink_dispose),
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(gst_auto_video_sink_clear_kid), (gst_auto_video_sink_reset),
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(gst_auto_video_sink_detect):
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Re-factor the gconfaudiosink into a "GstSwitchSink" base class
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and a child that implements the GConf key monitoring. The end goal of
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this is an audio sink that can be changed on the fly, but at the
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moment it still only changes on the next READY transition.
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gst/avi/gstavidemux.c: Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_loop):
Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
2007-02-13 11:57:18 +00:00
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2007-02-13 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
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(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
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(gst_avi_demux_sync), (gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
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(gst_avi_demux_loop):
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Put debug stuff into #ifndef GST_DISABLE_DEBUG #endif
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2007-02-13 09:46:26 +00:00
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2007-02-13 Stefan Kost <ensonic@users.sf.net>
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* configure.ac:
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* docs/plugins/Makefile.am:
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Add crossreferences to glib/gobject/gstream docs.
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2007-02-12 23:35:16 +00:00
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2007-02-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/monoscope/Makefile.am:
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* gst/monoscope/gstmonoscope.c:
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Fix copy'n'paste-o in docs chunk. Also add some missing CFLAGS
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(but no LIBS, since we only use defines from the headers).
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2007-02-12 23:27:31 +00:00
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2007-02-12 Tim-Philipp Müller <tim at centricular dot net>
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Based on patch by: Jonathan Matthew <jonathan at kaolin wh9 net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_stream_init),
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(gst_wavparse_stream_data):
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Fix massive memory leak when operating in streaming mode due to
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GST_BUFFER_MALLOCDATA() not being set on newly-created buffers.
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Fixes #407057.
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gst/avi/gstavidemux.*: Save some memory (8%) by repacking the index entry structure (more to come). Add more FIXMEs t...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Save some memory (8%) by repacking the index entry structure (more to
come). Add more FIXMEs to questionable parts.
2007-02-12 15:29:44 +00:00
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2007-02-12 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
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(gst_avi_demux_reset), (gst_avi_demux_index_entry_for_time),
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(gst_avi_demux_handle_src_query), (gst_avi_demux_parse_superindex),
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(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
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(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
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(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
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(gst_avi_demux_stream_scan), (gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_push_event), (gst_avi_demux_stream_header_pull),
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(gst_avi_demux_do_seek), (gst_avi_demux_process_next_entry),
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(gst_avi_demux_stream_data), (gst_avi_demux_loop):
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* gst/avi/gstavidemux.h:
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Save some memory (8%) by repacking the index entry structure (more to
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come). Add more FIXMEs to questionable parts.
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2007-02-12 12:57:22 +00:00
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2007-02-12 Stefan Kost <ensonic@users.sf.net>
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* sys/v4l2/gstv4l2src.c: (gst_v4l2src_v4l2fourcc_to_caps),
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(gst_v4l2src_get_caps):
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
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(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
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(gst_v4l2src_capture_init):
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More FIXME comments and messaging changes.
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2007-02-12 12:43:00 +00:00
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2007-02-12 Stefan Kost <ensonic@users.sf.net>
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* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
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(gst_goom_change_state):
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* gst/goom/gstgoom.h:
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Improved docs and use GST_DEBUG_FUNCPTR.
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* gst/level/gstlevel.c: (gst_level_class_init):
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Use GST_DEBUG_FUNCPTR.
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* gst/monoscope/gstmonoscope.c: (gst_monoscope_init),
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(gst_monoscope_chain), (gst_monoscope_change_state):
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Improved docs source cleanups.
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2007-02-12 10:29:57 +00:00
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2007-02-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/debug/Makefile.am:
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* gst/debug/gstdebug.c: (plugin_init):
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* gst/debug/gstpushfilesrc.c:
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* gst/debug/gstpushfilesrc.h:
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Add code for a pushfilesrc element that implements a pushfile:// URI
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handler, to make debugging push-mode operation of demuxer/decoders
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that support both easier in connection with seek/playbin/etc.
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The element isn't registered at the moment.
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2007-02-11 12:57:47 +00:00
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2007-02-11 Sébastien Moutte <sebastien@moutte.net>
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* gst/avi/gstavimux.c:
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Comment a #if 0 in caps template definition as VS6 seems to
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do not support it.
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* gst/rtsp/gstrtspsrc.c:(gst_rtspsrc_loop_udp):
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Use gst_guint64_to_gdouble for conversion.
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* gst/rtsp/rtspconnection.c:(rtsp_connection_send):
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Move variables declaration before the first instruction.
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* gst/rtsp/rtspdefs.c:(rtsp_strresult):
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Don't use hstrerror for error log on G_OS_WIN32 build as it's not supported.
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And don't include netdb.h for G_OS_WIN32
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* gst/rtsp/sdpmessage.c:(sdp_parse_line):
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This initialization SDPMedia nmedia = {.media = NULL }; is not supported
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by VS6 then use an other way to initialize SDPMedia structure.
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* gst/udp/gstdynudpsink.h:
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* gst/udp/gstdynudpnetutils.h:
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Do not include <sys/time.h> for G_OS_WIN32
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* gst/udp/gstudpsrc.c:
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Define socklen_t as int for G_OS_WIN32
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* win/common/config.h.in:
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Undef HAVE_NETINET_IN_H
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* win32/vs6/gst_plugins_good.dsw:
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* win32/vs6/libgstrtp.dsp:
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* win32/vs6/libgstrtsp.dsp:
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* win32/vs6/libgstautogen.dsp:
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* win32/vs6/libgstaudiofx.dsp:
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* win32/vs6/libgstudp.dsp:
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Add and update project files.
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* win32/common/gstudp-enumtypes.c:
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* win32/common/gstudp-enumtypes.h:
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Add a copy of udp enumtypes to win32/common as in core
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and base.
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2007-02-11 10:53:21 +00:00
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2007-02-11 Stefan Kost <ensonic@users.sf.net>
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* configure.ac:
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Activate monoscope when building with --enable-experimental. Fix
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--enable-external configure switch description.
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* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init):
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* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose):
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Help gst-indent.
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2007-02-09 09:24:58 +00:00
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2007-02-09 Tim-Philipp Müller <tim at centricular dot net>
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* gst/avi/gstavimux.c: (gst_avi_mux_riff_get_avi_header):
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Explicitly cast result of pointer arithmetic to integer in order to
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avoid compiler warnings on some 64-bit systems. Should fix #406018.
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2007-02-08 11:09:15 +00:00
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2007-02-08 Tim-Philipp Müller <tim at centricular dot net>
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* gst/debug/progressreport.c:
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Some more docs.
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2007-02-07 21:09:45 +00:00
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2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
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* docs/plugins/inspect/plugin-rtp.xml:
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Update for new elements.
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* gst/debug/progressreport.h:
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Commit newly-created header file as well.
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2007-02-07 20:39:16 +00:00
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2007-02-07 Tim-Philipp Müller <tim at centricular dot net>
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|
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* docs/plugins/Makefile.am:
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|
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|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
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|
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|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
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|
* gst/debug/Makefile.am:
|
|
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|
* gst/debug/progressreport.c: (gst_progress_report_post_progress),
|
|
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|
(gst_progress_report_do_query), (gst_progress_report_report):
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Make progressreport element post messages with the current progress
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|
on the bus. Also add some basic docs for it.
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2007-02-07 13:08:34 +00:00
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2007-01-30 Tim-Philipp Müller <tim at centricular dot net>
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|
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* ext/hal/hal.c: (gst_hal_get_string):
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|
* ext/hal/hal.h:
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|
Some small cleanups; deal with errors when parsing the HAL ALSA
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|
capabilities a bit better.
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2007-02-06 16:29:30 +00:00
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2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
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|
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* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
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Let's try this again and use the right cast this time.
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|
2007-02-06 16:24:57 +00:00
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2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
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* gst/smpte/gstsmpte.c: (gst_smpte_transition_type_get_type):
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Add cast to avoid compiler warnings with older GLib versions
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|
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where the nick/name members in GEnumValue are not declared as
|
|
|
|
constant strings.
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|
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|
2007-02-06 15:56:14 +00:00
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2007-02-06 Tim-Philipp Müller <tim at centricular dot net>
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|
|
|
|
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* ext/gconf/gconf.c: (gst_gconf_get_key_for_sink_profile),
|
|
|
|
(gst_gconf_render_bin_from_key),
|
|
|
|
(gst_gconf_get_default_audio_sink):
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|
|
|
* ext/gconf/gconf.h:
|
|
|
|
* ext/gconf/gstgconfaudiosink.c: (get_gconf_key_for_profile),
|
|
|
|
(do_toggle_element), (gst_gconf_audio_sink_set_property),
|
|
|
|
(gst_gconf_audio_sink_get_property):
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|
|
|
In gconfaudiosink, get the right key as the old key in do_toggle
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|
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|
(ie. one dependent on the profile selected). Log some more stuff so
|
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|
|
we can see what's actually going on.
|
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2007-02-06 11:16:49 +00:00
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2007-02-06 Sebastian Dröge <slomo@circular-chaos.org>
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|
|
|
|
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|
* gst/audiofx/audioamplify.c: (gst_audio_amplify_base_init),
|
|
|
|
(gst_audio_amplify_class_init), (gst_audio_amplify_init),
|
|
|
|
(gst_audio_amplify_set_process_function),
|
|
|
|
(gst_audio_amplify_setup):
|
|
|
|
* gst/audiofx/audioamplify.h:
|
|
|
|
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
|
|
|
|
(gst_audio_invert_class_init), (gst_audio_invert_setup):
|
|
|
|
* gst/audiofx/audioinvert.h:
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|
|
|
Some small cleanups and port both elements to the new GstAudioFilter
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|
|
|
base class to save a few lines of common code.
|
|
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|
* gst/audiofx/Makefile.am:
|
|
|
|
Link against libgstaudio for the above changes
|
|
|
|
|
2007-01-29 10:59:48 +00:00
|
|
|
2007-01-29 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* tests/check/elements/.cvsignore:
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|
|
|
Some more ignores.
|
|
|
|
|
2007-01-26 12:21:41 +00:00
|
|
|
2007-01-26 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: charles <charlesg3 at gmail dot com>
|
|
|
|
|
|
|
|
* ext/shout2/gstshout2.c: (gst_shout2send_init),
|
|
|
|
(set_shout_metadata), (gst_shout2send_event):
|
|
|
|
* ext/shout2/gstshout2.h:
|
|
|
|
Properly handle tags in shout2send. Fixes #399825.
|
|
|
|
|
2007-01-25 14:40:15 +00:00
|
|
|
2007-01-25 Wim Taymans <wim@fluendo.com>
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|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
|
|
|
|
(gst_rtspsrc_activate_streams):
|
|
|
|
Convert SDP fields to upper/lowercase following the rules in the SDP to
|
|
|
|
caps document.
|
|
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|
|
2007-01-25 14:22:53 +00:00
|
|
|
2007-01-25 Wim Taymans <wim@fluendo.com>
|
|
|
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|
|
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* gst/rtp/README:
|
|
|
|
* gst/rtp/gstrtpilbcdepay.c:
|
|
|
|
* gst/rtp/gstrtpilbcpay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gpay.c:
|
|
|
|
* gst/rtp/gstrtpspeexdepay.c:
|
|
|
|
* gst/rtp/gstrtpspeexpay.c:
|
|
|
|
* gst/rtp/gstrtpsv3vdepay.c:
|
|
|
|
* gst/rtp/gstrtptheoradepay.c:
|
|
|
|
* gst/rtp/gstrtptheorapay.c:
|
|
|
|
* gst/rtp/gstrtpvorbisdepay.c:
|
|
|
|
* gst/rtp/gstrtpvorbispay.c:
|
|
|
|
Fix case of encoding-name and key/value pairs to match the document.
|
|
|
|
This is to make interoperation with SDP case-insensitive as required by
|
|
|
|
the relevant RFCs.
|
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|
|
|
2007-01-25 11:02:01 +00:00
|
|
|
2007-01-25 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Bump required -core/-base to CVS
|
|
|
|
|
2007-01-25 10:54:19 +00:00
|
|
|
2007-01-25 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_setcaps),
|
|
|
|
(gst_rtp_L16_pay_flush), (gst_rtp_L16_pay_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpL16pay.h:
|
|
|
|
Fill up to MTU using adapter.
|
|
|
|
Timestamp rtp packets.
|
|
|
|
|
2007-01-25 10:36:35 +00:00
|
|
|
2007-01-25 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
|
|
|
|
* sys/ximage/ximageutil.c: (ximageutil_check_xshm_calls):
|
|
|
|
Use G_GSIZE_FORMAT in print statements for portability.
|
|
|
|
Fixes build on macosx.
|
|
|
|
|
gst/rtp/: Port and enable raw audio payloader/depayloader. Needs a bit more work on the payloader side.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
(gst_rtp_L16_depay_plugin_init):
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
(gst_rtp_L16_pay_plugin_init):
* gst/rtp/gstrtpL16pay.h:
Port and enable raw audio payloader/depayloader. Needs a bit more work
on the payloader side.
2007-01-24 18:20:14 +00:00
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|
|
2007-01-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16_depay_base_init),
|
|
|
|
(gst_rtp_L16_depay_class_init), (gst_rtp_L16_depay_init),
|
|
|
|
(gst_rtp_L16_depay_parse_int), (gst_rtp_L16_depay_setcaps),
|
|
|
|
(gst_rtp_L16_depay_process), (gst_rtp_L16_depay_set_property),
|
|
|
|
(gst_rtp_L16_depay_get_property), (gst_rtp_L16_depay_change_state),
|
|
|
|
(gst_rtp_L16_depay_plugin_init):
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|
|
|
* gst/rtp/gstrtpL16depay.h:
|
|
|
|
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_get_type),
|
|
|
|
(gst_rtp_L16_pay_base_init), (gst_rtp_L16_pay_class_init),
|
|
|
|
(gst_rtp_L16_pay_init), (gst_rtp_L16_pay_finalize),
|
|
|
|
(gst_rtp_L16_pay_setcaps), (gst_rtp_L16_pay_handle_buffer),
|
|
|
|
(gst_rtp_L16_pay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpL16pay.h:
|
|
|
|
Port and enable raw audio payloader/depayloader. Needs a bit more work
|
|
|
|
on the payloader side.
|
|
|
|
|
2007-01-24 16:25:55 +00:00
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|
2007-01-24 Wim Taymans <wim@fluendo.com>
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|
|
|
|
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|
* gst/rtsp/gstrtspsrc.c: (pad_blocked),
|
|
|
|
(gst_rtspsrc_stream_configure_transport),
|
|
|
|
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_udp):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Only unblock the udp pads when we linked and activated them all.
|
|
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|
Fixes #395688.
|
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|
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|
2007-01-24 15:18:34 +00:00
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|
|
2007-01-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpac3depay.c: (gst_rtp_ac3_depay_base_init),
|
|
|
|
(gst_rtp_ac3_depay_class_init), (gst_rtp_ac3_depay_init),
|
|
|
|
(gst_rtp_ac3_depay_setcaps), (gst_rtp_ac3_depay_process),
|
|
|
|
(gst_rtp_ac3_depay_set_property), (gst_rtp_ac3_depay_get_property),
|
|
|
|
(gst_rtp_ac3_depay_change_state), (gst_rtp_ac3_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpac3depay.h:
|
|
|
|
Added simple AC3 depayloader (RFC 4184).
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps):
|
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|
|
Fix a leak.
|
|
|
|
|
gst/audiofx/: Add new element "audioamplify". This allows scaling of raw audio samples, similar to the "volume" eleme...
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audioamplify.c:
(gst_audio_amplify_clipping_method_get_type),
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
(gst_audio_amplify_set_caps),
(gst_audio_amplify_transform_int_clip),
(gst_audio_amplify_transform_int_wrap_negative),
(gst_audio_amplify_transform_int_wrap_positive),
(gst_audio_amplify_transform_float_clip),
(gst_audio_amplify_transform_float_wrap_negative),
(gst_audio_amplify_transform_float_wrap_positive),
(gst_audio_amplify_transform_ip):
* gst/audiofx/audioamplify.h:
* gst/audiofx/audiofx.c: (plugin_init):
Add new element "audioamplify". This allows scaling of raw audio
samples, similar to the "volume" element, but provides different modes
for clipping and allows unlimited amplification. It's mainly targeted
for creative sound design and not as a replacement of the "volume"
element. Fixes #397162
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for audioamplify and integrate them into the build system
* tests/check/Makefile.am:
* tests/check/elements/audioamplify.c: (setup_amplify),
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
Add fairly extensive unit test suite for audioamplify
2007-01-24 12:41:03 +00:00
|
|
|
2007-01-24 Sebastian Dröge <slomo@circular-chaos.org>
|
|
|
|
|
|
|
|
reviewed by: Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/audiofx/Makefile.am:
|
|
|
|
* gst/audiofx/audioamplify.c:
|
|
|
|
(gst_audio_amplify_clipping_method_get_type),
|
|
|
|
(gst_audio_amplify_base_init), (gst_audio_amplify_class_init),
|
|
|
|
(gst_audio_amplify_init), (gst_audio_amplify_set_process_function),
|
|
|
|
(gst_audio_amplify_set_property), (gst_audio_amplify_get_property),
|
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|
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(gst_audio_amplify_set_caps),
|
|
|
|
(gst_audio_amplify_transform_int_clip),
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|
|
(gst_audio_amplify_transform_int_wrap_negative),
|
|
|
|
(gst_audio_amplify_transform_int_wrap_positive),
|
|
|
|
(gst_audio_amplify_transform_float_clip),
|
|
|
|
(gst_audio_amplify_transform_float_wrap_negative),
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|
|
(gst_audio_amplify_transform_float_wrap_positive),
|
|
|
|
(gst_audio_amplify_transform_ip):
|
|
|
|
* gst/audiofx/audioamplify.h:
|
|
|
|
* gst/audiofx/audiofx.c: (plugin_init):
|
|
|
|
Add new element "audioamplify". This allows scaling of raw audio
|
|
|
|
samples, similar to the "volume" element, but provides different modes
|
|
|
|
for clipping and allows unlimited amplification. It's mainly targeted
|
|
|
|
for creative sound design and not as a replacement of the "volume"
|
|
|
|
element. Fixes #397162
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
* docs/plugins/inspect/plugin-audiofx.xml:
|
|
|
|
Add docs for audioamplify and integrate them into the build system
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
* tests/check/elements/audioamplify.c: (setup_amplify),
|
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|
|
(cleanup_amplify), (GST_START_TEST), (amplify_suite), (main):
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|
Add fairly extensive unit test suite for audioamplify
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|
|
|
2007-01-24 12:26:41 +00:00
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|
2007-01-24 Wim Taymans <wim@fluendo.com>
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|
|
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|
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|
* gst/rtsp/gstrtspsrc.c: (pad_unblocked), (pad_blocked):
|
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Unblock pads after adding the pads to the element so that autopluggers
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|
|
get a change to link something. Possibly fixes #395688.
|
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|
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|
2007-01-24 12:22:51 +00:00
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|
2007-01-24 Wim Taymans <wim@fluendo.com>
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|
|
|
* gst/rtp/gstrtpamrdepay.c:
|
|
|
|
* gst/rtp/gstrtpgsmdepay.c:
|
|
|
|
* gst/rtp/gstrtph263pdepay.c:
|
|
|
|
* gst/rtp/gstrtph263ppay.c:
|
|
|
|
* gst/rtp/gstrtph264depay.c:
|
|
|
|
* gst/rtp/gstrtpilbcdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp2tdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gpay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c:
|
|
|
|
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init),
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|
|
(gst_rtp_mpa_depay_init), (gst_rtp_mpa_depay_setcaps),
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|
|
(gst_rtp_mpa_depay_process):
|
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|
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_class_init),
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|
|
(gst_rtp_mpv_depay_init), (gst_rtp_mpv_depay_process):
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|
* gst/rtp/gstrtppcmadepay.c:
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|
|
* gst/rtp/gstrtppcmudepay.c:
|
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* gst/rtp/gstrtpspeexdepay.c:
|
|
|
|
* gst/rtp/gstrtpspeexpay.c:
|
|
|
|
* gst/rtp/gstrtpsv3vdepay.c:
|
|
|
|
* gst/rtp/gstrtptheoradepay.c:
|
|
|
|
* gst/rtp/gstrtptheorapay.c:
|
|
|
|
* gst/rtp/gstrtpvorbisdepay.c:
|
|
|
|
* gst/rtp/gstrtpvorbispay.c:
|
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|
Fix caps with payload numbers.
|
|
|
|
Add some fixed payload numbers to caps when possible.
|
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|
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gst/audiofx/: Add new audiofx element "audioinvert". This element swaps the upper and lower half of samples and can b...
Original commit message from CVS:
reviewed by: Stefan Kost <ensonic@users.sf.net>
* gst/audiofx/Makefile.am:
* gst/audiofx/audiofx.c: (plugin_init):
* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
(gst_audio_invert_class_init), (gst_audio_invert_init),
(gst_audio_invert_set_property), (gst_audio_invert_get_property),
(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
(gst_audio_invert_transform_float),
(gst_audio_invert_transform_ip):
* gst/audiofx/audioinvert.h:
Add new audiofx element "audioinvert". This element swaps the upper
and lower half of samples and can be used for example for a
wide-stereo effect. Fixes #396057
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.args:
* docs/plugins/inspect/plugin-audiofx.xml:
Add docs for the audioinvert element and add them to the build system.
* tests/check/Makefile.am:
* tests/check/elements/audioinvert.c: (setup_invert),
(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
Add unit test suite for the audioinvert element.
2007-01-23 18:16:09 +00:00
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2007-01-23 Sebastian Dröge <slomo@circular-chaos.org>
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reviewed by: Stefan Kost <ensonic@users.sf.net>
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* gst/audiofx/Makefile.am:
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* gst/audiofx/audiofx.c: (plugin_init):
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* gst/audiofx/audioinvert.c: (gst_audio_invert_base_init),
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(gst_audio_invert_class_init), (gst_audio_invert_init),
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(gst_audio_invert_set_property), (gst_audio_invert_get_property),
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(gst_audio_invert_set_caps), (gst_audio_invert_transform_int),
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(gst_audio_invert_transform_float),
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(gst_audio_invert_transform_ip):
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* gst/audiofx/audioinvert.h:
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Add new audiofx element "audioinvert". This element swaps the upper
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and lower half of samples and can be used for example for a
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wide-stereo effect. Fixes #396057
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/inspect/plugin-audiofx.xml:
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Add docs for the audioinvert element and add them to the build system.
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* tests/check/Makefile.am:
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* tests/check/elements/audioinvert.c: (setup_invert),
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(cleanup_invert), (GST_START_TEST), (invert_suite), (main):
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Add unit test suite for the audioinvert element.
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2007-01-23 17:36:32 +00:00
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2007-01-23 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_parse_int),
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(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process):
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Parse config params as string and int.
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Parse and use AU header length
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2007-01-23 17:27:39 +00:00
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2007-01-23 Wim Taymans <wim@fluendo.com>
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* gst/smpte/barboxwipes.c: (gst_wipe_boxes_draw),
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(gst_wipe_triangles_clock_draw), (gst_wipe_triangles_draw):
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* gst/smpte/gstmask.c: (_gst_mask_register):
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* gst/smpte/gstmask.h:
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* gst/smpte/gstsmpte.c: (gst_smpte_update_mask):
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* gst/smpte/paint.c: (gst_smpte_paint_hbox), (draw_bresenham_line),
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(gst_smpte_paint_triangle_clock):
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constify some static structs.
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Don't update the mask if nothing changed to the params.
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Make sure we never draw outside of the picture. Fixes #398325.
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2007-01-22 13:06:43 +00:00
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2007-01-22 Tim-Philipp Müller <tim at centricular dot net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header_pull):
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Error out properly when pull_range fails while we're reading the
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headers, instead of just pausing the task silently. Fixes #399338.
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2007-01-19 13:06:07 +00:00
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2007-01-19 Tim-Philipp Müller <tim at centricular dot net>
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* gst/smpte/gstsmpte.c: (gst_smpte_collected):
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Some more sanity checks to make sure the input formats match and the
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input pads are actually negotiated, in case someone tries to feed
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buffers from fakesrc or filesrc. Fixes #398299.
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Also const-ify an array, just because we can.
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2007-01-19 10:35:13 +00:00
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2007-01-19 Edward Hervey <edward@fluendo.com>
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* gst/smpte/gstsmpte.c: (fill_i420), (gst_smpte_collected):
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Ignore previous commit, that was only valid for widths and heights
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that are multiples of 4.
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Copy over size/stride macros from jpegdec. This allows the element
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to work with any width,height...
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... but puts in evidence that the actual transformations only work
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with width/height that are multiples of 4.
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2007-01-19 09:48:47 +00:00
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2007-01-19 Edward Hervey <edward@fluendo.com>
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* gst/smpte/gstsmpte.c: (gst_smpte_collected):
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Allocate buffers of the right size.
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The proper size of a I420 buffer in bytes is:
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width * height * 3
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------------------
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2
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2007-01-18 18:37:39 +00:00
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2007-01-18 Tim-Philipp Müller <tim at centricular dot net>
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* gst/smpte/gstsmpte.c: (gst_smpte_init):
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Proxy getcaps on sink pads too, so that we either end up with the
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same dimensions on all pads or error out if that's not possible
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(seems to work even!). Fixes #398086, I think.
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2007-01-18 11:29:17 +00:00
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2007-01-18 Tim-Philipp Müller <tim at centricular dot net>
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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Remove ladspa from docs; add hierarchy info for GstAudioPanorama;
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fix integer properties with -1 as minimum value.
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* docs/plugins/inspect/plugin-1394.xml:
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* docs/plugins/inspect/plugin-aasink.xml:
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* docs/plugins/inspect/plugin-alaw.xml:
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* docs/plugins/inspect/plugin-alpha.xml:
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* docs/plugins/inspect/plugin-alphacolor.xml:
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* docs/plugins/inspect/plugin-annodex.xml:
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* docs/plugins/inspect/plugin-apetag.xml:
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* docs/plugins/inspect/plugin-audiofx.xml:
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* docs/plugins/inspect/plugin-auparse.xml:
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* docs/plugins/inspect/plugin-autodetect.xml:
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* docs/plugins/inspect/plugin-avi.xml:
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* docs/plugins/inspect/plugin-cacasink.xml:
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* docs/plugins/inspect/plugin-cairo.xml:
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* docs/plugins/inspect/plugin-cdio.xml:
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* docs/plugins/inspect/plugin-cutter.xml:
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* docs/plugins/inspect/plugin-debug.xml:
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* docs/plugins/inspect/plugin-dv.xml:
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* docs/plugins/inspect/plugin-efence.xml:
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* docs/plugins/inspect/plugin-effectv.xml:
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* docs/plugins/inspect/plugin-esdsink.xml:
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* docs/plugins/inspect/plugin-flac.xml:
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* docs/plugins/inspect/plugin-flxdec.xml:
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* docs/plugins/inspect/plugin-gconfelements.xml:
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* docs/plugins/inspect/plugin-gdkpixbuf.xml:
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* docs/plugins/inspect/plugin-goom.xml:
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* docs/plugins/inspect/plugin-halelements.xml:
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* docs/plugins/inspect/plugin-icydemux.xml:
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* docs/plugins/inspect/plugin-id3demux.xml:
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* docs/plugins/inspect/plugin-jpeg.xml:
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* docs/plugins/inspect/plugin-level.xml:
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* docs/plugins/inspect/plugin-matroska.xml:
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* docs/plugins/inspect/plugin-mulaw.xml:
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* docs/plugins/inspect/plugin-multipart.xml:
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* docs/plugins/inspect/plugin-navigationtest.xml:
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* docs/plugins/inspect/plugin-ossaudio.xml:
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* docs/plugins/inspect/plugin-png.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-rtsp.xml:
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* docs/plugins/inspect/plugin-shout2send.xml:
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* docs/plugins/inspect/plugin-smpte.xml:
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* docs/plugins/inspect/plugin-speex.xml:
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* docs/plugins/inspect/plugin-taglib.xml:
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* docs/plugins/inspect/plugin-udp.xml:
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* docs/plugins/inspect/plugin-videobalance.xml:
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* docs/plugins/inspect/plugin-videobox.xml:
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* docs/plugins/inspect/plugin-videoflip.xml:
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* docs/plugins/inspect/plugin-videomixer.xml:
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* docs/plugins/inspect/plugin-wavenc.xml:
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* docs/plugins/inspect/plugin-wavparse.xml:
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* docs/plugins/inspect/plugin-ximagesrc.xml:
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Update to CVS.
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2007-01-18 11:23:36 +00:00
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2007-01-18 Stefan Kost <ensonic@users.sf.net>
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2007-02-12 12:43:00 +00:00
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Patch by: Sebastian Dröge <slomo circular-chaos org>
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2007-01-18 11:23:36 +00:00
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* gst/audiofx/audiopanorama.c:
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Fix doc section name (Fixes #397946)
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sys/v4l2/: Fix EIO handing when capturing. Add new property to specify the number of buffers to enque (and remove the...
Original commit message from CVS:
* sys/v4l2/gstv4l2object.c:
(gst_v4l2_object_install_properties_helper),
(gst_v4l2_object_set_property_helper),
(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
* sys/v4l2/gstv4l2object.h:
* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
(gst_v4l2src_init), (gst_v4l2src_set_property),
(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
(gst_v4l2src_capture_deinit):
Fix EIO handing when capturing. Add new property to specify the number of
buffers to enque (and remove the borked num-buffers usage).
2007-01-17 14:30:50 +00:00
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2007-01-17 Stefan Kost <ensonic@users.sf.net>
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* sys/v4l2/gstv4l2object.c:
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(gst_v4l2_object_install_properties_helper),
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(gst_v4l2_object_set_property_helper),
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(gst_v4l2_object_get_property_helper), (gst_v4l2_set_defaults):
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* sys/v4l2/gstv4l2object.h:
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* sys/v4l2/gstv4l2src.c: (gst_v4l2src_class_init),
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(gst_v4l2src_init), (gst_v4l2src_set_property),
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(gst_v4l2src_get_property), (gst_v4l2src_set_caps):
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
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(gst_v4l2src_grab_frame), (gst_v4l2src_set_capture),
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(gst_v4l2src_capture_init), (gst_v4l2src_capture_start),
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(gst_v4l2src_capture_deinit):
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Fix EIO handing when capturing. Add new property to specify the number of
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buffers to enque (and remove the borked num-buffers usage).
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2007-01-16 08:29:11 +00:00
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2007-01-16 Stefan Kost <ensonic@users.sf.net>
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Patch by: Sebastian Dröge <slomo circular-chaos org>
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* gst/audiofx/audiopanorama.c: (gst_audio_panorama_class_init),
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(gst_audio_panorama_set_process_function):
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Use a function array for process methods, add more docs and define the
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startindex of enums.
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Add support for more than one audio stream; write better AVIX header; refactor code a bit; don't announce vorbis caps...
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
* tests/check/elements/avimux.c: (teardown_src_pad):
Add support for more than one audio stream; write better AVIX
header; refactor code a bit; don't announce vorbis caps on our audio
sink pads since we don't support it anyway. Closes #379298.
2007-01-14 17:55:33 +00:00
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2007-01-14 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/avi/gstavimux.c: (gst_avi_mux_finalize),
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(gst_avi_mux_pad_reset), (gst_avi_mux_reset), (gst_avi_mux_init),
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(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
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(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
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(gst_avi_mux_riff_get_avi_header),
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(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_riff_get_header),
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(gst_avi_mux_write_avix_index), (gst_avi_mux_add_index),
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(gst_avi_mux_bigfile), (gst_avi_mux_start_file),
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(gst_avi_mux_stop_file), (gst_avi_mux_handle_event),
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(gst_avi_mux_do_buffer), (gst_avi_mux_do_one_buffer),
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(gst_avi_mux_change_state):
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* gst/avi/gstavimux.h:
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* tests/check/elements/avimux.c: (teardown_src_pad):
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Add support for more than one audio stream; write better AVIX
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header; refactor code a bit; don't announce vorbis caps on our audio
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sink pads since we don't support it anyway. Closes #379298.
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gst/audiofx/audiopanorama.*: Add 'method' property and provide a simple (non-psychoacustic) processing method (#394859).
Original commit message from CVS:
Patch by: Sebastian Dröge <slomo circular-chaos org>
* gst/audiofx/audiopanorama.c:
(gst_audio_panorama_method_get_type),
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
(gst_audio_panorama_set_process_function),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s_int_simple),
(gst_audio_panorama_transform_s2s_int_simple),
(gst_audio_panorama_transform_m2s_float_simple),
(gst_audio_panorama_transform_s2s_float_simple):
* gst/audiofx/audiopanorama.h:
Add 'method' property and provide a simple (non-psychoacustic)
processing method (#394859).
* tests/check/elements/audiopanorama.c: (GST_START_TEST),
(panorama_suite):
Tests for new method.
2007-01-13 15:52:18 +00:00
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2007-01-13 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Sebastian Dröge <slomo circular-chaos org>
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* gst/audiofx/audiopanorama.c:
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(gst_audio_panorama_method_get_type),
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(gst_audio_panorama_class_init), (gst_audio_panorama_init),
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(gst_audio_panorama_set_process_function),
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(gst_audio_panorama_set_property),
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(gst_audio_panorama_get_property), (gst_audio_panorama_set_caps),
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(gst_audio_panorama_transform_m2s_int_simple),
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(gst_audio_panorama_transform_s2s_int_simple),
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(gst_audio_panorama_transform_m2s_float_simple),
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(gst_audio_panorama_transform_s2s_float_simple):
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* gst/audiofx/audiopanorama.h:
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Add 'method' property and provide a simple (non-psychoacustic)
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processing method (#394859).
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* tests/check/elements/audiopanorama.c: (GST_START_TEST),
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(panorama_suite):
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|
Tests for new method.
|
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|
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|
2007-01-11 16:59:40 +00:00
|
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|
2007-01-11 Tim-Philipp Müller <tim at centricular dot net>
|
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_read_range):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_read_range):
|
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|
Set correct caps on outgoing pulled buffers, or things blow up
|
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|
after recent core changes.
|
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|
2007-01-11 11:05:04 +00:00
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2007-01-11 Wim Taymans <wim@fluendo.com>
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Based on patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
|
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* gst/multipart/multipartmux.c: (gst_multipart_mux_init),
|
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|
(gst_multipart_mux_request_new_pad),
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(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
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(gst_multipart_mux_change_state):
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Return FLOW errors ASAP. Fixes #394977.
|
|
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|
Misc cleanups.
|
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|
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|
2007-01-11 09:30:59 +00:00
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2007-01-11 Wim Taymans <wim@fluendo.com>
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Patch by: Lutz Mueller <lutz at topfrose dot de>
|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_activate_streams):
|
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|
Check for stream pad before activating.
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gst/rtsp/: Allow url to be NULL to be able to use it for server connections.
Original commit message from CVS:
Patch by: Peter Kjellerstedt <pkj at axis com>
* gst/rtsp/COPYING.MIT:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
(gst_rtspsrc_open), (gst_rtspsrc_close):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_send), (read_line),
(parse_request_line), (parse_line), (rtsp_connection_read),
(rtsp_connection_close):
* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
(rtsp_method_as_text), (rtsp_header_as_text),
(rtsp_status_as_text), (rtsp_find_header_field),
(rtsp_find_method):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
(rtsp_ext_wms_configure_stream):
* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
(rtsp_message_new_request), (rtsp_message_init_request),
(rtsp_message_new_response), (rtsp_message_init_response),
(rtsp_message_init_data), (rtsp_message_unset),
(rtsp_message_free), (rtsp_message_add_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
(sdp_message_dump):
Allow url to be NULL to be able to use it for server connections.
Can now send responses as well as requests.
No longer hangs in an endless loop if EOF is received.
Can now convert a status code to a text string.
Return RTSP_HDR_INVALID for unknown headers.
Return RTSP_INVALID for unknown methods.
Copy CSeq and Session headers from the request.
Only free memory corresponding to the currently set message type.
Added const to function arguments as appropriate.
Avoid a compiler warning when initializing nmedia.
Use guint rather than gint to avoid compiler warnings.
Fix crasher in wms extension.
Factor out stream setup from open_connection.
Delay activation of streams when actual data is received from the
server, this prepares us to do proper protocol switching.
Added new license.
Fixes #380895.
2007-01-10 15:19:48 +00:00
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2007-01-10 Wim Taymans <wim@fluendo.com>
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Patch by: Peter Kjellerstedt <pkj at axis com>
|
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|
|
|
|
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* gst/rtsp/COPYING.MIT:
|
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|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
|
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|
(gst_rtspsrc_stream_free), (gst_rtspsrc_cleanup),
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|
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(gst_rtspsrc_alloc_udp_ports), (pad_unblocked), (pad_blocked),
|
|
|
|
(gst_rtspsrc_stream_configure_transport),
|
|
|
|
(gst_rtspsrc_activate_streams), (gst_rtspsrc_loop_interleaved),
|
|
|
|
(gst_rtspsrc_loop_udp), (gst_rtspsrc_send),
|
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|
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(gst_rtspsrc_parse_methods),
|
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(gst_rtspsrc_create_transports_string),
|
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|
|
(gst_rtspsrc_prepare_transports), (gst_rtspsrc_setup_streams),
|
|
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(gst_rtspsrc_open), (gst_rtspsrc_close):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
|
|
|
|
(rtsp_connection_connect), (rtsp_connection_send), (read_line),
|
|
|
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(parse_request_line), (parse_line), (rtsp_connection_read),
|
|
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|
(rtsp_connection_close):
|
|
|
|
* gst/rtsp/rtspdefs.c: (rtsp_init_status), (rtsp_strresult),
|
|
|
|
(rtsp_method_as_text), (rtsp_header_as_text),
|
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|
|
(rtsp_status_as_text), (rtsp_find_header_field),
|
|
|
|
(rtsp_find_method):
|
|
|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_after_send),
|
|
|
|
(rtsp_ext_wms_configure_stream):
|
|
|
|
* gst/rtsp/rtspmessage.c: (rtsp_message_new), (rtsp_message_init),
|
|
|
|
(rtsp_message_new_request), (rtsp_message_init_request),
|
|
|
|
(rtsp_message_new_response), (rtsp_message_init_response),
|
|
|
|
(rtsp_message_init_data), (rtsp_message_unset),
|
|
|
|
(rtsp_message_free), (rtsp_message_add_header),
|
|
|
|
(rtsp_message_get_header), (rtsp_message_set_body),
|
|
|
|
(rtsp_message_get_body), (dump_mem), (rtsp_message_dump):
|
|
|
|
* gst/rtsp/rtspmessage.h:
|
|
|
|
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
|
|
|
|
(sdp_media_get_attribute_val_n), (read_string), (read_string_del),
|
|
|
|
(sdp_parse_line), (sdp_message_parse_buffer), (print_media),
|
|
|
|
(sdp_message_dump):
|
|
|
|
Allow url to be NULL to be able to use it for server connections.
|
|
|
|
Can now send responses as well as requests.
|
|
|
|
No longer hangs in an endless loop if EOF is received.
|
|
|
|
Can now convert a status code to a text string.
|
|
|
|
Return RTSP_HDR_INVALID for unknown headers.
|
|
|
|
Return RTSP_INVALID for unknown methods.
|
|
|
|
Copy CSeq and Session headers from the request.
|
|
|
|
Only free memory corresponding to the currently set message type.
|
|
|
|
Added const to function arguments as appropriate.
|
|
|
|
Avoid a compiler warning when initializing nmedia.
|
|
|
|
Use guint rather than gint to avoid compiler warnings.
|
|
|
|
Fix crasher in wms extension.
|
|
|
|
Factor out stream setup from open_connection.
|
|
|
|
Delay activation of streams when actual data is received from the
|
|
|
|
server, this prepares us to do proper protocol switching.
|
|
|
|
Added new license.
|
|
|
|
Fixes #380895.
|
|
|
|
|
|
|
|
|
2007-01-10 09:47:43 +00:00
|
|
|
2007-01-10 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Sebastian Dröge <slomo ubuntu com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* gst/audiofx/audiopanorama.c:
|
|
|
|
Some small docs fixes (#394851).
|
|
|
|
|
2007-01-09 12:25:26 +00:00
|
|
|
2007-01-09 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c:
|
|
|
|
Fix docs.
|
|
|
|
|
gst/rtp/: Added RFC 2250 MPEG Video Depayloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
* gst/rtp/gstrtpmpvdepay.h:
Added RFC 2250 MPEG Video Depayloader.
* gst/rtp/gstrtpL16depay.h:
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
(gst_rtp_h263p_depay_process):
Fix Header file. Small cleanups.
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
(gst_rtp_mp4v_depay_change_state):
Remove usused code. Remove Adapter from state Change. Added debug.
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
* gst/rtp/gstrtpmpadepay.h:
Subclass base depayloader.
Added debug.
Support static payload type assignment as well.
* gst/rtp/gstrtpmpapay.c:
Fix caps.
2007-01-09 12:23:48 +00:00
|
|
|
2007-01-09 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpmpvdepay.c: (gst_rtp_mpv_depay_base_init),
|
|
|
|
(gst_rtp_mpv_depay_class_init), (gst_rtp_mpv_depay_init),
|
|
|
|
(gst_rtp_mpv_depay_setcaps), (gst_rtp_mpv_depay_process),
|
|
|
|
(gst_rtp_mpv_depay_set_property), (gst_rtp_mpv_depay_get_property),
|
|
|
|
(gst_rtp_mpv_depay_change_state), (gst_rtp_mpv_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmpvdepay.h:
|
|
|
|
Added RFC 2250 MPEG Video Depayloader.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpL16depay.h:
|
|
|
|
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_setcaps),
|
|
|
|
(gst_rtp_h263p_depay_process):
|
|
|
|
Fix Header file. Small cleanups.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init),
|
|
|
|
(gst_rtp_mp4g_depay_init), (gst_rtp_mp4g_depay_finalize),
|
|
|
|
(gst_rtp_mp4g_depay_process), (gst_rtp_mp4g_depay_change_state):
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init),
|
|
|
|
(gst_rtp_mp4v_depay_init), (gst_rtp_mp4v_depay_finalize),
|
|
|
|
(gst_rtp_mp4v_depay_setcaps), (gst_rtp_mp4v_depay_process),
|
|
|
|
(gst_rtp_mp4v_depay_change_state):
|
|
|
|
Remove usused code. Remove Adapter from state Change. Added debug.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_base_init),
|
|
|
|
(gst_rtp_mpa_depay_class_init), (gst_rtp_mpa_depay_init),
|
|
|
|
(gst_rtp_mpa_depay_setcaps), (gst_rtp_mpa_depay_process):
|
|
|
|
* gst/rtp/gstrtpmpadepay.h:
|
|
|
|
Subclass base depayloader.
|
|
|
|
Added debug.
|
|
|
|
Support static payload type assignment as well.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmpapay.c:
|
|
|
|
Fix caps.
|
|
|
|
|
2007-01-08 12:45:10 +00:00
|
|
|
2007-01-08 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Vincent Torri <vtorri at univ-evry fr>
|
|
|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c:
|
|
|
|
* ext/jpeg/gstjpegenc.c:
|
|
|
|
* ext/jpeg/smokecodec.c:
|
|
|
|
These libjpeg callbacks should return a 'boolean' (unsigned char
|
|
|
|
apparently) and not a 'gboolean' (which maps to gint). Fixes
|
|
|
|
warnings when compiling with MingW (#393427).
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
|
|
|
|
Use ioctlsocket on win32.
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
|
|
|
|
Some printf format fixes for win32.
|
|
|
|
|
2007-01-07 10:44:12 +00:00
|
|
|
2007-01-07 Sébastien Moutte <sebastien@moutte.net>
|
|
|
|
|
|
|
|
* gst/cutter/gstcutter.c: (gst_cutter_chain):
|
|
|
|
Use gst_guint64_to_gdouble for conversion.
|
|
|
|
* win32/vs6/libgstmatroska.dsp:
|
|
|
|
Add zlib to the link.
|
|
|
|
* win32/vs6/libgstvideobox.dsp:
|
|
|
|
Update liboil library name (project is linked to
|
|
|
|
liboil-0.3-0.lib now).
|
|
|
|
|
2007-01-05 17:23:04 +00:00
|
|
|
2007-01-05 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/Makefile.am:
|
|
|
|
If zlib is available and used, we must link it explicitly for
|
|
|
|
things to work on MingW (fixes #392855).
|
|
|
|
|
2007-01-04 11:02:29 +00:00
|
|
|
2007-01-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_delay):
|
|
|
|
Don't return bogus values when esd_get_delay() fails for some
|
|
|
|
reason (#392189).
|
|
|
|
|
2006-12-24 11:36:31 +00:00
|
|
|
2006-12-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/ximage/gstximagesrc.c: (composite_pixel):
|
|
|
|
Fix presumably copy'n'pasto for 16bpp depth.
|
|
|
|
|
2006-12-24 11:24:59 +00:00
|
|
|
2006-12-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c:
|
|
|
|
(gst_matroska_mux_audio_pad_setcaps):
|
|
|
|
The "signed" field in audio caps is of boolean type, trying to use
|
|
|
|
gst_structure_get_int() to extract it will fail. Fixing this makes
|
|
|
|
matroskamux accept raw audio input (#387121) (use at your own risk
|
|
|
|
though, due to the matroska spec being not entirely useful in this
|
|
|
|
respect).
|
|
|
|
Also fix up raw audio structures in template caps so that they
|
|
|
|
represent what our setcaps function will actually accept, so that
|
|
|
|
converters know what to convert to.
|
|
|
|
Finally, don't fail if there isn't an "endianness" field in 8-bit
|
|
|
|
PCM caps.
|
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|
|
2006-12-22 10:15:23 +00:00
|
|
|
2006-12-22 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* tests/check/elements/audiopanorama.c: (cleanup_panorama):
|
|
|
|
* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
|
|
|
|
* tests/check/elements/cmmldec.c: (setup_cmmldec),
|
|
|
|
(teardown_cmmldec):
|
|
|
|
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
|
|
|
|
(teardown_cmmlenc):
|
|
|
|
* tests/check/elements/level.c: (setup_level), (cleanup_level):
|
|
|
|
reapply consistent pad (de)activation
|
|
|
|
|
2006-12-21 17:03:39 +00:00
|
|
|
2006-12-21 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Back to CVS
|
|
|
|
|
|
|
|
* gst-plugins-good.doap:
|
|
|
|
Add 0.10.5 doap entry
|
|
|
|
|
2006-12-21 15:40:55 +00:00
|
|
|
=== release 0.10.5 ===
|
|
|
|
|
|
|
|
2006-12-21 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.10.5, "The Path of Thorns"
|
|
|
|
|
2006-12-21 12:48:32 +00:00
|
|
|
2006-12-21 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* tests/check/elements/audiopanorama.c: (cleanup_panorama):
|
|
|
|
* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
|
|
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* tests/check/elements/cmmldec.c: (setup_cmmldec),
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(teardown_cmmldec):
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* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
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(teardown_cmmlenc):
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* tests/check/elements/level.c: (setup_level), (cleanup_level):
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revert my freeze breakage
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2006-12-21 08:15:23 +00:00
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2006-12-21 Stefan Kost <ensonic@users.sf.net>
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* tests/check/elements/audiopanorama.c: (cleanup_panorama):
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* tests/check/elements/avimux.c: (setup_avimux), (cleanup_avimux):
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* tests/check/elements/cmmldec.c: (setup_cmmldec),
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(teardown_cmmldec):
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* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
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(teardown_cmmlenc):
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* tests/check/elements/level.c: (setup_level), (cleanup_level):
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consistent pad (de)activation
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2006-12-18 16:46:17 +00:00
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2006-12-18 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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* ext/Makefile.am:
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Disable LADPSA, as it has moved to the -bad module for the duration.
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2006-12-18 15:51:54 +00:00
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2006-12-18 Wim Taymans <wim@fluendo.com>
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* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
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(gst_signal_processor_event):
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Reset flow_state back to _OK after a flush stop so that we exit our
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error state after the flush. Fixes #374213
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2006-12-17 06:11:39 +00:00
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2006-12-16 David Schleef <ds@schleef.org>
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* sys/osxvideo/osxvideosink.h:
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* sys/osxvideo/osxvideosink.m:
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Decent effort at porting to 0.10. Needs cleanup on OS/X.
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2006-12-17 05:07:07 +00:00
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2006-12-16 David Schleef <ds@schleef.org>
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Patch by: Vijay Santhanam <vijay santhanam gmail com>
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* sys/osxvideo/Makefile.am:
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* sys/osxvideo/osxvideosink.h:
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* sys/osxvideo/osxvideosink.m:
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Preliminary patch for porting osxvideosink
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gst/videomixer/videomixer.c: Introduce some locking around the videomixer state so that it does not crash when adding...
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
(gst_videomixer_reset), (gst_videomixer_init),
(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
(gst_videomixer_release_pad), (gst_videomixer_collected),
(gst_videomixer_change_state):
Introduce some locking around the videomixer state so that it does not
crash when adding/removing pads. Fixes #383043.
2006-12-16 16:21:26 +00:00
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2006-12-16 Wim Taymans <wim@fluendo.com>
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Patch by: Sjoerd Simons <sjoerd at luon dot net>
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* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
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(gst_videomixer_set_master_geometry),
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(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free),
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(gst_videomixer_reset), (gst_videomixer_init),
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(gst_videomixer_finalize), (gst_videomixer_request_new_pad),
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(gst_videomixer_release_pad), (gst_videomixer_collected),
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(gst_videomixer_change_state):
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Introduce some locking around the videomixer state so that it does not
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crash when adding/removing pads. Fixes #383043.
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2006-12-16 11:42:56 +00:00
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2006-12-16 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Make sure libcaca can actually be used instead of just checking for
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/usr/bin/caca-config, so we don't wrongly try to build cacasink when
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cross-compiling (fixes #384587).
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2006-12-15 10:54:28 +00:00
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2006-12-15 Thomas Vander Stichele <thomas at apestaart dot org>
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* Makefile.am:
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* gst-plugins-good.doap:
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* gst-plugins-good.spec.in:
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adding doap file
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2006-12-14 16:20:15 +00:00
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2006-12-14 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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libflac-1.1.3 changed API again, but we can't build against it yet,
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so make sure our check doesn't use libflac-1.1.3 and add a comment
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to this effect.
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2006-12-14 14:25:17 +00:00
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2006-12-14 Tim-Philipp Müller <tim at centricular dot net>
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* gst/effectv/gstquark.c: (gst_quarktv_transform),
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(gst_quarktv_planetable_clear):
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Add some NULL pointer checks (possibly related to #385623).
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2006-12-12 18:45:58 +00:00
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2006-12-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag),
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(gst_tag_demux_chain):
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* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
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In streaming mode, if the first buffer we get doesn't have an
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offset, fix it up to be 0, otherwise trimming won't work later on
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and we'll be typefinding application/x-id3, which may result in
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decodebin plugging an endless number of id3demux elements as a
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consequence. Fixes #385031.
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2006-12-11 21:21:16 +00:00
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2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
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Ignore the buffer_time the sound device reports. Turns out it is
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sometimes completely bogus and we're better off without it.
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2006-12-11 13:59:33 +00:00
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2006-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
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(gst_matroska_demux_video_caps):
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* gst/matroska/matroska-ids.c:
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(gst_matroska_track_init_video_context):
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* gst/matroska/matroska-ids.h:
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Try harder to extract the framerate for video tracks correctly and
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save it directly instead of converting it back and forth a few
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times. Mostly makes a difference for very small framerates (<1).
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Fixes #380199.
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2006-12-11 11:41:18 +00:00
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2006-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_init),
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(gst_gconf_audio_src_dispose), (do_toggle_element):
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* ext/gconf/gstgconfaudiosrc.h:
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Remove gconf notify hook when the gconfaudiosrc element is
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destroyed, otherwise the callback may be called on an
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already-destroyed instance and bad things happen. Should fix
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#378184.
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Also ignore gconf key changes when the source is already running.
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2006-12-09 19:27:28 +00:00
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2006-12-09 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Sebastian Dröge <mail at slomosnail de>
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* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
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We need to be able to read and parse any possible floating point string
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format ("1,234" or "1.234") irrespective of the current locale. g_strod()
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will parse the former only in certain locales though, so we really need
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to canonicalise the separator to '.' and then use g_ascii_strtod() to
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make sure we can parse either version at all times.
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Fixes #382982 for real.
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sys/sunaudio/: Use the sunaudio debug category.
Original commit message from CVS:
* sys/sunaudio/gstsunaudiomixerctrl.c:
* sys/sunaudio/gstsunaudiosrc.c:
Use the sunaudio debug category.
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
(gst_sunaudiosink_open), (gst_sunaudiosink_close),
(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
(gst_sunaudiosink_reset):
* sys/sunaudio/gstsunaudiosink.h:
Uses the sunaudio debug category for all debug output
Implements the _delay() callback to synchronise video playback better
Change the segtotal and segsize values back to the parent class
defaults (taken from buffer_time and latency_times of 200ms and 10ms
respectively)
Measure the samples written to the device vs. played.
Keep track of segments in the device by writing empty eof frames, and
sleep using a GCond when we get too far ahead and risk overrunning the
sink's ringbuffer.
Fixes: #360673
2006-12-09 16:17:33 +00:00
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2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/sunaudio/gstsunaudiomixerctrl.c:
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* sys/sunaudio/gstsunaudiosrc.c:
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Use the sunaudio debug category.
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* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_finalize),
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(gst_sunaudiosink_class_init), (gst_sunaudiosink_init),
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(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
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(gst_sunaudiosink_open), (gst_sunaudiosink_close),
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(gst_sunaudiosink_prepare), (gst_sunaudio_sink_do_delay),
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(gst_sunaudiosink_write), (gst_sunaudiosink_delay),
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(gst_sunaudiosink_reset):
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* sys/sunaudio/gstsunaudiosink.h:
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|
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Uses the sunaudio debug category for all debug output
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Implements the _delay() callback to synchronise video playback better
|
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Change the segtotal and segsize values back to the parent class
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|
defaults (taken from buffer_time and latency_times of 200ms and 10ms
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|
|
respectively)
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Measure the samples written to the device vs. played.
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|
|
Keep track of segments in the device by writing empty eof frames, and
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sleep using a GCond when we get too far ahead and risk overrunning the
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sink's ringbuffer.
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Fixes: #360673
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2006-12-08 16:38:18 +00:00
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2006-12-08 Wim Taymans <wim@fluendo.com>
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Patch by: Sebastian Dröge <mail at slomosnail de >
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* gst/audiofx/audiopanorama.c: (gst_audio_panorama_init),
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(gst_audio_panorama_set_caps), (gst_audio_panorama_transform):
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* gst/audiofx/audiopanorama.h:
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Fix audiopanorame with float samples. Fixes #383726.
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2006-12-08 15:12:01 +00:00
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2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_reset):
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* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open),
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(gst_sunaudiosrc_reset):
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Implement reset functions to unblock the src/sink more quickly on
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state change requests.
|
2006-12-08 21:12:47 +00:00
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Patch by: Brian Cameron <brian dot cameron at sun com>
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2006-12-08 15:12:01 +00:00
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2006-12-08 14:42:42 +00:00
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2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/sunaudio/gstsunaudiomixer.c:
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(gst_sunaudiomixer_change_state):
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Construct the correct mixer device name when the AUDIODEV env var
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|
is set.
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|
Patch by: Jerry Tan <jerry.tan at sun dot com>
|
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|
Fixes: #383596
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2006-12-08 14:32:51 +00:00
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2006-12-08 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
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Apply patch to open the mixer control and set the MULTIPLE_OPEN
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ioctl. On solaris, the mixer device doesn't need opening non-blocking
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- it can be opened by multiple processes by default, but needs the ioctl for multiple opens within 1 process.
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Patch by: Jerry Tan <jerry.tan at sun dot com>
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Fixes: #349015
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2006-12-07 17:30:03 +00:00
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2006-12-07 Wim Taymans <wim@fluendo.com>
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* gst/smpte/gstmask.h:
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* gst/smpte/gstsmpte.c: (gst_smpte_class_init),
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(gst_smpte_setcaps), (gst_smpte_init), (gst_smpte_reset),
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(gst_smpte_collected), (gst_smpte_set_property),
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|
(gst_smpte_get_property), (gst_smpte_change_state), (plugin_init):
|
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|
* gst/smpte/gstsmpte.h:
|
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|
Port to 0.10 some more.
|
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|
Added duration property to specify the duration of the transition.
|
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Make framerate a fraction.
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Deprecate fps property, we only use negotiated fps.
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Added docs.
|
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Fix collectpad usage.
|
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Reset state in READY.
|
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Send NEWSEGMENT event.
|
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Fix racy updates of object properties.
|
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|
|
Added debug category.
|
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|
|
Fixes #383323.
|
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|
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2006-12-06 15:06:04 +00:00
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2006-12-06 Wim Taymans <wim@fluendo.com>
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|
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|
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|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
|
|
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* gst/videomixer/videomixer.c:
|
|
|
|
(gst_videomixer_set_master_geometry),
|
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|
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_collect_free):
|
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Don't reset xpos and ypos in the setcaps function because causes
|
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|
unexpected behaviour.
|
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|
Fixes #382179.
|
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|
|
|
2006-12-06 14:45:30 +00:00
|
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|
2006-12-06 Wim Taymans <wim@fluendo.com>
|
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|
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* gst/multipart/multipartmux.c: (gst_multipart_mux_compare_pads),
|
|
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|
(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected):
|
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|
Keep track of the buffer timestamp in the collectdata member instead
|
|
|
|
of modifying the buffer without making the metadata writable first.
|
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|
Fixes #382277.
|
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|
2006-12-06 14:33:54 +00:00
|
|
|
2006-12-06 Wim Taymans <wim@fluendo.com>
|
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|
|
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|
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|
Patch by: Rob Taylor <robtaylor at floopily dot org>
|
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|
|
|
|
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|
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
|
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|
If using multicast in udpsrc, bind to the multicast address rather than
|
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|
|
IN_ADDR_ANY.
|
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|
This allows the simultanous use of multiple udpsrcs listening on
|
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|
|
different multicat addresses. Without this all udpsrcs will receive all
|
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|
packets from all subscribed multicast addresses.
|
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|
Fixes #383001.
|
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2006-12-06 13:35:52 +00:00
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|
2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
|
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* ext/taglib/gstid3v2mux.cc:
|
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|
Don't attempt to write a NULL frame into the ID3 tag set when the
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|
|
|
createFrame method returned NULL.
|
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|
Fixes: #381857
|
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|
|
Patch by: Jonathan Matthew <jonathan at 0kaolin wh9 net >
|
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|
2006-12-06 13:16:59 +00:00
|
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|
2006-12-06 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
|
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|
Use g_strtod() instead of sscanf to parse doubles, so that it will
|
|
|
|
try parsing in the C locale if the current locale fails.
|
|
|
|
Fixes: #382982
|
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|
Patch by: Sebastian Dröge <mail at slomosnail de >
|
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|
2006-12-01 10:31:46 +00:00
|
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|
2006-12-01 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
* win32/MANIFEST:
|
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|
Fix compilation on win32 under VS8
|
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Patch by: Sergey Scobich <sergey dot scobich at gmail dot com>
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Partially fixes #381175
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2006-11-30 16:48:51 +00:00
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2006-11-30 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavimux.c:
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accept all mpegversions,fixes #380825
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spotted by: Jerome Alet
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2006-11-30 16:46:13 +00:00
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2006-11-30 Stefan Kost <ensonic@users.sf.net>
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_fill_format_list),
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(gst_v4l2src_queue_frame), (gst_v4l2src_grab_frame),
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(gst_v4l2src_get_capture), (gst_v4l2src_set_capture),
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(gst_v4l2src_capture_init), (gst_v4l2src_buffer_finalize):
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cleanup the error message a bit more
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2006-11-28 12:30:10 +00:00
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2006-11-28 Wim Taymans <wim@fluendo.com>
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* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
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Fix width and height properties.
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* ext/libcaca/gstcacasink.h:
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Fix compilation on newer libcaca that require us to include a new
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header. Fixes #379918.
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2006-11-28 11:52:27 +00:00
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2006-11-28 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
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* gst/rtsp/gstrtspsrc.h:
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* gst/rtsp/rtspext.h:
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_configure_stream),
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(rtsp_ext_wms_get_context):
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Add method so that extensions can choose to disable the setup of
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a stream.
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Make the WMS extension skip setup of x-wms-rtx streams. Fixes #377792.
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2006-11-27 16:29:07 +00:00
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2006-11-27 Wim Taymans <wim@fluendo.com>
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Patch by: Jonas Holmberg <jonas dot holmberg at axis dot com>
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* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
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2006-11-27 16:30:49 +00:00
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Push header in a separate buffer instead of memcpy:ing all data.
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Change LF => CRLF in headers.
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Move trailing LF to header. Fixes #379792.
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2006-11-27 16:29:07 +00:00
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2006-11-27 16:26:50 +00:00
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2006-11-27 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_chain):
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Small buffer overflow fix and improve debugging.
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2006-11-24 08:58:53 +00:00
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2006-11-24 Stefan Kost <ensonic@users.sf.net>
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* ext/esd/esdmon.h:
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* ext/esd/esdsink.h:
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remove obsolete _factory_init protos
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gst/avi/gstavidemux.c: remove dead code, tweak debugs statements, add comments, use _uint64_scale instead _uint64_sca...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex),
(gst_avi_demux_read_subindexes_push),
(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
(gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
remove dead code, tweak debugs statements, add comments, use
_uint64_scale instead _uint64_scale_int when using guint64 values,
small optimizations, reflow some error handling
2006-11-24 07:46:54 +00:00
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2006-11-24 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_index_entry_for_time),
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(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
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(gst_avi_demux_peek_chunk), (gst_avi_demux_parse_subindex),
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(gst_avi_demux_read_subindexes_push),
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(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
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(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
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(gst_avi_demux_sync), (gst_avi_demux_next_data_buffer),
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(gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
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(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
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(gst_avi_demux_stream_data), (gst_avi_demux_loop):
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remove dead code, tweak debugs statements, add comments, use
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_uint64_scale instead _uint64_scale_int when using guint64 values,
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small optimizations, reflow some error handling
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2006-11-22 17:39:13 +00:00
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2006-11-22 Edward Hervey <edward@fluendo.com>
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* po/.cvsignore:
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We never put .pot files in cvs. Let's ignore them all.
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2006-11-19 18:46:03 +00:00
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2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
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* po/POTFILES.in:
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... but better exclude files that aren't disted.
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2006-11-19 16:32:49 +00:00
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2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
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* po/POTFILES.in:
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Add v4l2 source files to list of files with translations, so the
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strings are actually extracted (however bad they still may be).
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2006-11-19 16:30:19 +00:00
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2006-11-19 Tim-Philipp Müller <tim at centricular dot net>
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
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Minor clean-ups: const-ify static array, remove trailing comma from
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last enum (gcc-2.9x trips over that), use GST_DEBUG_FUNCPTR.
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2006-11-19 13:41:53 +00:00
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2006-11-19 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
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Make sure that g_free always gets called on the same pointer that was
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returned by g_malloc. Fixes #376594.
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Do not leak memory if decompressed size is wrong.
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Remove unneeded check of return value of g_malloc.
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Patch by: René Stadler <mail@renestadler.de>
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2006-11-18 18:14:34 +00:00
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2006-11-18 Tim-Philipp Müller <tim at centricular dot net>
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_capture_deinit):
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Add missing curly brackets.
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2006-11-17 14:54:01 +00:00
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2006-11-17 Edgard Lima <edgard.lima@indt.org.br>
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* sys/v4l2/v4l2src_calls.c:
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Fix capture_deinit.
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2006-11-16 15:36:48 +00:00
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2006-11-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
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(gst_matroska_mux_request_new_pad):
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Use GST_DEBUG_FUNCPTR; activate request pad before returning it.
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* tests/check/elements/matroskamux.c: (setup_src_pad),
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(setup_sink_pad), (GST_START_TEST):
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Activate pads before using them.
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2006-11-16 15:04:55 +00:00
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2006-11-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
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Initialise variable to get rid of bogus compiler warning.
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2006-11-16 07:26:17 +00:00
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2006-11-16 Stefan Kost <ensonic@users.sf.net>
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Patch by: Ville Syrjala <ville.syrjala@movial.fi>
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* gst/rtp/gstrtph263pay.c:
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* gst/rtp/gstrtph263pdepay.c:
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* gst/rtp/gstrtph263ppay.c:
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Specify H.263 variant and version in the caps (fixes #361637)
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2006-11-15 17:44:01 +00:00
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2006-11-15 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtspconnection.c: (read_body):
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Don't set a data pointer to NULL and a size > 0 when we deal
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with empty packets.
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* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
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(rtsp_message_init_response), (rtsp_message_init_data),
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(rtsp_message_unset), (rtsp_message_free),
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(rtsp_message_take_body):
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Check that we can't create invalid empty packets.
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gst/matroska/matroska-mux.*: Add basic tag writing support; implement releasing pads (#374658).
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces),
(gst_matroska_mux_class_init), (gst_matroska_pad_free),
(gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
(gst_matroska_mux_track_header), (gst_matroska_mux_start),
(gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish):
* gst/matroska/matroska-mux.h:
Add basic tag writing support; implement releasing pads (#374658).
2006-11-15 12:08:20 +00:00
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2006-11-15 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_add_interfaces),
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(gst_matroska_mux_class_init), (gst_matroska_pad_free),
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(gst_matroska_mux_reset), (gst_matroska_mux_handle_sink_event),
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(gst_matroska_mux_request_new_pad), (gst_matroska_mux_release_pad),
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(gst_matroska_mux_track_header), (gst_matroska_mux_start),
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(gst_matroska_mux_write_simple_tag), (gst_matroska_mux_finish):
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* gst/matroska/matroska-mux.h:
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Add basic tag writing support; implement releasing pads (#374658).
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2006-11-15 11:19:13 +00:00
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2006-11-15 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
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(gst_matroska_demux_audio_caps):
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Handle opaque/unspecified A_AAC audio codec ID (fixes #374737).
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2006-11-15 00:12:19 +00:00
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2006-11-14 David Schleef <ds@schleef.org>
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* gst/matroska/matroska-mux.c: Add Dirac fourcc.
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2006-11-14 20:07:22 +00:00
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2006-11-14 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Sergey Scobich <sergey.scobich at gmail com>
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* win32/vs8/gst-plugins-good.sln:
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* win32/vs8/libgst1394.vcproj:
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* win32/vs8/libgstaasink.vcproj:
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* win32/vs8/libgstalaw.vcproj:
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* win32/vs8/libgstalpha.vcproj:
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* win32/vs8/libgstalphacolor.vcproj:
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* win32/vs8/libgstannodex.vcproj:
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* win32/vs8/libgstapetag.vcproj:
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* win32/vs8/libgstaudiofx.vcproj:
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* win32/vs8/libgstauparse.vcproj:
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* win32/vs8/libgstautodetect.vcproj:
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* win32/vs8/libgstavi.vcproj:
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* win32/vs8/libgstcacasink.vcproj:
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* win32/vs8/libgstcdio.vcproj:
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* win32/vs8/libgstcutter.vcproj:
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* win32/vs8/libgstdv.vcproj:
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* win32/vs8/libgsteffectv.vcproj:
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* win32/vs8/libgstflac.vcproj:
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* win32/vs8/libgstflxdec.vcproj:
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* win32/vs8/libgstgoom.vcproj:
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* win32/vs8/libgsticydemux.vcproj:
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* win32/vs8/libgstid3demux.vcproj:
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* win32/vs8/libgstjpeg.vcproj:
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* win32/vs8/libgstladspa.vcproj:
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* win32/vs8/libgstlevel.vcproj:
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* win32/vs8/libgstmatroska.vcproj:
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* win32/vs8/libgstmikmod.vcproj:
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* win32/vs8/libgstmng.vcproj:
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* win32/vs8/libgstmonoscope.vcproj:
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* win32/vs8/libgstmulaw.vcproj:
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* win32/vs8/libgstmultipart.vcproj:
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* win32/vs8/libgstpng.vcproj:
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* win32/vs8/libgstrtp.vcproj:
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* win32/vs8/libgstrtsp.vcproj:
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* win32/vs8/libgstshout2.vcproj:
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* win32/vs8/libgstsmpte.vcproj:
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* win32/vs8/libgstspeex.vcproj:
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* win32/vs8/libgsttaglib.vcproj:
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* win32/vs8/libgstudp.vcproj:
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* win32/vs8/libgstvideobalance.vcproj:
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* win32/vs8/libgstvideobox.vcproj:
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* win32/vs8/libgstvideoflip.vcproj:
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* win32/vs8/libgstvideomixer.vcproj:
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* win32/vs8/libgstwavenc.vcproj:
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* win32/vs8/libgstwavparse.vcproj:
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Make end-of-line returns unixy, so that when the files are checked
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out on win32 the line returns will be 0d 0a and not 0d 0d 0a.
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Hopefully fixes #366492.
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2006-11-14 15:55:32 +00:00
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2006-11-14 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
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Disable init_frames delay timestamp adjustment, it does not
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seem to be needed at all. Fixes #369621.
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gst/videomixer/videomixer.c: Fix memleak by unref'ing collectpads instance (when finalizing)
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/videomixer/videomixer.c:
(gst_videomixer_set_master_geometry),
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init),
(gst_videomixer_collect_free), (gst_videomixer_reset),
(gst_videomixer_init), (gst_videomixer_finalize),
(gst_videomixer_request_new_pad), (gst_videomixer_release_pad),
(gst_videomixer_collected), (gst_videomixer_change_state):
Fix memleak by unref'ing collectpads instance (when finalizing)
Implement releasing a request pad. Fixes #374479.
2006-11-13 18:31:18 +00:00
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2006-11-13 Wim Taymans <wim@fluendo.com>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/videomixer/videomixer.c:
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(gst_videomixer_set_master_geometry),
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(gst_videomixer_pad_sink_setcaps), (gst_videomixer_class_init),
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(gst_videomixer_collect_free), (gst_videomixer_reset),
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(gst_videomixer_init), (gst_videomixer_finalize),
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(gst_videomixer_request_new_pad), (gst_videomixer_release_pad),
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(gst_videomixer_collected), (gst_videomixer_change_state):
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Fix memleak by unref'ing collectpads instance (when finalizing)
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Implement releasing a request pad. Fixes #374479.
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2006-11-10 20:08:42 +00:00
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2006-11-10 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Sergey Scobich <sergey.scobich at gmail com>
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* win32/vs8/gst-plugins-good.sln:
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* win32/vs8/libgst1394.vcproj:
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* win32/vs8/libgstaasink.vcproj:
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* win32/vs8/libgstalaw.vcproj:
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* win32/vs8/libgstalpha.vcproj:
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* win32/vs8/libgstalphacolor.vcproj:
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* win32/vs8/libgstannodex.vcproj:
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* win32/vs8/libgstapetag.vcproj:
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* win32/vs8/libgstaudiofx.vcproj:
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* win32/vs8/libgstauparse.vcproj:
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* win32/vs8/libgstautodetect.vcproj:
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* win32/vs8/libgstavi.vcproj:
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* win32/vs8/libgstcacasink.vcproj:
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* win32/vs8/libgstcdio.vcproj:
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* win32/vs8/libgstcutter.vcproj:
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* win32/vs8/libgstdv.vcproj:
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* win32/vs8/libgsteffectv.vcproj:
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* win32/vs8/libgstflac.vcproj:
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* win32/vs8/libgstflxdec.vcproj:
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* win32/vs8/libgstgoom.vcproj:
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* win32/vs8/libgsticydemux.vcproj:
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* win32/vs8/libgstid3demux.vcproj:
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* win32/vs8/libgstjpeg.vcproj:
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* win32/vs8/libgstladspa.vcproj:
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* win32/vs8/libgstlevel.vcproj:
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* win32/vs8/libgstmatroska.vcproj:
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* win32/vs8/libgstmikmod.vcproj:
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* win32/vs8/libgstmng.vcproj:
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* win32/vs8/libgstmonoscope.vcproj:
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* win32/vs8/libgstmulaw.vcproj:
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* win32/vs8/libgstmultipart.vcproj:
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* win32/vs8/libgstpng.vcproj:
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* win32/vs8/libgstrtp.vcproj:
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* win32/vs8/libgstrtsp.vcproj:
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* win32/vs8/libgstshout2.vcproj:
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* win32/vs8/libgstsmpte.vcproj:
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* win32/vs8/libgstspeex.vcproj:
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* win32/vs8/libgsttaglib.vcproj:
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* win32/vs8/libgstudp.vcproj:
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* win32/vs8/libgstvideobalance.vcproj:
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* win32/vs8/libgstvideobox.vcproj:
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* win32/vs8/libgstvideoflip.vcproj:
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* win32/vs8/libgstvideomixer.vcproj:
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* win32/vs8/libgstwavenc.vcproj:
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* win32/vs8/libgstwavparse.vcproj:
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Add VS8 project files (note that many of the plugins in ext are
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disabled by default). Fixes #366492.
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2006-11-10 08:09:05 +00:00
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2006-11-10 Stefan Kost <ensonic@users.sf.net>
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_grab_frame):
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we do not translate debug messages
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2006-11-08 12:04:03 +00:00
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2006-11-08 Stefan Kost <ensonic@users.sf.net>
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* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
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fix categorisation, make short desc more explicit, remove unused code
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Fixes #372021
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2006-11-08 01:30:39 +00:00
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2006-11-08 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpL16depay.c:
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* gst/rtp/gstrtpamrdepay.c:
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* gst/rtp/gstrtpamrpay.c:
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* gst/rtp/gstrtpgsmdepay.c:
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* gst/rtp/gstrtph263pay.c:
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* gst/rtp/gstrtph263pdepay.c:
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* gst/rtp/gstrtph263ppay.c:
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* gst/rtp/gstrtph264depay.c:
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* gst/rtp/gstrtpmp2tdepay.c:
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* gst/rtp/gstrtpmp4gdepay.c:
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* gst/rtp/gstrtpmp4gpay.c:
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* gst/rtp/gstrtpmp4vdepay.c:
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* gst/rtp/gstrtpmp4vpay.c:
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* gst/rtp/gstrtpmpadepay.c:
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* gst/rtp/gstrtpmpapay.c:
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* gst/rtp/gstrtppcmadepay.c:
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* gst/rtp/gstrtppcmapay.c:
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* gst/rtp/gstrtppcmudepay.c:
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* gst/rtp/gstrtppcmupay.c:
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* gst/rtp/gstrtpspeexdepay.c:
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* gst/rtp/gstrtpspeexpay.c:
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* gst/rtp/gstrtpsv3vdepay.c:
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Fix element descriptions.
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2006-11-08 01:29:51 +00:00
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2006-11-08 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpvorbisdepay.c:
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_handle_buffer):
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Fix description.
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Small cleanup in the payloader.
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2006-11-08 Wim Taymans <wim@fluendo.com>
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gst/rtp/: Add theora pay/depayloaders.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
(gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
(gst_rtp_theora_depay_finalize),
(gst_rtp_theora_depay_parse_configuration),
(gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_switch_codebook),
(gst_rtp_theora_depay_process),
(gst_rtp_theora_depay_set_property),
(gst_rtp_theora_depay_get_property),
(gst_rtp_theora_depay_change_state),
(gst_rtp_theora_depay_plugin_init):
* gst/rtp/gstrtptheoradepay.h:
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
(gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
(gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
(gst_rtp_theora_pay_init_packet),
(gst_rtp_theora_pay_flush_packet),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer),
(gst_rtp_theora_pay_plugin_init):
* gst/rtp/gstrtptheorapay.h:
Add theora pay/depayloaders.
2006-11-08 01:28:00 +00:00
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
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(gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
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(gst_rtp_theora_depay_finalize),
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(gst_rtp_theora_depay_parse_configuration),
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(gst_rtp_theora_depay_setcaps),
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(gst_rtp_theora_depay_switch_codebook),
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(gst_rtp_theora_depay_process),
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(gst_rtp_theora_depay_set_property),
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(gst_rtp_theora_depay_get_property),
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(gst_rtp_theora_depay_change_state),
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(gst_rtp_theora_depay_plugin_init):
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* gst/rtp/gstrtptheoradepay.h:
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* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
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(gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
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(gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
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(gst_rtp_theora_pay_init_packet),
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(gst_rtp_theora_pay_flush_packet),
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(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
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(gst_rtp_theora_pay_handle_buffer),
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(gst_rtp_theora_pay_plugin_init):
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* gst/rtp/gstrtptheorapay.h:
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Add theora pay/depayloaders.
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2006-11-07 01:43:06 +00:00
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2006-11-06 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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gst/rtp/: Add theora pay/depayloaders.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtptheoradepay.c: (gst_rtp_theora_depay_base_init),
(gst_rtp_theora_depay_class_init), (gst_rtp_theora_depay_init),
(gst_rtp_theora_depay_finalize),
(gst_rtp_theora_depay_parse_configuration),
(gst_rtp_theora_depay_setcaps),
(gst_rtp_theora_depay_switch_codebook),
(gst_rtp_theora_depay_process),
(gst_rtp_theora_depay_set_property),
(gst_rtp_theora_depay_get_property),
(gst_rtp_theora_depay_change_state),
(gst_rtp_theora_depay_plugin_init):
* gst/rtp/gstrtptheoradepay.h:
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_base_init),
(gst_rtp_theora_pay_class_init), (gst_rtp_theora_pay_init),
(gst_rtp_theora_pay_setcaps), (gst_rtp_theora_pay_reset_packet),
(gst_rtp_theora_pay_init_packet),
(gst_rtp_theora_pay_flush_packet),
(gst_rtp_theora_pay_finish_headers), (gst_rtp_theora_pay_parse_id),
(gst_rtp_theora_pay_handle_buffer),
(gst_rtp_theora_pay_plugin_init):
* gst/rtp/gstrtptheorapay.h:
Add theora pay/depayloaders.
2006-11-08 01:28:00 +00:00
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We depend on gsttag to generate the vorbis comments.
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2006-11-07 01:43:06 +00:00
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* gst/rtp/gstrtpvorbisdepay.c:
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(gst_rtp_vorbis_depay_parse_configuration),
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(gst_rtp_vorbis_depay_setcaps),
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(gst_rtp_vorbis_depay_switch_codebook),
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(gst_rtp_vorbis_depay_process):
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* gst/rtp/gstrtpvorbisdepay.h:
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Parse configuration string in the depayloader.
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Implement selecting and switching to a new codebook.
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Receiving vorbis over RTP now works.
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_reset_packet),
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(gst_rtp_vorbis_pay_init_packet),
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(gst_rtp_vorbis_pay_finish_headers),
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(gst_rtp_vorbis_pay_handle_buffer):
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* gst/rtp/gstrtpvorbispay.h:
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Set timestamps on outgoing buffers and RTP packets.
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Fix configuration string, prepend number of Packet headers.
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Fix encoding of ident string.
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Add delivery-method to caps.
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Streaming vorbis over RTP now works.
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2006-11-06 20:52:10 +00:00
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2006-11-06 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
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(gst_rtp_vorbis_pay_finish_headers), (gst_rtp_vorbis_pay_parse_id),
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(gst_rtp_vorbis_pay_handle_buffer):
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* gst/rtp/gstrtpvorbispay.h:
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Generate a valid configuration string in the caps based on the
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vorbis headers.
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2006-11-02 14:43:11 +00:00
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2006-11-02 Tim-Philipp Müller <tim at centricular dot net>
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* ext/cdio/gstcdio.c: (gst_cdio_get_cdtext):
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* ext/cdio/gstcdio.h:
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* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_open):
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Move CD-TEXT utility function into common file so it can also be
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used by a future cdioparanoiasrc.
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2006-11-01 19:48:26 +00:00
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2006-11-01 Edgard Lima <edgard.lima@indt.org.br>
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* sys/v4l2/Makefile.am:
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* sys/v4l2/gstv4l2object.c:
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* sys/v4l2/gstv4l2src.c:
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* sys/v4l2/gstv4l2xoverlay.c:
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* sys/v4l2/v4l2_calls.c:
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* sys/v4l2/v4l2src_calls.c:
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Improved comments in ELEMENT_ERROR/WARNING and added "#if 0" to
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xoverlay code that is still not implemented.
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2006-11-01 13:59:49 +00:00
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2006-11-01 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3v2frames.c: (parse_picture_frame):
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We require a -base more recent than 0.10.9, so it's safe to use
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GST_TYPE_TAG_IMAGE_TYPE unconditionally now.
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* ext/dv/gstdvdec.c: (gst_dvdec_sink_event):
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_sink_event):
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Use _newsegment_full() now that we depend on a recent enough core.
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* gst/wavparse/gstwavparse.c:
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Remove cruft that we don't need any longer now that we depend on
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a recent enough -base.
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2006-10-31 10:52:31 +00:00
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2006-10-31 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpilbcpay.c: (gst_rtpilbcpay_init),
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(gst_rtpilbcpay_setcaps):
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Fix and activate ILBC pay and depayloaders. Fixes #368162.
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2006-10-31 09:44:39 +00:00
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2006-10-31 Wim Taymans <wim@fluendo.com>
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* ext/speex/gstspeexdec.c: (speex_dec_convert),
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(speex_dec_sink_event), (speex_dec_chain_parse_header):
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Some small cleanups, use _scale.
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2006-10-31 09:29:36 +00:00
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2006-10-31 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
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Use higher precision scale function.
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2006-10-30 16:18:18 +00:00
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2006-10-30 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Michal Benes <michal dot benes at itonis tv>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_encoding_cmp),
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(gst_matroska_demux_read_track_encodings),
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(gst_matroska_decode_buffer):
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Fix several issues with encoded/compressed/encrypted/signed tracks;
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also, remove superfluous newline characters from some debug
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statements. (#366155)
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ext/jpeg/: Various cleanups, capsnego and leak fixes.
Original commit message from CVS:
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
(gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
(gst_smokedec_change_state):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
(gst_smokeenc_init), (gst_smokeenc_finalize),
(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
(gst_smokeenc_resync), (gst_smokeenc_chain),
(gst_smokeenc_set_property), (gst_smokeenc_get_property),
(gst_smokeenc_change_state):
Various cleanups, capsnego and leak fixes.
2006-10-30 09:24:53 +00:00
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2006-10-30 Wim Taymans <wim@fluendo.com>
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* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps):
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* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init),
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(gst_smokedec_init), (gst_smokedec_finalize), (gst_smokedec_chain),
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(gst_smokedec_change_state):
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init),
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(gst_smokeenc_init), (gst_smokeenc_finalize),
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(gst_smokeenc_getcaps), (gst_smokeenc_setcaps),
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(gst_smokeenc_resync), (gst_smokeenc_chain),
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(gst_smokeenc_set_property), (gst_smokeenc_get_property),
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(gst_smokeenc_change_state):
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Various cleanups, capsnego and leak fixes.
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2006-10-30 08:17:08 +00:00
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2006-10-30 Wim Taymans <wim@fluendo.com>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/videomixer/videomixer.c: (gst_videomixer_update_queues):
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Fix videomixer so that it can handle any combination of framerates.
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Fixes #367221.
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2006-10-28 16:37:20 +00:00
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2006-10-28 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
|
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(gst_avi_demux_parse_file_header),
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(gst_avi_demux_stream_init_push), (gst_avi_demux_parse_stream),
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(gst_avi_demux_stream_header_push), (gst_avi_demux_stream_data),
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(gst_avi_demux_chain):
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Fix position query for audio. also fixes timestamps in streaming
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mode and bug #364958.
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Small cleanups.
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2006-10-27 17:10:42 +00:00
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2006-10-27 Wim Taymans <wim@fluendo.com>
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* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps), (gst_pngenc_chain):
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* ext/libpng/gstpngenc.h:
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Fix strides. Fixes #364856.
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Cleanup capsnego.
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Set caps on outgoing buffers.
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2006-10-18 17:06:21 +00:00
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2006-10-18 Wim Taymans <wim@fluendo.com>
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Patch by: Ville Syrjala <ville dot syrjala at movial dot fi>
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* gst/rtp/gstrtpgsmpay.c:
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* gst/rtp/gstrtph263pay.c:
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* gst/rtp/gstrtpmpapay.c:
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* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
|
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(gst_rtp_pcma_pay_handle_buffer):
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* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush):
|
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Add static payload numbers in addition to the dynamic ones.
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Fixes #361639.
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2006-10-18 16:18:55 +00:00
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2006-10-18 Wim Taymans <wim@fluendo.com>
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|
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* gst/rtsp/gstrtspsrc.c: (gst_rtsp_lower_trans_get_type),
|
|
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|
(gst_rtspsrc_class_init), (gst_rtspsrc_loop_interleaved),
|
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|
(gst_rtspsrc_loop_udp), (gst_rtspsrc_open),
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(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_set_uri):
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_create):
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|
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|
* gst/rtsp/rtspdefs.h:
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|
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
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* gst/rtsp/rtspurl.h:
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Reuse already existing enum for lower transport.
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Add rtspt and rtspu protocols.
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Send redirect to rtspt when udp times out.
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2006-10-18 14:00:44 +00:00
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2006-10-18 Wim Taymans <wim@fluendo.com>
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|
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|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
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(gst_wavparse_stream_data):
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|
Fix seeking some more, mostly for speed changes.
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|
2006-10-18 11:04:09 +00:00
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2006-10-18 Tim-Philipp Müller <tim at centricular dot net>
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|
2006-10-18 11:28:05 +00:00
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Patch by: Fredrik Persson <frepe at bredband net>
|
2006-10-18 11:04:09 +00:00
|
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|
|
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* sys/v4l2/gstv4l2tuner.c:
|
|
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|
* sys/v4l2/gstv4l2tuner.h:
|
|
|
|
Fix _set_channel(): remove useless g_object_notify() for "channel"
|
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|
property that doesn't exist any longer and therefore now also
|
|
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useless redirect (#338818).
|
|
|
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|
2006-10-17 14:37:49 +00:00
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|
2006-10-17 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
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|
* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
|
|
|
|
Some drivers do not support unsetting the non-blocking flag once the
|
|
|
|
device is opened. In those cases, close/open the device in
|
|
|
|
non-blocking mode. Fixes #362673.
|
|
|
|
|
2006-10-17 13:44:14 +00:00
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|
2006-10-17 Stefan Kost <ensonic@users.sf.net>
|
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|
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* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
|
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
|
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(gst_v4l2src_get_fps):
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|
dear stefan, framespersecond is not frameperiod, reverting but adding
|
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comment
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2006-10-17 11:28:50 +00:00
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2006-10-17 Stefan Kost <ensonic@users.sf.net>
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* sys/v4l2/v4l2_calls.c: (gst_v4l2_fill_lists):
|
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* sys/v4l2/v4l2src_calls.c: (gst_v4l2src_set_fps),
|
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(gst_v4l2src_get_fps):
|
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|
Numerator is numerator and denominator is denominator. Say that aloud
|
|
|
|
5 times and retry after next beer.
|
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|
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|
2006-10-16 18:22:47 +00:00
|
|
|
2006-10-16 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Josep Torra Valles <josep at fluendo com>
|
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|
|
|
|
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* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_write):
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_length),
|
|
|
|
(gst_flac_dec_read_seekable), (gst_flac_dec_chain),
|
|
|
|
(gst_flac_dec_send_newsegment):
|
|
|
|
* ext/flac/gstflacenc.c: (gst_flac_enc_seek_callback),
|
|
|
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(gst_flac_enc_tell_callback):
|
|
|
|
* ext/jpeg/smokecodec.c: (find_best_size), (smokecodec_encode),
|
|
|
|
(smokecodec_parse_header), (smokecodec_decode):
|
|
|
|
* gst/avi/gstavimux.c: (gst_avi_mux_write_avix_index):
|
|
|
|
* gst/debug/efence.c: (gst_fenced_buffer_alloc):
|
|
|
|
* gst/goom/Makefile.am:
|
|
|
|
* gst/goom/gstgoom.c:
|
|
|
|
* gst/icydemux/gsticydemux.c: (gst_icydemux_typefind_or_forward):
|
|
|
|
* gst/rtsp/gstrtspsrc.c:
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
|
|
|
|
* gst/udp/gstudpsink.c:
|
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|
|
* gst/udp/gstudpsrc.c:
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_change_state):
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.h:
|
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|
|
Fix a bunch of problems discovered by the Forte compiler, mostly type
|
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|
|
mixups and pointer arithmetics with void pointers. Fixes #362603.
|
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|
|
|
ext/speex/: Miscellaneous clean-ups, among other things: speexenc => enc to enhance code readability; change speexenc...
Original commit message from CVS:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
(gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
(gst_speex_enc_class_init), (gst_speex_enc_finalize),
(gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
(gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
(gst_speex_enc_src_query), (gst_speex_enc_sink_query),
(gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
(gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
(gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
(gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
(gst_speex_enc_chain), (gst_speex_enc_get_property),
(gst_speex_enc_set_property), (gst_speex_enc_change_state):
* ext/speex/gstspeexenc.h:
Miscellaneous clean-ups, among other things: speexenc => enc to
enhance code readability; change speexenc => speex_enc; in chain
function unref input buffer in case of error; take reference in
event function; use boilerplate macro; use gst_pad_query_peer_*
convenience functions.
2006-10-12 19:02:51 +00:00
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2006-10-12 Tim-Philipp Müller <tim at centricular dot net>
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* ext/speex/gstspeex.c: (plugin_init):
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* ext/speex/gstspeexenc.c: (gst_speex_enc_get_formats),
|
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(gst_speex_enc_setup_interfaces), (gst_speex_enc_base_init),
|
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|
(gst_speex_enc_class_init), (gst_speex_enc_finalize),
|
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(gst_speex_enc_sink_setcaps), (gst_speex_enc_convert_src),
|
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(gst_speex_enc_convert_sink), (gst_speex_enc_get_query_types),
|
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(gst_speex_enc_src_query), (gst_speex_enc_sink_query),
|
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(gst_speex_enc_init), (gst_speex_enc_create_metadata_buffer),
|
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(gst_speex_enc_set_last_msg), (gst_speex_enc_setup),
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(gst_speex_enc_buffer_from_data), (gst_speex_enc_push_buffer),
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(gst_speex_enc_set_header_on_caps), (gst_speex_enc_sinkevent),
|
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(gst_speex_enc_chain), (gst_speex_enc_get_property),
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(gst_speex_enc_set_property), (gst_speex_enc_change_state):
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* ext/speex/gstspeexenc.h:
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|
Miscellaneous clean-ups, among other things: speexenc => enc to
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enhance code readability; change speexenc => speex_enc; in chain
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function unref input buffer in case of error; take reference in
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event function; use boilerplate macro; use gst_pad_query_peer_*
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convenience functions.
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2006-10-12 18:35:10 +00:00
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2006-10-12 Tim-Philipp Müller <tim at centricular dot net>
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* ext/speex/gstspeexenc.c: (gst_speexenc_finalize),
|
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(gst_speexenc_set_last_msg), (gst_speexenc_setup),
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(gst_speexenc_set_header_on_caps):
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Fix some mem leaks.
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2006-10-11 16:21:53 +00:00
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2006-10-11 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/URLS:
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Added some other URL.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_udp),
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(gst_rtspsrc_handle_request), (gst_rtspsrc_send),
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(gst_rtspsrc_open), (gst_rtspsrc_play),
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Work on fallback to TCP connection when the UDP socket times out.
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Handler server requests, just reply with OK for now.
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* gst/rtsp/rtspdefs.c: (rtsp_strresult):
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* gst/rtsp/rtspdefs.h:
|
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|
Added some more Real extension headers.
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* gst/rtsp/rtspurl.c: (rtsp_url_parse):
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Fix parsing of urls with a ':' that is not part of the hostname:port
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part of the url.
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2006-10-11 13:24:42 +00:00
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2006-10-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_add_srcpad):
|
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|
* gst/icydemux/gsticydemux.c: (gst_icydemux_add_srcpad):
|
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|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
|
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Activate pad before adding it to the already-running element.
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* tests/check/elements/icydemux.c: (icydemux_found_pad):
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Activate newly-created pad too.
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2006-10-11 08:34:14 +00:00
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2006-10-11 Wim Taymans <wim@fluendo.com>
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Patch by: Sebastien Cote <sebas642 at yahoo dot ca>
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
|
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(gst_udpsrc_finalize), (gst_udpsrc_create), (gst_udpsrc_set_uri),
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(gst_udpsrc_start):
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|
Fix some leaks in caps and uris. Fixes #361252.
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2006-10-10 08:29:07 +00:00
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2006-10-10 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavparse/Makefile.am:
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|
Fix copy'n'paste-o (spotted by Mark Nauwelaerts, #341489).
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2006-10-09 07:01:19 +00:00
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2006-10-09 Jan Schmidt <thaytan@mad.scientist.com>
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* sys/v4l2/gstv4l2xoverlay.c:
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* sys/v4l2/gstv4l2xoverlay.h:
|
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Fix build as per the patch in #338818 comment 36.
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|
|
2006-10-07 21:15:40 +00:00
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2006-10-07 Tim-Philipp Müller <tim at centricular dot net>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
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Activate pads before adding them to the source.
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2006-10-06 17:00:14 +00:00
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2006-10-06 Wim Taymans <wim@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads), (gst_dvdemux_chain):
|
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|
* gst/auparse/gstauparse.c: (gst_au_parse_add_srcpad):
|
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Activate pads before adding.
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2006-10-06 16:03:23 +00:00
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2006-10-06 Wim Taymans <wim@fluendo.com>
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* gst/multipart/multipartdemux.c: (gst_multipart_demux_init),
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(gst_multipart_find_pad_by_mime):
|
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Activate pads before adding.
|
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|
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|
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|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
|
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BOILERPLATE sets parent_class for us.
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gst/rtsp/gstrtspsrc.*: Rework how the transport string is constructed, try to share channels and udp ports.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
(gst_rtspsrc_class_init), (gst_rtspsrc_init),
(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_alloc_udp_ports),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_create_transports_string),
(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Rework how the transport string is constructed, try to share channels
and udp ports.
Make most of the stuff less dependant on RTP as we are also going to use
it for RDT.
Add support for transport specific session managers.
* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
Implement _flush().
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
Add generic error return code.
* gst/rtsp/rtspext.h:
Add support for pluggable tranport strings.
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
Detect WMServer and activate the extension.
* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
(rtsp_transport_get_manager), (rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Added methods to get mime/manager for certain transports.
2006-10-06 12:55:53 +00:00
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2006-10-06 Wim Taymans <wim@fluendo.com>
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type),
|
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|
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(gst_rtspsrc_class_init), (gst_rtspsrc_init),
|
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|
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(gst_rtspsrc_create_stream), (gst_rtspsrc_media_to_caps),
|
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|
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(gst_rtspsrc_alloc_udp_ports),
|
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(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
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(gst_rtspsrc_push_event), (gst_rtspsrc_loop_interleaved),
|
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|
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(gst_rtspsrc_create_transports_string),
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|
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(gst_rtspsrc_configure_transports), (gst_rtspsrc_open),
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|
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
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* gst/rtsp/gstrtspsrc.h:
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Rework how the transport string is constructed, try to share channels
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|
and udp ports.
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Make most of the stuff less dependant on RTP as we are also going to use
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it for RDT.
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Add support for transport specific session managers.
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* gst/rtsp/rtspconnection.c: (rtsp_connection_flush):
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Implement _flush().
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* gst/rtsp/rtspdefs.c: (rtsp_strresult):
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* gst/rtsp/rtspdefs.h:
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Add generic error return code.
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* gst/rtsp/rtspext.h:
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Add support for pluggable tranport strings.
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_before_send),
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(rtsp_ext_wms_after_send), (rtsp_ext_wms_parse_sdp),
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(rtsp_ext_wms_get_context):
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Detect WMServer and activate the extension.
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* gst/rtsp/rtsptransport.c: (rtsp_transport_get_mime),
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(rtsp_transport_get_manager), (rtsp_transport_parse):
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* gst/rtsp/rtsptransport.h:
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Added methods to get mime/manager for certain transports.
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2006-10-05 16:37:33 +00:00
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2006-10-05 Tim-Philipp Müller <tim at centricular dot net>
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* ext/cairo/gsttimeoverlay.c:
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(gst_cairo_time_overlay_update_font_height):
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* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_transform_caps):
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_parse_image_data):
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* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
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* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
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* ext/libpng/gstpngdec.c: (user_endrow_callback):
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|
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|
* gst/auparse/gstauparse.c: (gst_au_parse_parse_header):
|
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|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
|
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|
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
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(gst_avi_demux_stream_data):
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|
* gst/cutter/gstcutter.c: (gst_cutter_chain):
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* gst/debug/efence.c: (gst_efence_buffer_alloc),
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(gst_fenced_buffer_copy):
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* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
|
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
|
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_send),
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(gst_rtspsrc_handle_message):
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|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
|
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* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
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Printf format fixes.
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2006-10-04 17:53:12 +00:00
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2006-10-04 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/Makefile.am:
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Dist new .h file too.
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gst/rtsp/: Factor out extension in separate module.
Original commit message from CVS:
* gst/rtsp/Makefile.am:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
(gst_rtspsrc_parse_rtpmap),
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
* gst/rtsp/gstrtspsrc.h:
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
* gst/rtsp/rtspdefs.h:
* gst/rtsp/rtspext.h:
* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
(rtsp_ext_wms_get_context):
* gst/rtsp/rtspextwms.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
(rtsp_transport_parse):
* gst/rtsp/rtsptransport.h:
Factor out extension in separate module.
Fix getcaps to filter against the padtemplate.
Use Content-Base if the server gives one.
Rework the transport parsing a bit for future extensions.
Added some Real Header field definitions.
2006-10-04 17:24:40 +00:00
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2006-10-04 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/Makefile.am:
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* gst/rtsp/gstrtpdec.c: (gst_rtpdec_getcaps),
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(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp):
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_init),
|
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(gst_rtspsrc_finalize), (gst_rtspsrc_create_stream),
|
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(gst_rtspsrc_parse_rtpmap),
|
|
|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
|
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|
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(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
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(gst_rtspsrc_play), (gst_rtspsrc_handle_message):
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* gst/rtsp/gstrtspsrc.h:
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|
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
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* gst/rtsp/rtspdefs.h:
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* gst/rtsp/rtspext.h:
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* gst/rtsp/rtspextwms.c: (rtsp_ext_wms_parse_sdp),
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(rtsp_ext_wms_get_context):
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* gst/rtsp/rtspextwms.h:
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* gst/rtsp/rtsptransport.c: (rtsp_transport_init), (parse_mode),
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(rtsp_transport_parse):
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* gst/rtsp/rtsptransport.h:
|
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Factor out extension in separate module.
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Fix getcaps to filter against the padtemplate.
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Use Content-Base if the server gives one.
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Rework the transport parsing a bit for future extensions.
|
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Added some Real Header field definitions.
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2006-10-04 10:29:11 +00:00
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2006-10-04 Thomas Vander Stichele <thomas at apestaart dot org>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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added v4l2 stubs
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* gst-plugins-good.spec.in:
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add v4l2
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2006-10-04 10:24:49 +00:00
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2006-10-04 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gstapedemux.c: (ape_demux_parse_tags):
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Extract disc/album/medium number and count and try harder
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to extract track number/count.
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2006-10-03 18:15:58 +00:00
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2006-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* sys/Makefile.am:
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add build stuff for v4l2, needs --enable-experimental until
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the last bits are resolved
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2006-09-29 15:39:41 +00:00
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2006-09-29 Tim-Philipp Müller <tim at centricular dot net>
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* tests/check/Makefile.am:
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Disable autodetect test temporarily, so that the build bots
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update -bad and the ranks of unreliable video sinks in there.
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* tests/check/elements/autodetect.c: (GST_START_TEST):
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Skip test if no usable videosink is found.
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gst/rtsp/URLS: Add some more URLs.
Original commit message from CVS:
* gst/rtsp/URLS:
Add some more URLs.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
* gst/rtsp/gstrtspsrc.h:
Add timeout property to control UDP timeouts.
Fix error messages.
Also start a loop function when operating in UDP mode so that we can
do some more stuff async.
Handle element messages from udpsrc to detect timeouts. If a timeout
happens we currently generate an error.
API: rtspsrc::timeout property.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
(gst_udpsrc_create):
Really implement the timeout in microseconds and not milliseconds.
2006-09-29 15:37:29 +00:00
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2006-09-29 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/URLS:
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Add some more URLs.
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|
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|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
|
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(gst_rtspsrc_init), (gst_rtspsrc_finalize),
|
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|
|
(gst_rtspsrc_set_property), (gst_rtspsrc_get_property),
|
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|
|
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_loop_interleaved),
|
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|
|
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
|
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|
|
(gst_rtspsrc_loop), (gst_rtspsrc_send),
|
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|
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(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
|
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|
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(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause),
|
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|
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(gst_rtspsrc_handle_message), (gst_rtspsrc_change_state):
|
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|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Add timeout property to control UDP timeouts.
|
|
|
|
Fix error messages.
|
|
|
|
Also start a loop function when operating in UDP mode so that we can
|
|
|
|
do some more stuff async.
|
|
|
|
Handle element messages from udpsrc to detect timeouts. If a timeout
|
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|
|
happens we currently generate an error.
|
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|
API: rtspsrc::timeout property.
|
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|
|
|
|
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
|
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(gst_udpsrc_create):
|
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|
Really implement the timeout in microseconds and not milliseconds.
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2006-09-29 11:09:40 +00:00
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|
2006-09-29 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
|
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(gst_udpsrc_create), (gst_udpsrc_set_property),
|
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|
|
(gst_udpsrc_get_property), (gst_udpsrc_unlock), (gst_udpsrc_stop):
|
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|
|
* gst/udp/gstudpsrc.h:
|
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|
Added property to post a message on timeout.
|
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|
Updated docs.
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When restarting the select, initialize the fdsets again.
|
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Init control sockets so we don't accidentally close a random socket.
|
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API: GstUDPSrc::timeout property
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|
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2006-09-29 08:15:05 +00:00
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2006-09-29 Wim Taymans <wim@fluendo.com>
|
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|
|
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* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
|
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Fix flag registration.
|
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|
|
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* gst/rtsp/rtspconnection.c: (rtsp_connection_read):
|
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Reading 0 also means 'no more commands'
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|
2006-09-29 08:09:24 +00:00
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2006-09-29 Wim Taymans <wim@fluendo.com>
|
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|
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
|
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|
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|
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* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
|
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|
Fix possible infinite loop when shutting down, a read can also return
|
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0 to indicate no more messages are available. Fixes #358156.
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2006-09-25 13:55:44 +00:00
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2006-09-25 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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* gst/autodetect/gstautoaudiosink.c:
|
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|
|
(gst_auto_audio_sink_base_init), (gst_auto_audio_sink_class_init),
|
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|
|
(gst_auto_audio_sink_find_best):
|
|
|
|
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_detect):
|
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|
Small cleanups.
|
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don't try to set "sync" property when it is not available.
|
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2006-09-25 11:47:42 +00:00
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2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
|
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Patch by: Peter Kjellerstedt <pkj at axis com>
|
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|
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* gst/alpha/gstalpha.c:
|
|
|
|
* gst/rtp/gstrtpamrdepay.c:
|
|
|
|
* gst/rtsp/gstrtspsrc.c:
|
|
|
|
* gst/udp/gstudpsrc.c:
|
|
|
|
* gst/videomixer/videomixer.c:
|
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|
Include stdlib.h in some more places, makes things compile
|
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with uClibc and -Werror (#357592).
|
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|
|
|
2006-09-25 09:15:10 +00:00
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|
|
2006-09-25 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c:
|
|
|
|
Set minimum height to 8 (from 16), our code should handle
|
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|
|
that fine. Some of the buttons on the apple trailer site
|
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|
are apparently only 15 pixels high (see #357470).
|
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|
2006-09-23 15:31:56 +00:00
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|
2006-09-23 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop), (gst_rtspsrc_send),
|
|
|
|
(gst_rtspsrc_open):
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
|
|
|
|
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
|
|
|
|
(rtsp_connection_receive):
|
|
|
|
* gst/rtsp/rtspdefs.c: (rtsp_strresult):
|
|
|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
Improve error reporting.
|
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|
|
2006-09-23 15:30:40 +00:00
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|
2006-09-23 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
|
|
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_plugin_init):
|
|
|
|
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpdepay.c:
|
|
|
|
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_setcaps),
|
|
|
|
(gst_rtp_mp2t_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_plugin_init):
|
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|
|
Fix klass typos.
|
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|
|
Mark RANK_MARGINAL, decodebin can handle the depayloaders fine.
|
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2006-09-22 17:53:48 +00:00
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|
2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
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|
* configure.ac:
|
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|
|
Need -base CVS for gst_base_rtp_depayload_push_ts().
|
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2006-09-22 17:22:34 +00:00
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|
|
2006-09-22 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index):
|
|
|
|
Don't check for a tag that is never there and check if we read the
|
|
|
|
correct tag. Fixes seeking again.
|
|
|
|
We must post an error when all pads are unlinked.
|
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2006-09-22 15:15:13 +00:00
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2006-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_process):
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_setcaps),
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(gst_rtp_vorbis_pay_reset_packet),
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(gst_rtp_vorbis_pay_init_packet),
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(gst_rtp_vorbis_pay_flush_packet), (gst_rtp_vorbis_pay_parse_id),
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(gst_rtp_vorbis_pay_handle_buffer):
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More fixage, set endoder-params correctly in the payloader.
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2006-09-22 12:12:10 +00:00
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2006-09-22 Tim-Philipp Müller <tim at centricular dot net>
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* gst/autodetect/gstautoaudiosink.c:
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(gst_auto_audio_sink_base_init):
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* gst/autodetect/gstautovideosink.c:
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(gst_auto_video_sink_base_init):
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Make static pad templates static to appease valgrind's leak
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detector.
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* tests/check/Makefile.am:
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* tests/check/elements/.cvsignore:
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* tests/check/elements/autodetect.c: (GST_START_TEST),
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(autodetect_suite):
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Add simple test for the ghostpad lockup on shutdown fixed in core
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CVS (audio bit disabled because it would need dozens of alsa
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suppressions and I'm too lazy to add those now).
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gst/rtp/: Small cleanups.
Original commit message from CVS:
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
Small cleanups.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
(gst_rtp_vorbis_depay_process),
(gst_rtp_vorbis_depay_set_property),
(gst_rtp_vorbis_depay_get_property),
(gst_rtp_vorbis_depay_change_state),
(gst_rtp_vorbis_depay_plugin_init):
* gst/rtp/gstrtpvorbisdepay.h:
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
(gst_rtp_vorbis_pay_flush_packet),
(gst_rtp_vorbis_pay_append_buffer),
(gst_rtp_vorbis_pay_handle_buffer),
(gst_rtp_vorbis_pay_plugin_init):
* gst/rtp/gstrtpvorbispay.h:
Add experimental vorbis pay and depayloaders.
2006-09-22 12:08:14 +00:00
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2006-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_change_state):
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_class_init):
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Small cleanups.
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpvorbisdepay.c: (gst_rtp_vorbis_depay_base_init),
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(gst_rtp_vorbis_depay_class_init), (gst_rtp_vorbis_depay_init),
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(gst_rtp_vorbis_depay_finalize), (gst_rtp_vorbis_depay_setcaps),
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(gst_rtp_vorbis_depay_process),
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(gst_rtp_vorbis_depay_set_property),
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(gst_rtp_vorbis_depay_get_property),
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(gst_rtp_vorbis_depay_change_state),
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(gst_rtp_vorbis_depay_plugin_init):
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* gst/rtp/gstrtpvorbisdepay.h:
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* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_base_init),
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(gst_rtp_vorbis_pay_class_init), (gst_rtp_vorbis_pay_init),
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(gst_rtp_vorbis_pay_setcaps), (gst_rtp_vorbis_pay_init_packet),
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(gst_rtp_vorbis_pay_flush_packet),
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(gst_rtp_vorbis_pay_append_buffer),
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(gst_rtp_vorbis_pay_handle_buffer),
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(gst_rtp_vorbis_pay_plugin_init):
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* gst/rtp/gstrtpvorbispay.h:
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Add experimental vorbis pay and depayloaders.
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2006-09-21 13:33:16 +00:00
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2006-09-21 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_parse_audio_config):
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Fix profile-level-id parsing and setup.
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2006-09-21 09:50:41 +00:00
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2006-09-21 Wim Taymans <wim@fluendo.com>
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* gst/udp/README:
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* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
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Update README, simple cleanup.
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2006-09-21 09:35:13 +00:00
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2006-09-21 Wim Taymans <wim@fluendo.com>
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* gst/rtp/README:
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Update README with some examples.
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_init),
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(gst_rtp_mp4g_pay_finalize), (gst_rtp_mp4g_pay_parse_audio_config),
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(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
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(gst_rtp_mp4g_pay_setcaps):
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* gst/rtp/gstrtpmp4gpay.h:
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Make optional RTP parameters of type STRING, as required by the
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application/x-rtp caps specification.
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2006-09-20 19:37:45 +00:00
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2006-09-20 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
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2006-09-21 09:35:13 +00:00
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* gst/rtp/gstrtph263pdepay.c:
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2006-09-20 19:37:45 +00:00
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* gst/rtp/gstrtph263ppay.c:
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Correctly calculate size of each H263+ RTP buffer taking into account MTU and
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RTP header.
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2006-09-20 16:41:48 +00:00
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2006-09-20 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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And makefile too.
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2006-09-20 16:09:03 +00:00
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2006-09-20 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpasfdepay.c: (gst_rtp_asf_depay_base_init),
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(gst_rtp_asf_depay_class_init), (gst_rtp_asf_depay_init),
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(decode_base64), (gst_rtp_asf_depay_setcaps),
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(gst_rtp_asf_depay_process), (gst_rtp_asf_depay_set_property),
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(gst_rtp_asf_depay_get_property), (gst_rtp_asf_depay_change_state),
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(gst_rtp_asf_depay_plugin_init):
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* gst/rtp/gstrtpasfdepay.h:
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Added preliminary ASF depayloader.
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* gst/rtp/gstrtph264depay.c: (decode_base64):
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Fix base64 decoding.
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gst/rtsp/URLS: Added some test URLS.
Original commit message from CVS:
* gst/rtsp/URLS:
Added some test URLS.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(gst_rtspsrc_loop), (gst_rtspsrc_open):
* gst/rtsp/gstrtspsrc.h:
When creating streams, give access to the complete SDP.
Fix some leaks.
Collect and merge global stream properties in stream caps.
Preliminary support for WMServer.
* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
(rtsp_connection_connect), (rtsp_connection_read), (read_body),
(rtsp_connection_receive):
* gst/rtsp/rtspconnection.h:
Make connection interruptable.
Refactor to make it reconnectable.
Don't fail on short reads when reading data packets.
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
(rtsp_url_get_port):
* gst/rtsp/rtspurl.h:
Add methods for getting/setting the port.
* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
(sdp_message_get_attribute_val), (sdp_media_get_attribute),
(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
(sdp_media_get_format), (sdp_parse_line),
(sdp_message_parse_buffer):
Fix headers.
Add methods for getting multiple attributes with the same name.
Increase buffer size when parsing.
Fix parsing of a=foo fields.
* gst/rtsp/test.c: (main):
Update to new connection API.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
* gst/rtsp/rtsptransport.h:
* gst/rtsp/sdp.h:
* gst/rtsp/sdpmessage.h:
* gst/rtsp/gstrtsp.c:
* gst/rtsp/gstrtsp.h:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtpdec.h:
* gst/rtsp/rtsp.h:
* gst/rtsp/rtspdefs.c:
* gst/rtsp/rtspdefs.h:
Dual licensed under MIT and LGPL now.
2006-09-20 16:06:27 +00:00
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2006-09-20 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/URLS:
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Added some test URLS.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
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(gst_rtspsrc_loop), (gst_rtspsrc_open):
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* gst/rtsp/gstrtspsrc.h:
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When creating streams, give access to the complete SDP.
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Fix some leaks.
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|
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Collect and merge global stream properties in stream caps.
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Preliminary support for WMServer.
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|
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* gst/rtsp/rtspconnection.c: (rtsp_connection_create),
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(rtsp_connection_connect), (rtsp_connection_read), (read_body),
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(rtsp_connection_receive):
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* gst/rtsp/rtspconnection.h:
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Make connection interruptable.
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Refactor to make it reconnectable.
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Don't fail on short reads when reading data packets.
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* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_set_port),
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(rtsp_url_get_port):
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* gst/rtsp/rtspurl.h:
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Add methods for getting/setting the port.
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* gst/rtsp/sdpmessage.c: (sdp_message_get_attribute_val_n),
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(sdp_message_get_attribute_val), (sdp_media_get_attribute),
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(sdp_media_get_attribute_val_n), (sdp_media_get_attribute_val),
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(sdp_media_get_format), (sdp_parse_line),
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(sdp_message_parse_buffer):
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Fix headers.
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Add methods for getting multiple attributes with the same name.
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Increase buffer size when parsing.
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Fix parsing of a=foo fields.
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* gst/rtsp/test.c: (main):
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Update to new connection API.
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* gst/rtsp/rtspmessage.c: (rtsp_message_new_response),
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(rtsp_message_init_response), (rtsp_message_init_data),
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(rtsp_message_unset), (rtsp_message_free), (rtsp_message_dump):
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* gst/rtsp/rtspmessage.h:
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* gst/rtsp/rtsptransport.c: (rtsp_transport_free):
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* gst/rtsp/rtsptransport.h:
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* gst/rtsp/sdp.h:
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* gst/rtsp/sdpmessage.h:
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* gst/rtsp/gstrtsp.c:
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* gst/rtsp/gstrtsp.h:
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* gst/rtsp/gstrtpdec.c:
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* gst/rtsp/gstrtpdec.h:
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* gst/rtsp/rtsp.h:
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* gst/rtsp/rtspdefs.c:
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* gst/rtsp/rtspdefs.h:
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Dual licensed under MIT and LGPL now.
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gst/rtsp/gstrtspsrc.*: Reorganize stream parsing and creation.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Reorganize stream parsing and creation.
Detect container formats in interleaved mode.
Keep more state about the streams.
Assume a server also supports PLAY if it does not say.
Add unicast and interleaved properties to TCP transport requests to make
some servers happy (WMServer).
* gst/rtsp/sdpmessage.h:
Add some defines for the standard Bandwidth types.
2006-09-19 17:25:15 +00:00
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2006-09-19 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (find_stream_by_pt),
|
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(gst_rtspsrc_create_stream), (gst_rtspsrc_free_stream),
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(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
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(gst_rtspsrc_stream_configure_transport), (find_stream_by_channel),
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(gst_rtspsrc_push_event), (gst_rtspsrc_loop), (gst_rtspsrc_send),
|
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(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
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(gst_rtspsrc_parse_rtpinfo), (gst_rtspsrc_play):
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|
* gst/rtsp/gstrtspsrc.h:
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|
Reorganize stream parsing and creation.
|
|
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|
Detect container formats in interleaved mode.
|
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|
|
Keep more state about the streams.
|
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Assume a server also supports PLAY if it does not say.
|
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Add unicast and interleaved properties to TCP transport requests to make
|
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some servers happy (WMServer).
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|
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* gst/rtsp/sdpmessage.h:
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Add some defines for the standard Bandwidth types.
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2006-09-19 10:53:56 +00:00
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2006-09-19 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/test.c: (main):
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Fix build.
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2006-09-19 10:14:52 +00:00
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2006-09-19 Wim Taymans <wim@fluendo.com>
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* gst/wavparse/gstwavparse.c:
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Add ms-gsm to the src template.
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gst/rtsp/gstrtspsrc.*: Small cleanups, added documentation.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
* gst/rtsp/gstrtspsrc.h:
Small cleanups, added documentation.
Try to clean up the requests and responses.
Refactor parsing the supported methods.
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
(rtsp_connection_create), (rtsp_connection_send),
(parse_response_status), (parse_request_line),
(rtsp_connection_receive), (rtsp_connection_close),
(rtsp_connection_free):
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
(rtsp_transport_init), (rtsp_transport_parse),
(rtsp_transport_free):
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
Use g_return_val some more.
* gst/rtsp/rtspdefs.h:
Add more enum values to track initial states.
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
(rtsp_message_init_request), (rtsp_message_new_response),
(rtsp_message_init_response), (rtsp_message_init_data),
(rtsp_message_unset), (rtsp_message_free),
(rtsp_message_add_header), (rtsp_message_remove_header),
(rtsp_message_get_header), (rtsp_message_set_body),
(rtsp_message_take_body), (rtsp_message_get_body),
(rtsp_message_steal_body), (rtsp_message_dump):
* gst/rtsp/rtspmessage.h:
Reorder arguments, object goes as the first one.
Use g_return_val some more.
2006-09-18 17:37:46 +00:00
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2006-09-18 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_send), (gst_rtspsrc_parse_methods),
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|
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
|
|
|
|
(gst_rtspsrc_pause), (gst_rtspsrc_change_state),
|
|
|
|
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Small cleanups, added documentation.
|
|
|
|
Try to clean up the requests and responses.
|
|
|
|
Refactor parsing the supported methods.
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_open),
|
|
|
|
(rtsp_connection_create), (rtsp_connection_send),
|
|
|
|
(parse_response_status), (parse_request_line),
|
|
|
|
(rtsp_connection_receive), (rtsp_connection_close),
|
|
|
|
(rtsp_connection_free):
|
|
|
|
* gst/rtsp/rtsptransport.c: (rtsp_transport_new),
|
|
|
|
(rtsp_transport_init), (rtsp_transport_parse),
|
|
|
|
(rtsp_transport_free):
|
|
|
|
* gst/rtsp/rtspurl.c: (rtsp_url_parse):
|
|
|
|
* gst/rtsp/sdpmessage.c: (sdp_message_new), (sdp_message_init),
|
|
|
|
(sdp_message_clean), (sdp_message_free), (sdp_media_new),
|
|
|
|
(sdp_media_init), (sdp_message_parse_buffer), (sdp_message_dump):
|
|
|
|
Use g_return_val some more.
|
|
|
|
|
|
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|
* gst/rtsp/rtspdefs.h:
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|
|
|
Add more enum values to track initial states.
|
|
|
|
|
|
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|
* gst/rtsp/rtspmessage.c: (rtsp_message_new_request),
|
|
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|
(rtsp_message_init_request), (rtsp_message_new_response),
|
|
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|
(rtsp_message_init_response), (rtsp_message_init_data),
|
|
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|
(rtsp_message_unset), (rtsp_message_free),
|
|
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|
(rtsp_message_add_header), (rtsp_message_remove_header),
|
|
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|
(rtsp_message_get_header), (rtsp_message_set_body),
|
|
|
|
(rtsp_message_take_body), (rtsp_message_get_body),
|
|
|
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(rtsp_message_steal_body), (rtsp_message_dump):
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|
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|
* gst/rtsp/rtspmessage.h:
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|
Reorder arguments, object goes as the first one.
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|
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|
Use g_return_val some more.
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|
|
|
|
2006-09-18 14:00:41 +00:00
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2006-09-18 Wim Taymans <wim@fluendo.com>
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|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_base_init),
|
|
|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_push_event),
|
|
|
|
(gst_rtspsrc_loop), (gst_rtspsrc_uri_set_uri):
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|
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|
* gst/rtsp/gstrtspsrc.h:
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|
|
|
Export sometimes source pad with correct caps on the template, create
|
|
|
|
the ghostpad from the template.
|
|
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|
Remove RTCP template as we never expose RTCP.
|
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|
Protect against invalid body size.
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|
Avoid memcpy when creating the output buffer.
|
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|
Properly post an error and send EOS when the loop function is shut down.
|
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|
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|
2006-09-18 11:29:12 +00:00
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|
2006-09-18 Wim Taymans <wim@fluendo.com>
|
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Based on patch by: Lutz Mueller <lutz at topfrose dot de>
|
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|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
|
|
|
|
(gst_rtspsrc_init), (gst_rtspsrc_set_property), (gst_rtspsrc_open),
|
|
|
|
(gst_rtspsrc_uri_get_uri), (gst_rtspsrc_uri_set_uri):
|
|
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|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Make sure we can never set an invalid location.
|
|
|
|
|
|
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|
* gst/rtsp/rtspmessage.c: (rtsp_message_steal_body):
|
|
|
|
* gst/rtsp/rtspmessage.h:
|
|
|
|
Added _steal_body method for future use.
|
|
|
|
|
|
|
|
* gst/rtsp/rtspurl.c: (rtsp_url_parse), (rtsp_url_free):
|
|
|
|
Make freeing of NULL url return immediatly.
|
|
|
|
|
2006-09-18 10:42:52 +00:00
|
|
|
2006-09-18 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Based on patch by: Lutz Mueller <lutz at topfrose dot de>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (_do_init), (gst_rtspsrc_class_init),
|
|
|
|
(gst_rtspsrc_init), (gst_rtspsrc_stream_setup_rtp),
|
|
|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_play),
|
|
|
|
(gst_rtspsrc_change_state):
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
Use boilerplate.
|
|
|
|
Make rtspsrc subclass GstBin to make state changes easier.
|
|
|
|
Add Range header field on the PLAY request.
|
|
|
|
|
2006-09-18 08:59:17 +00:00
|
|
|
2006-09-18 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Based on patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
|
|
|
|
(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
|
|
|
|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_open),
|
|
|
|
(gst_rtspsrc_close), (gst_rtspsrc_play), (gst_rtspsrc_pause):
|
|
|
|
* gst/rtsp/rtspconnection.c: (inet_aton):
|
|
|
|
Small cleanups.
|
|
|
|
when multicast is selected as the transport, create UDP sources and
|
|
|
|
connect to the multicast group.
|
|
|
|
Move parsing and setting of caps to a common place.
|
|
|
|
Fixes #349894.
|
|
|
|
|
2006-09-16 21:57:29 +00:00
|
|
|
2006-09-17 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* ext/flac/gstflactag.c:
|
|
|
|
* gst/alpha/gstalpha.c:
|
|
|
|
* gst/debug/breakmydata.c:
|
|
|
|
* gst/debug/negotiation.c:
|
|
|
|
* gst/debug/testplugin.c:
|
|
|
|
* gst/effectv/gstaging.c:
|
|
|
|
* gst/effectv/gstdice.c:
|
|
|
|
* gst/effectv/gstedge.c:
|
|
|
|
* gst/effectv/gstquark.c:
|
|
|
|
* gst/effectv/gstrev.c:
|
|
|
|
* gst/effectv/gstshagadelic.c:
|
|
|
|
* gst/effectv/gstvertigo.c:
|
|
|
|
* gst/effectv/gstwarp.c:
|
|
|
|
* gst/multipart/multipartdemux.c:
|
|
|
|
* gst/multipart/multipartmux.c:
|
|
|
|
* gst/videobox/gstvideobox.c:
|
|
|
|
* gst/videofilter/gstgamma.c:
|
|
|
|
* gst/videofilter/gstvideotemplate.c:
|
|
|
|
* gst/videomixer/videomixer.c:
|
|
|
|
* sys/sunaudio/gstsunaudiosrc.h:
|
2006-09-18 08:59:17 +00:00
|
|
|
More G_OBJECT macro fixing.
|
2006-09-16 21:57:29 +00:00
|
|
|
|
2006-09-16 14:30:59 +00:00
|
|
|
2006-09-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Yves Lefebvre <ivanohe at abacom dot com>
|
|
|
|
|
|
|
|
* gst/avi/gstavimux.c: (gst_avi_mux_stop_file):
|
|
|
|
Correctly set the dwLength in strh.
|
|
|
|
With this patch, the file duration is now displayed correctly in window
|
|
|
|
media player and the AVI plays completely. Fixes #356147
|
|
|
|
|
2006-09-15 17:10:22 +00:00
|
|
|
2006-09-15 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Darren Kenny <darren dot kenny at sun dot com>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_build_list):
|
|
|
|
Set the output track as the MASTER so that the gnome-settings-daemon
|
|
|
|
keybindings for changing the volume using the keyboard works.
|
|
|
|
Fixes #356142.
|
|
|
|
|
2006-09-15 16:01:48 +00:00
|
|
|
2006-09-15 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
|
|
|
|
Fix documentation, it is not possible to control the framerate of jpegdec
|
|
|
|
using filtered caps yet. Fixes #355210.
|
|
|
|
Return the downstream GstFlowReturn instead of GST_FLOW_OK so that we
|
|
|
|
stop when there is an error.
|
|
|
|
|
2006-09-14 11:05:35 +00:00
|
|
|
2006-09-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain_parse_tag):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
|
|
|
|
Don't interpret a first buffer with an offset of NONE as
|
|
|
|
'from the middle of the stream', but only a first buffer
|
|
|
|
that has a valid buffer offset that's non-zero (see #345449).
|
|
|
|
|
2006-09-14 10:38:42 +00:00
|
|
|
2006-09-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
|
|
|
|
(gst_icydemux_typefind_or_forward):
|
|
|
|
* gst/icydemux/gsticydemux.h:
|
|
|
|
When we merge/collect multiple incoming buffers for typefinding
|
|
|
|
purposes, keep an initial 0 offset on the first outgoing buffer
|
|
|
|
as well (otherwise id3demux won't work right). Fixes #345449.
|
|
|
|
Also Make buffer metadata writable before setting buffer caps.
|
|
|
|
|
|
|
|
* tests/check/elements/icydemux.c: (typefind_succeed),
|
|
|
|
(cleanup_icydemux), (push_data), (GST_START_TEST),
|
|
|
|
(icydemux_suite):
|
|
|
|
Small test case for the above.
|
|
|
|
|
2006-09-13 13:26:15 +00:00
|
|
|
2006-09-13 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_peek_chunk),
|
|
|
|
(gst_avi_demux_stream_index), (gst_avi_demux_sync),
|
|
|
|
(gst_avi_demux_stream_header_push),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
|
|
|
|
(gst_avi_demux_loop):
|
|
|
|
More code reuse and better logging in _peek_chunk(). Reintroduce check
|
|
|
|
for chunk sizes before reading them (avoid oom). Better handling for
|
|
|
|
invalid chunksizes when streaming.
|
|
|
|
|
2006-09-11 20:38:41 +00:00
|
|
|
2006-09-11 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/level/gstlevel.c: (gst_level_set_property):
|
|
|
|
* gst/level/gstlevel.h:
|
|
|
|
Fix type mixup in level->interval (gdouble<->guint64). Spotted by
|
|
|
|
René Stadler
|
|
|
|
|
2006-09-06 09:05:33 +00:00
|
|
|
2006-09-06 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
|
|
|
|
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
|
|
|
|
(gst_avi_demux_stream_data):
|
|
|
|
Revert one change to fix streaming avi (adapter size != data size).
|
|
|
|
|
2006-09-04 16:21:17 +00:00
|
|
|
2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Frédéric Riss <frederic.riss at gmail dot com>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_track_free),
|
|
|
|
(gst_matroska_demux_reset),
|
|
|
|
(gst_matroska_demux_read_track_encodings),
|
|
|
|
(gst_matroska_demux_add_stream), (gst_matroska_decode_buffer),
|
|
|
|
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
|
|
|
|
(gst_matroska_demux_subtitle_caps):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Add support for VOBSUB subtitle tracks and zlib-compressed
|
|
|
|
tracks. Make sure we start on a keyframe after a seek. (#343348)
|
|
|
|
|
gst/matroska/: not perfect yet though, needs some tweaking in flacdec; also, seeking could be better.
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
(gst_matroska_demux_push_flac_codec_priv_data),
(gst_matroska_demux_push_xiph_codec_priv_data),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
Add basic FLAC support (#311586), not perfect yet though, needs some
tweaking in flacdec; also, seeking could be better.
Do better bounds checking when deserialising vorbis stream headers
to make sure we don't read beyond the end of the buffer on bad input.
2006-09-04 15:06:25 +00:00
|
|
|
2006-09-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_push_hdr_buf),
|
|
|
|
(gst_matroska_demux_push_flac_codec_priv_data),
|
|
|
|
(gst_matroska_demux_push_xiph_codec_priv_data),
|
|
|
|
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
|
|
|
|
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Add basic FLAC support (#311586), not perfect yet though, needs some
|
|
|
|
tweaking in flacdec; also, seeking could be better.
|
|
|
|
Do better bounds checking when deserialising vorbis stream headers
|
|
|
|
to make sure we don't read beyond the end of the buffer on bad input.
|
|
|
|
|
2006-09-04 09:34:25 +00:00
|
|
|
2006-09-04 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Alessandro Decina <alessandro at nnva dot org>
|
|
|
|
|
|
|
|
* ext/annodex/gstcmmldec.c: (gst_cmml_dec_chain):
|
|
|
|
Seeking back in a file containing a CMML stream errors out if the seek
|
|
|
|
goes back up to the CMML headers. This is because after the seek the xml
|
|
|
|
processing instruction <?xml ...?> is submitted to the xml parser again,
|
|
|
|
which results in an error. The attached patch fixes the problem.
|
|
|
|
Fixes #353908.
|
|
|
|
|
|
|
|
* ext/annodex/gstcmmlenc.h:
|
|
|
|
Fix authors name.
|
|
|
|
|
|
|
|
|
2006-08-28 16:59:13 +00:00
|
|
|
2006-08-28 Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* ext/raw1394/gstdv1394src.c (gst_dv1394src_from_raw1394handle):
|
|
|
|
New helper function to lessen the ifdefs.
|
|
|
|
(GST_INFO_OBJECT):
|
|
|
|
(gst_dv1394src_iso_receive): Use it.
|
|
|
|
(gst_dv1394src_create): Also use the control sockets in iec61883
|
|
|
|
mode.
|
|
|
|
(gst_dv1394src_start, gst_dv1394src_stop): Always use a separate
|
|
|
|
handle for AVC operations; fixes #348233.
|
|
|
|
|
2006-08-27 17:14:06 +00:00
|
|
|
2006-08-27 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/inspect/plugin-audiofx.xml:
|
|
|
|
* docs/plugins/inspect/plugin-audiofxgood.xml:
|
|
|
|
* gst/audiofx/Makefile.am:
|
|
|
|
* gst/audiofx/audiofx.c:
|
|
|
|
* gst/audiofxgood/.cvsignore:
|
|
|
|
* gst/audiofxgood/Makefile.am:
|
|
|
|
* gst/audiofxgood/audiofx.c:
|
|
|
|
* gst/audiofxgood/audiopanorama.c:
|
|
|
|
* gst/audiofxgood/audiopanorama.h:
|
|
|
|
Rename again (audiofxgood -> audiofx).
|
|
|
|
|
2006-08-27 13:12:52 +00:00
|
|
|
2006-08-27 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_next_data_buffer),
|
|
|
|
(gst_avi_demux_stream_scan):
|
|
|
|
Initialze variables.
|
|
|
|
|
gst/avi/gstavidemux.*: More attempts to turn this into readable code.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_finalize),
(gst_avi_demux_reset), (gst_avi_demux_index_last),
(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
More attempts to turn this into readable code.
Don't leak adapters.
Calculate duration according to index more efficiently.
Don't try to act like we drive the pipeline in chain mode.
2006-08-25 16:21:37 +00:00
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2006-08-25 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
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(gst_avi_demux_init), (gst_avi_demux_finalize),
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(gst_avi_demux_reset), (gst_avi_demux_index_last),
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(gst_avi_demux_index_next), (gst_avi_demux_index_entry_for_time),
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(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_index),
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(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
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(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
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(gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_stream_header_pull), (gst_avi_demux_do_seek),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
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(gst_avi_demux_chain), (gst_avi_demux_sink_activate),
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(gst_avi_demux_change_state):
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* gst/avi/gstavidemux.h:
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More attempts to turn this into readable code.
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Don't leak adapters.
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Calculate duration according to index more efficiently.
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Don't try to act like we drive the pipeline in chain mode.
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2006-08-25 09:53:18 +00:00
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2006-08-25 Wim Taymans <wim@fluendo.com>
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* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt):
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Fix build.
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ext/annodex/gstannodex.c: Do some extra sanity checks.
Original commit message from CVS:
Patch by: Alessandro Decina <alessandro at nnva dot org>
* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
Do some extra sanity checks.
Fixes #350340.
* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
Check if clip->start_time is valid before adding the clip to the
track list.
Reset enc->preamble going from PAUSED to READY.
Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
only used for EOS.
Only post an error message if we were the one that created the fatal
GstFlowReturn value.
* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
Parse the seconds field of the npt-sec time format using %llu rather than
%d and check that the value scaled by GST_SECOND doesn't overflow.
Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
Lookup a clip's track with clip->track rather than clip->id which
makes no sense.
Identify a clip by its track and start time and not its xml id.
do some more input checking and make sure we don't do undefined shifts.
* tests/check/elements/cmmldec.c: (setup_cmmldec),
(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
(cmml_tag_message_pop), (check_headers), (push_clip_full),
(push_clip), (push_empty_clip), (check_output_clip),
(GST_START_TEST), (cmmldec_suite):
* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
(check_headers), (push_clip), (check_clip_times), (check_clip),
(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
Added some more checks.
2006-08-25 09:42:43 +00:00
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2006-08-25 Wim Taymans <wim@fluendo.com>
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Patch by: Alessandro Decina <alessandro at nnva dot org>
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* ext/annodex/gstannodex.c: (gst_annodex_granule_to_time):
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Do some extra sanity checks.
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Fixes #350340.
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* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_change_state),
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(gst_cmml_enc_parse_tag_head), (gst_cmml_enc_parse_tag_clip),
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(gst_cmml_enc_push_clip), (gst_cmml_enc_push):
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Check if clip->start_time is valid before adding the clip to the
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track list.
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Reset enc->preamble going from PAUSED to READY.
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Don't use GST_FLOW_UNEXPECTED for wrong usage of the element, it is
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only used for EOS.
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Only post an error message if we were the one that created the fatal
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GstFlowReturn value.
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* ext/annodex/gstcmmlutils.c: (gst_cmml_clock_time_from_npt),
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(gst_cmml_clock_time_to_granule), (gst_cmml_track_list_has_clip):
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Parse the seconds field of the npt-sec time format using %llu rather than
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%d and check that the value scaled by GST_SECOND doesn't overflow.
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Use guint64(s) to represent the keyindex and keyoffset fields of a granulepos.
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Lookup a clip's track with clip->track rather than clip->id which
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makes no sense.
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Identify a clip by its track and start time and not its xml id.
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do some more input checking and make sure we don't do undefined shifts.
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* tests/check/elements/cmmldec.c: (setup_cmmldec),
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(teardown_cmmldec), (check_output_buffer_is_equal), (push_data),
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(cmml_tag_message_pop), (check_headers), (push_clip_full),
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(push_clip), (push_empty_clip), (check_output_clip),
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(GST_START_TEST), (cmmldec_suite):
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* tests/check/elements/cmmlenc.c: (setup_cmmlenc),
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(teardown_cmmlenc), (check_output_buffer_is_equal), (push_data),
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(check_headers), (push_clip), (check_clip_times), (check_clip),
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(check_empty_clip), (GST_START_TEST), (cmmlenc_suite):
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Added some more checks.
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2006-08-24 19:00:22 +00:00
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2006-08-24 Stefan Kost <ensonic@users.sf.net>
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* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_class_init),
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(gst_audio_panorama_set_property),
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(gst_audio_panorama_get_property),
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(gst_audio_panorama_transform_m2s_int),
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(gst_audio_panorama_transform_s2s_int),
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(gst_audio_panorama_transform_m2s_float),
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(gst_audio_panorama_transform_s2s_float):
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* gst/audiofxgood/audiopanorama.h:
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* tests/check/elements/audiopanorama.c: (GST_START_TEST):
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Make also the pan-property float (saves scaling and yields better
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resolution)
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2006-08-24 18:17:20 +00:00
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2006-08-24 Stefan Kost <ensonic@users.sf.net>
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2006-08-24 18:23:14 +00:00
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* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
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(gst_audio_panorama_transform_m2s_float),
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(gst_audio_panorama_transform_s2s_float):
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ChangeLog surgery to add cymax's real name
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2006-08-24 Stefan Kost <ensonic@users.sf.net>
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Patch by: René Stadler <mail@renestadler.de>
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2006-08-24 18:17:20 +00:00
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* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_set_caps),
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(gst_audio_panorama_transform_m2s_int),
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(gst_audio_panorama_transform_s2s_int),
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(gst_audio_panorama_transform_m2s_float),
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(gst_audio_panorama_transform_s2s_float),
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(gst_audio_panorama_transform):
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* gst/audiofxgood/audiopanorama.h:
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2006-08-24 18:23:14 +00:00
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Added float support
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2006-08-24 18:17:20 +00:00
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2006-08-24 14:16:55 +00:00
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2006-08-24 Stefan Kost <ensonic@users.sf.net>
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* gst/audiofxgood/audiopanorama.c:
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(gst_audio_panorama_transform_m2s):
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Fix docs & debug category. Add Fixme for volume pan levels.
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gst/avi/gstavidemux.c: unbreak AVI index handling, some more debug, remove an obsolete adapter_flush that caused stre...
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(gst_avi_demux_chain):
unbreak AVI index handling, some more debug, remove an obsolete
adapter_flush that caused streaming to wander off in the wild
2006-08-24 13:51:15 +00:00
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2006-08-24 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes_pull),
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(gst_avi_demux_sync), (gst_avi_demux_stream_header_push),
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(gst_avi_demux_stream_header_pull),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
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(gst_avi_demux_chain):
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unbreak AVI index handling, some more debug, remove an obsolete
|
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adapter_flush that caused streaming to wander off in the wild
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2006-08-24 11:21:06 +00:00
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2006-08-24 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
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(gst_avi_demux_parse_superindex), (gst_avi_demux_parse_subindex),
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(gst_avi_demux_parse_stream), (gst_avi_demux_parse_odml),
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(gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_stream_header_push),
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(gst_avi_demux_stream_header_pull):
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* gst/avi/gstavidemux.h:
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Some more cleanups.
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Fix totalFrames parsing in ODML.
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Disable use of index for length calculation in case of ODML as this is
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broken now.
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2006-08-24 10:03:03 +00:00
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2006-08-24 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (gst_flac_dec_update_metadata):
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Use libgsttag helper function here too.
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gst/avi/gstavidemux.*: Initial streaming support for avidemux (fixes #336465)
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_dispose),
(gst_avi_demux_reset), (gst_avi_demux_index_next),
(gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk),
(gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull),
(gst_avi_demux_parse_subindex),
(gst_avi_demux_read_subindexes_push),
(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
(sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_peek_tag),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push),
(gst_avi_demux_stream_header_pull),
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
(push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain),
(gst_avi_demux_sink_activate), (gst_avi_demux_activate_push),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Initial streaming support for avidemux (fixes #336465)
2006-08-23 15:33:47 +00:00
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2006-08-23 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
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(gst_avi_demux_init), (gst_avi_demux_dispose),
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(gst_avi_demux_reset), (gst_avi_demux_index_next),
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(gst_avi_demux_index_entry_for_time), (gst_avi_demux_src_convert),
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(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
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(gst_avi_demux_peek_chunk_info), (gst_avi_demux_peek_chunk),
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(gst_avi_demux_stream_init_push), (gst_avi_demux_stream_init_pull),
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(gst_avi_demux_parse_subindex),
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(gst_avi_demux_read_subindexes_push),
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(gst_avi_demux_read_subindexes_pull), (gst_avi_demux_parse_stream),
|
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(sort), (gst_avi_demux_parse_index), (gst_avi_demux_stream_index),
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(gst_avi_demux_sync), (gst_avi_demux_peek_tag),
|
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(gst_avi_demux_massage_index), (gst_avi_demux_stream_header_push),
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(gst_avi_demux_stream_header_pull),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
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(push_tag_lists), (gst_avi_demux_loop), (gst_avi_demux_chain),
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(gst_avi_demux_sink_activate), (gst_avi_demux_activate_push),
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(gst_avi_demux_change_state):
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* gst/avi/gstavidemux.h:
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Initial streaming support for avidemux (fixes #336465)
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2006-08-22 17:20:41 +00:00
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2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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There is no taglibmux element ...
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* gst/rtsp/gstrtspsrc.c:
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Use '%' rather than '&perc;' in gtk-doc blurb, docs build
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was complaining about unknown entity here.
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2006-08-22 17:02:39 +00:00
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2006-08-22 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
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(gst_avi_demux_do_seek), (gst_avi_demux_handle_seek),
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(gst_avi_demux_process_next_entry):
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* gst/avi/gstavidemux.h:
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Mark DISCONT.
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Remove old unused fields and reorder the struct a bit.
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2006-08-22 16:45:37 +00:00
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2006-08-22 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_send), (gst_rtspsrc_close), (gst_rtspsrc_play),
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(gst_rtspsrc_pause):
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* gst/rtsp/gstrtspsrc.h:
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* sys/oss/gstosssink.c: (gst_oss_sink_open),
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(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
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Small documentation updates.
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gst/avi/gstavidemux.*: Precalc most of the duration query for each stream.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_index_entry_for_time),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Precalc most of the duration query for each stream.
Make seeking more correct.
Use GstSegment to track position and duration.
Code cleanups and leak fixes.
Calculate correct total duration based on index length.
2006-08-22 16:42:22 +00:00
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2006-08-22 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
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(gst_avi_demux_index_entry_for_time),
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(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
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(gst_avi_demux_stream_init), (gst_avi_demux_parse_stream),
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(gst_avi_demux_stream_index), (gst_avi_demux_peek_tag),
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(gst_avi_demux_next_data_buffer),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_stream_header), (gst_avi_demux_do_seek),
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(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
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(gst_avi_demux_process_next_entry), (gst_avi_demux_loop),
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(gst_avi_demux_sink_activate_pull), (gst_avi_demux_change_state):
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* gst/avi/gstavidemux.h:
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|
|
Precalc most of the duration query for each stream.
|
|
|
|
Make seeking more correct.
|
|
|
|
Use GstSegment to track position and duration.
|
|
|
|
Code cleanups and leak fixes.
|
|
|
|
Calculate correct total duration based on index length.
|
|
|
|
|
2006-08-22 13:53:34 +00:00
|
|
|
2006-08-22 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
|
|
|
|
(parse_insert_string_field):
|
|
|
|
If strings in text fields are marked ISO8859-1, but contain
|
|
|
|
valid UTF-8 already, then handle them as UTF-8 and ignore
|
|
|
|
the encoding. (#351794)
|
|
|
|
|
2006-08-22 12:28:24 +00:00
|
|
|
2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_scan_got_frame),
|
|
|
|
(gst_flac_dec_write), (gst_flac_dec_loop),
|
|
|
|
(gst_flac_dec_sink_event), (gst_flac_dec_chain),
|
|
|
|
(gst_flac_dec_src_query):
|
|
|
|
* ext/flac/gstflacdec.h:
|
|
|
|
Make flac-in-ogg work (#352100).
|
|
|
|
|
2006-08-22 12:10:32 +00:00
|
|
|
2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
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|
|
* gst/monoscope/gstmonoscope.c: (gst_monoscope_chain):
|
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|
Don't unref buffers of which we've already given away
|
|
|
|
ownership to the adapter.
|
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|
|
|
2006-08-22 10:32:34 +00:00
|
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|
2006-08-22 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
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|
|
* ext/speex/gstspeexdec.c: (speex_dec_chain_parse_comments):
|
|
|
|
Make metadata extraction actually work.
|
|
|
|
|
|
|
|
* ext/speex/gstspeexenc.c: (gst_speexenc_base_init),
|
|
|
|
(gst_speexenc_init), (gst_speexenc_create_metadata_buffer),
|
|
|
|
(gst_speexenc_chain):
|
|
|
|
Fix metadata writing: replace old code which wrote completely
|
|
|
|
broken tags with libgsttag-based code. Plus miscellaneous
|
|
|
|
code cleanups (use static pad templates etc.) and a bunch
|
|
|
|
of leak fixes.
|
|
|
|
|
2006-08-21 19:34:03 +00:00
|
|
|
2006-08-21 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/audiopanorama/.cvsignore:
|
|
|
|
* gst/audiopanorama/Makefile.am:
|
|
|
|
* gst/audiopanorama/audiofx.c:
|
|
|
|
* gst/audiopanorama/audiopanorama.c:
|
|
|
|
* gst/audiopanorama/audiopanorama.h:
|
|
|
|
die! die! die! you should never have been there
|
|
|
|
|
2006-08-21 13:12:47 +00:00
|
|
|
2006-08-21 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
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|
|
* tests/check/elements/audiopanorama.c: (GST_START_TEST):
|
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|
|
Fix invalid memory access in audiopanorama test suite.
|
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|
|
|
2006-08-21 11:34:41 +00:00
|
|
|
2006-08-21 Edward Hervey <edward@fluendo.com>
|
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|
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|
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|
|
* tests/check/elements/.cvsignore:
|
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|
|
ignore built file
|
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|
|
|
2006-08-21 10:46:21 +00:00
|
|
|
2006-08-21 Wim Taymans <wim@fluendo.com>
|
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|
|
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|
|
* gst/rtp/Makefile.am:
|
|
|
|
Fix the build again.
|
|
|
|
|
gst/audiofxgood/: resubmit with the desired name *again*
Original commit message from CVS:
* gst/audiofxgood/.cvsignore:
* gst/audiofxgood/Makefile.am:
* gst/audiofxgood/audiofx.c: (plugin_init):
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property),
(gst_audio_panorama_get_unit_size),
(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s),
(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
* gst/audiofxgood/audiopanorama.h:
resubmit with the desired name *again*
2006-08-21 09:21:27 +00:00
|
|
|
2006-08-21 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
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|
|
* gst/audiofxgood/.cvsignore:
|
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|
|
* gst/audiofxgood/Makefile.am:
|
|
|
|
* gst/audiofxgood/audiofx.c: (plugin_init):
|
|
|
|
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
|
|
|
|
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
|
|
|
|
(gst_audio_panorama_set_property),
|
|
|
|
(gst_audio_panorama_get_property),
|
|
|
|
(gst_audio_panorama_get_unit_size),
|
|
|
|
(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
|
|
|
|
(gst_audio_panorama_transform_m2s),
|
|
|
|
(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
|
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|
|
* gst/audiofxgood/audiopanorama.h:
|
|
|
|
resubmit with the desired name *again*
|
|
|
|
|
2006-08-20 13:09:51 +00:00
|
|
|
2006-08-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
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|
|
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_get_unit_size):
|
|
|
|
* gst/videobox/gstvideobox.c: (gst_video_box_get_unit_size):
|
|
|
|
use g_assert in _get_unit_size
|
|
|
|
|
2006-08-20 13:06:44 +00:00
|
|
|
2006-08-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-audiofxgood.xml:
|
|
|
|
cleanup -unused.txt to make it useful, add previously missing docs
|
|
|
|
|
|
|
|
* ext/Makefile.am:
|
2006-12-18 15:49:08 +00:00
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|
|
Quietly (accidentally) enable LADSPA for building by default,
|
|
|
|
despite the fact that it doesn't meet the plugin checklist.
|
|
|
|
-- Added by Jan Schmidt 18 Dec 2006
|
|
|
|
|
2006-08-20 13:06:44 +00:00
|
|
|
* ext/esd/esdmon.c:
|
|
|
|
* ext/esd/esdsink.c:
|
|
|
|
* ext/esd/gstesd.c: (plugin_init):
|
|
|
|
reflow to get rid of two external symbols
|
|
|
|
|
|
|
|
* gst/audiofxgood/audiofx.c: (plugin_init):
|
|
|
|
re-add
|
|
|
|
|
gst/audiofxgood/audiofx.c
Original commit message from CVS:
* configure.ac:
* gst/audiofxgood/.cvsignore:
* gst/audiofxgood/Makefile.am:
* gst/audiofxgood/audiofx.c
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
(gst_audio_panorama_set_property),
(gst_audio_panorama_get_property),
(gst_audio_panorama_get_unit_size),
(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
(gst_audio_panorama_transform_m2s),
(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
* gst/audiofxgood/audiopanorama.h:
* tests/check/Makefile.am:
* tests/check/elements/audiopanorama.c: (setup_panorama_m),
(setup_panorama_s), (cleanup_panorama), (GST_START_TEST),
(panorama_suite), (main):
Add audiofxgood plugin with audiopanorama element
2006-08-20 12:09:16 +00:00
|
|
|
2006-08-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* gst/audiofxgood/.cvsignore:
|
|
|
|
* gst/audiofxgood/Makefile.am:
|
|
|
|
* gst/audiofxgood/audiofx.c
|
|
|
|
* gst/audiofxgood/audiopanorama.c: (gst_audio_panorama_base_init),
|
|
|
|
(gst_audio_panorama_class_init), (gst_audio_panorama_init),
|
|
|
|
(gst_audio_panorama_set_property),
|
|
|
|
(gst_audio_panorama_get_property),
|
|
|
|
(gst_audio_panorama_get_unit_size),
|
|
|
|
(gst_audio_panorama_transform_caps), (gst_audio_panorama_set_caps),
|
|
|
|
(gst_audio_panorama_transform_m2s),
|
|
|
|
(gst_audio_panorama_transform_s2s), (gst_audio_panorama_transform):
|
|
|
|
* gst/audiofxgood/audiopanorama.h:
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
* tests/check/elements/audiopanorama.c: (setup_panorama_m),
|
|
|
|
(setup_panorama_s), (cleanup_panorama), (GST_START_TEST),
|
|
|
|
(panorama_suite), (main):
|
|
|
|
Add audiofxgood plugin with audiopanorama element
|
|
|
|
|
2006-08-18 17:00:53 +00:00
|
|
|
2006-08-18 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
More Oss docs fixage.
|
|
|
|
|
2006-08-18 16:52:21 +00:00
|
|
|
2006-08-18 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpsv3vdepay.c: (gst_rtp_sv3v_depay_base_init),
|
|
|
|
(gst_rtp_sv3v_depay_class_init), (gst_rtp_sv3v_depay_init),
|
|
|
|
(gst_rtp_sv3v_depay_finalize), (gst_rtp_sv3v_depay_setcaps),
|
|
|
|
(gst_rtp_sv3v_depay_process), (gst_rtp_sv3v_depay_set_property),
|
|
|
|
(gst_rtp_sv3v_depay_get_property),
|
|
|
|
(gst_rtp_sv3v_depay_change_state),
|
|
|
|
(gst_rtp_sv3v_depay_plugin_init):
|
|
|
|
* gst/rtp/gstrtpsv3vdepay.h:
|
|
|
|
Added experimental SVQ3 depayloader.
|
|
|
|
|
2006-08-18 13:25:06 +00:00
|
|
|
2006-08-18 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek),
|
|
|
|
(gst_dvdemux_loop), (gst_dvdemux_change_state):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
When handling seek requests, don't send the newsegment event from the
|
|
|
|
calling thread. Instead save it so it can be sent from the streaming
|
|
|
|
thread.
|
|
|
|
|
2006-08-17 15:51:50 +00:00
|
|
|
2006-08-17 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
|
|
|
|
|
|
|
|
* gst/multipart/multipartdemux.c: (multipart_parse_header):
|
|
|
|
Accept leading whitespace before the boundary
|
|
|
|
This patch makes the demuxer allow some whitespace before the actual
|
|
|
|
boundary. This makes the demuxer work with the ``old'' gstreamer
|
|
|
|
multipartmuxer again (which placed an extra \n before the start
|
|
|
|
of the stream) Fixes #349068.
|
|
|
|
|
2006-08-17 15:47:28 +00:00
|
|
|
2006-08-17 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_process):
|
|
|
|
Error out on non-implemented stuff.
|
|
|
|
|
2006-08-16 16:50:00 +00:00
|
|
|
2006-08-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Andy Wingo <wingo at pobox dot com>
|
|
|
|
|
|
|
|
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setup),
|
|
|
|
(gst_signal_processor_start), (gst_signal_processor_stop),
|
|
|
|
(gst_signal_processor_cleanup), (gst_signal_processor_setcaps),
|
|
|
|
(gst_signal_processor_pen_buffer), (gst_signal_processor_flush),
|
|
|
|
(gst_signal_processor_do_pulls), (gst_signal_processor_do_pushes),
|
|
|
|
(gst_signal_processor_change_state):
|
|
|
|
Make ladspa elements reusable. Fixes #350006.
|
|
|
|
|
2006-08-16 15:33:12 +00:00
|
|
|
2006-08-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/ladspa/gstladspa.c: (gst_ladspa_base_init):
|
|
|
|
Convert ' ' into '_'. Try to keep as many characters in the padtemplate
|
2006-08-16 16:50:00 +00:00
|
|
|
names as possible. Fixes #349901.
|
2006-08-16 15:33:12 +00:00
|
|
|
|
2006-08-16 14:47:50 +00:00
|
|
|
2006-08-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_flush),
|
|
|
|
(gst_signal_processor_do_pushes):
|
|
|
|
A push() gives away our refcount so we should not use the buffer on the
|
|
|
|
pen anymore.
|
|
|
|
|
2006-08-16 13:48:00 +00:00
|
|
|
2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
|
|
|
|
(gst_oss_mixer_element_finalize):
|
|
|
|
Don't leak device string.
|
|
|
|
|
2006-08-16 13:01:32 +00:00
|
|
|
2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Require CVS of GStreamer core and -base (for
|
|
|
|
GST_TAG_EXTENDED_COMMENT and gst_tag_parse_extended_comment()).
|
|
|
|
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Write extended comment tags properly (#348762).
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(parse_comment_frame):
|
|
|
|
Extract COMM frames into extended comments, which makes it
|
|
|
|
easier to properly retain the description bit of the tag
|
|
|
|
and maintain this information when re-tagging (#348762).
|
|
|
|
|
2006-08-16 12:02:48 +00:00
|
|
|
2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
Don't try to run annodex unit tests if the annodex
|
|
|
|
plugin has not been built (Fixes #351116).
|
|
|
|
|
2006-08-16 10:53:32 +00:00
|
|
|
2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
(gst_auto_audio_sink_find_best):
|
|
|
|
When we can't find a usable audiosink, don't error out,
|
|
|
|
but use a fake sink instead and post a warning message
|
|
|
|
on the bus (#341278).
|
|
|
|
|
2006-08-16 10:22:32 +00:00
|
|
|
2006-08-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init):
|
|
|
|
* sys/oss/gstosssink.c:
|
|
|
|
* sys/oss/gstosssrc.c:
|
|
|
|
Document OSS elements; add gtk-doc blurb with 'Since 0.10.5' for
|
|
|
|
ossmixer's new device property.
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
Add docs for OSS elements.
|
|
|
|
|
|
|
|
* docs/plugins/inspect/plugin-aasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alpha.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alphacolor.xml:
|
|
|
|
* docs/plugins/inspect/plugin-annodex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-apetag.xml:
|
|
|
|
* docs/plugins/inspect/plugin-auparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-autodetect.xml:
|
|
|
|
* docs/plugins/inspect/plugin-avi.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cacasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cairo.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cdio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cutter.xml:
|
|
|
|
* docs/plugins/inspect/plugin-debug.xml:
|
|
|
|
* docs/plugins/inspect/plugin-dv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-efence.xml:
|
|
|
|
* docs/plugins/inspect/plugin-effectv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-esdsink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flac.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flxdec.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gconfelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
|
|
|
|
* docs/plugins/inspect/plugin-goom.xml:
|
|
|
|
* docs/plugins/inspect/plugin-halelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-icydemux.xml:
|
|
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|
* docs/plugins/inspect/plugin-id3demux.xml:
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* docs/plugins/inspect/plugin-jpeg.xml:
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* docs/plugins/inspect/plugin-level.xml:
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* docs/plugins/inspect/plugin-matroska.xml:
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* docs/plugins/inspect/plugin-mulaw.xml:
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* docs/plugins/inspect/plugin-multipart.xml:
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* docs/plugins/inspect/plugin-navigationtest.xml:
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* docs/plugins/inspect/plugin-ossaudio.xml:
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* docs/plugins/inspect/plugin-png.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-rtsp.xml:
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* docs/plugins/inspect/plugin-shout2send.xml:
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* docs/plugins/inspect/plugin-smpte.xml:
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* docs/plugins/inspect/plugin-speex.xml:
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* docs/plugins/inspect/plugin-taglib.xml:
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* docs/plugins/inspect/plugin-udp.xml:
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* docs/plugins/inspect/plugin-videobalance.xml:
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* docs/plugins/inspect/plugin-videobox.xml:
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* docs/plugins/inspect/plugin-videoflip.xml:
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* docs/plugins/inspect/plugin-videomixer.xml:
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* docs/plugins/inspect/plugin-wavenc.xml:
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* docs/plugins/inspect/plugin-wavparse.xml:
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* docs/plugins/inspect/plugin-ximagesrc.xml:
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Update to CVS version.
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gst/rtp/: Caps extra properties must be defined as strings for depayloaders because they are generated from an SDP.
Original commit message from CVS:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpmp4gdepay.c:
Caps extra properties must be defined as strings for
depayloaders because they are generated from an SDP.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
(gst_rtp_h264_depay_finalize), (decode_base64),
(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
(gst_rtp_h264_depay_set_property),
(gst_rtp_h264_depay_get_property),
(gst_rtp_h264_depay_change_state),
(gst_rtp_h264_depay_plugin_init):
* gst/rtp/gstrtph264depay.h:
Added basic, not completely functional RFC 3984 H264 depayloader.
2006-08-16 10:05:00 +00:00
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2006-08-16 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpamrdepay.c:
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* gst/rtp/gstrtpmp4gdepay.c:
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Caps extra properties must be defined as strings for
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depayloaders because they are generated from an SDP.
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtph264depay.c: (gst_rtp_h264_depay_base_init),
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(gst_rtp_h264_depay_class_init), (gst_rtp_h264_depay_init),
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(gst_rtp_h264_depay_finalize), (decode_base64),
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(gst_rtp_h264_depay_setcaps), (gst_rtp_h264_depay_process),
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(gst_rtp_h264_depay_set_property),
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(gst_rtp_h264_depay_get_property),
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(gst_rtp_h264_depay_change_state),
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(gst_rtp_h264_depay_plugin_init):
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* gst/rtp/gstrtph264depay.h:
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Added basic, not completely functional RFC 3984 H264 depayloader.
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gst/rtsp/gstrtpdec.c: Add pads after setting them up.
Original commit message from CVS:
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
Add pads after setting them up.
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
(gst_rtspsrc_init), (gst_rtspsrc_finalize),
(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp),
(gst_rtspsrc_stream_configure_transport),
(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
(gst_rtspsrc_pause):
* gst/rtsp/gstrtspsrc.h:
Fix interleaved mode.
- Protect streaming with lock.
- Combine flows
- set caps on outgoing buffers.
- strip trailing \0 from data packets.
- Configure RTP/RTCP in stream.
Use DEBUG_OBJECT more.
2006-08-16 09:48:26 +00:00
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2006-08-16 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
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Add pads after setting them up.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init),
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(gst_rtspsrc_init), (gst_rtspsrc_finalize),
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(gst_rtspsrc_free_stream), (gst_rtspsrc_media_to_caps),
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(gst_rtspsrc_stream_setup_rtp),
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(gst_rtspsrc_stream_configure_transport),
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(gst_rtspsrc_combine_flows), (gst_rtspsrc_loop),
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(gst_rtspsrc_open), (gst_rtspsrc_close), (gst_rtspsrc_play),
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(gst_rtspsrc_pause):
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* gst/rtsp/gstrtspsrc.h:
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Fix interleaved mode.
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- Protect streaming with lock.
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- Combine flows
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- set caps on outgoing buffers.
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- strip trailing \0 from data packets.
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- Configure RTP/RTCP in stream.
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Use DEBUG_OBJECT more.
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2006-08-16 09:29:20 +00:00
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2006-08-16 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_add):
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Turn a g_print into a DEBUG line.
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2006-08-16 09:25:17 +00:00
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2006-08-13 Wim Taymans <wim@fluendo.com>
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* sys/oss/gstossmixer.c: (gst_ossmixer_open), (gst_ossmixer_new):
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* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
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(gst_oss_mixer_element_init), (gst_oss_mixer_element_set_property),
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(gst_oss_mixer_element_get_property),
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(gst_oss_mixer_element_change_state):
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* sys/oss/gstossmixerelement.h:
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Small cleanups. Better error reporting.
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Add device property for the mixer instead of the hardcoded
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/dev/mixer. Fixes #350785.
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API: GstOssMixerElement::device property
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2006-08-15 22:44:27 +00:00
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2006-08-15 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Jens Granseuer <jensgr at gmx net>
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* gconf/Makefile.am:
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Make --disable-schemas work right (they still need
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to be copied to the installation directory, just not
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applied). Fixes #351347 (also #344100).
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2006-08-14 11:37:10 +00:00
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2006-08-14 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac: back to HEAD
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2006-08-14 11:14:43 +00:00
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=== release 0.10.4 ===
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2006-08-14 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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releasing 0.10.4, "Dear Leader"
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2006-08-10 14:10:28 +00:00
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2006-08-10 Thomas Vander Stichele <thomas at apestaart dot org>
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Patch by: Edward Hervey <edward@fluendo.com>
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* configure.ac:
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* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
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(gst_wavparse_stream_data):
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Send the newsegment event in the streaming thread.
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Fixes #347529
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2006-08-08 14:40:47 +00:00
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2006-08-08 Tim-Philipp Müller <tim at centricular dot net>
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* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
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(gst_smokeenc_resync), (gst_smokeenc_chain):
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Refuse sink caps in the encoder if width or height is not a
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2006-08-08 14:46:00 +00:00
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multiple of 16, the encoder does not support that yet (#349939);
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along the same lines, check the return value of the encoder
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setup function; also remove some debug log clutter.
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2006-08-08 14:40:47 +00:00
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2006-08-04 11:38:54 +00:00
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2006-08-04 Andy Wingo <wingo@pobox.com>
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* ext/ladspa/gstsignalprocessor.h: Add infrastructure for storing
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whether a processor can work in place or not, and for keeping
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track of its state. Change the FlowReturn instance variable from
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"state" to "flow_state", all callers changed.
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* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setup)
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(gst_signal_processor_start, gst_signal_processor_stop)
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(gst_signal_processor_cleanup): New functions to manage the
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processor's state.
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(gst_signal_processor_setcaps): start() as well as setup() here.
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(gst_signal_processor_prepare): Respect CAN_PROCESS_IN_PLACE.
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(gst_signal_processor_change_state): Stop and cleanup the
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processor as we go to NULL.
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* ext/ladspa/gstladspa.c (gst_ladspa_base_init): Reuse buffers if
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INPLACE_BROKEN is not set.
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* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_prepare):
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Do the alloc_buffer in bytes, not frames.
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2006-08-04 10:21:26 +00:00
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2006-08-04 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* sys/ximage/ximageutil.c: (ximageutil_xcontext_get):
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Fix rgb masks when recording in < 24bpp.
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2006-08-04 09:05:53 +00:00
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2006-08-04 Andy Wingo <wingo@pobox.com>
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* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_setcaps)
|
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(gst_signal_processor_prepare)
|
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(gst_signal_processor_update_inputs)
|
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(gst_signal_processor_process, gst_signal_processor_pen_buffer)
|
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(gst_signal_processor_flush)
|
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(gst_signal_processor_sink_activate_push)
|
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(gst_signal_processor_src_activate_pull)
|
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(gst_signal_processor_change_state): Remove the last of the code
|
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that assumes that we process whole buffers at a time. Fix some
|
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debugging. Seems to work now in some cases.
|
2006-08-04 09:20:26 +00:00
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(gst_signal_processor_src_activate_pull): BPB
|
2006-08-04 09:05:53 +00:00
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2006-07-31 22:03:09 +00:00
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2006-08-01 Andy Wingo <wingo@pobox.com>
|
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2006-07-31 22:27:22 +00:00
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* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_process):
|
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Fix nframes-choosing.
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(gst_signal_processor_init): Init pending_in and pending_out.
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2006-07-31 22:03:09 +00:00
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* ext/ladspa/gstsignalprocessor.c (gst_signal_processor_init): No
|
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more default sample rate, although we never check that the sample
|
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rate actually gets set. Something for the future.
|
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(gst_signal_processor_setcaps): Some refcount fixes, flow fixes.
|
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(gst_signal_processor_event): Refcount fixen.
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(gst_signal_processor_process): Pull the number of frames to
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process from the sizes of the buffers in the input pens.
|
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(gst_signal_processor_pen_buffer): Remove an incorrect FIXME :)
|
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(gst_signal_processor_do_pulls): Add an nframes argument, and use
|
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it instead of buffer_frames.
|
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(gst_signal_processor_getrange): Refcount fixen, pass nframes on
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to do_pulls.
|
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(gst_signal_processor_chain)
|
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(gst_signal_processor_sink_activate_push)
|
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(gst_signal_processor_src_activate_pull): Refcount fixen.
|
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* ext/ladspa/gstsignalprocessor.h: No more buffer_frames, yay.
|
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2006-07-31 19:44:18 +00:00
|
|
|
2006-07-31 Stefan Kost <ensonic@users.sf.net>
|
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* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_setcaps),
|
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(gst_signal_processor_process):
|
2006-08-14 11:14:43 +00:00
|
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don't query buffer-frames from caps, add lots of debug-log,
|
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|
|
try fix for assert (#349189)
|
2006-07-31 19:44:18 +00:00
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2006-07-31 15:58:43 +00:00
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2006-07-31 Wim Taymans <wim@fluendo.com>
|
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|
* gst/udp/gstudpsrc.c:
|
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|
Fix docs.
|
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|
ext/ladspa/gstsignalprocessor.c: Add debugs logs here and there, add more error handling, add some
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c:
(gst_signal_processor_add_pad_from_template),
(gst_signal_processor_init), (gst_signal_processor_setcaps),
(gst_signal_processor_process), (gst_signal_processor_pen_buffer),
(gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
(gst_signal_processor_sink_activate_push),
(gst_signal_processor_src_activate_pull),
(gst_signal_processor_change_state):
Add debugs logs here and there, add more error handling, add some
FIXME comments, filed #349189
2006-07-29 16:32:26 +00:00
|
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|
2006-07-29 Stefan Kost <ensonic@users.sf.net>
|
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* ext/ladspa/gstsignalprocessor.c:
|
|
|
|
(gst_signal_processor_add_pad_from_template),
|
|
|
|
(gst_signal_processor_init), (gst_signal_processor_setcaps),
|
|
|
|
(gst_signal_processor_process), (gst_signal_processor_pen_buffer),
|
|
|
|
(gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
|
|
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(gst_signal_processor_sink_activate_push),
|
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|
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(gst_signal_processor_src_activate_pull),
|
|
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(gst_signal_processor_change_state):
|
2006-08-14 11:14:43 +00:00
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Add debugs logs here and there, add more error handling, add some
|
|
|
|
FIXME comments, filed #349189
|
ext/ladspa/gstsignalprocessor.c: Add debugs logs here and there, add more error handling, add some
Original commit message from CVS:
* ext/ladspa/gstsignalprocessor.c:
(gst_signal_processor_add_pad_from_template),
(gst_signal_processor_init), (gst_signal_processor_setcaps),
(gst_signal_processor_process), (gst_signal_processor_pen_buffer),
(gst_signal_processor_do_pulls), (gst_signal_processor_getrange),
(gst_signal_processor_sink_activate_push),
(gst_signal_processor_src_activate_pull),
(gst_signal_processor_change_state):
Add debugs logs here and there, add more error handling, add some
FIXME comments, filed #349189
2006-07-29 16:32:26 +00:00
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2006-07-29 11:22:47 +00:00
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2006-07-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
|
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(gst_smokeenc_setcaps), (gst_smokeenc_chain):
|
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Set caps on buffer correctly. Fixes bug #349155.
|
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|
gst/multipart/multipartdemux.c: Uses GstAdapter instead of own buffering.
Original commit message from CVS:
Patch by: Sjoerd Simons <sjoerd at luon dot net>
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (get_line_end),
(multipart_parse_header), (multipart_find_boundary),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Uses GstAdapter instead of own buffering.
Actually parses the mime-type correctly (In tests the mime-type was
always "" with the old version).
Uses the Content-length header if available to speed up things.
Reliably autoscans the boundary name by default.
Fixes #349068.
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
Don't start the stream with a \n.
2006-07-28 16:17:17 +00:00
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2006-07-28 Wim Taymans <wim@fluendo.com>
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|
Patch by: Sjoerd Simons <sjoerd at luon dot net>
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* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
|
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|
|
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
|
|
|
|
(gst_multipart_demux_finalize), (get_line_end),
|
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|
|
(multipart_parse_header), (multipart_find_boundary),
|
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|
|
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
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|
(gst_multipart_set_property), (gst_multipart_get_property):
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Uses GstAdapter instead of own buffering.
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Actually parses the mime-type correctly (In tests the mime-type was
|
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|
always "" with the old version).
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|
|
Uses the Content-length header if available to speed up things.
|
|
|
|
Reliably autoscans the boundary name by default.
|
|
|
|
Fixes #349068.
|
|
|
|
|
|
|
|
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
|
|
|
|
Don't start the stream with a \n.
|
|
|
|
|
2006-07-28 08:32:47 +00:00
|
|
|
2006-07-28 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Brian Cameron <brian dot cameron at sun com>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_open):
|
|
|
|
Open source with O_NONBLOCK (#349015).
|
|
|
|
|
2006-07-28 08:21:27 +00:00
|
|
|
2006-07-28 Stefan Kost,,, <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_index),
|
|
|
|
(gst_avi_demux_massage_index):
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
Whitespace fixes and more debug
|
|
|
|
|
2006-07-27 11:21:53 +00:00
|
|
|
2006-07-27 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
(gst_auto_audio_sink_create_element_with_pretty_name),
|
|
|
|
(gst_auto_audio_sink_find_best),
|
|
|
|
(gst_auto_audio_sink_change_state):
|
|
|
|
Get rid of old and unused magic sound-server properties stuff.
|
|
|
|
Add suffix to child sink's name that makes it easy to see from
|
|
|
|
the name alone which type it actually is (alsa, oss, esd, etc.).
|
|
|
|
|
2006-07-27 10:05:27 +00:00
|
|
|
2006-07-27 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
|
|
|
|
(gst_udpsrc_set_property), (gst_udpsrc_get_property),
|
|
|
|
(gst_udpsrc_start):
|
|
|
|
* gst/udp/gstudpsrc.h:
|
|
|
|
Rename "buffer" to "buffer-size" to make clear it is a size we set and
|
|
|
|
not some sort of feature we enable.
|
|
|
|
|
2006-07-27 10:01:49 +00:00
|
|
|
2006-07-27 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
|
|
|
|
Use CLOSE_SOCKET() here instead of close() to maintain
|
|
|
|
win32 workiness.
|
|
|
|
|
2006-07-27 09:04:51 +00:00
|
|
|
2006-07-27 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Thijs Vermeir <thijs dot vermeir at barco dot com>
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
|
|
|
|
(gst_udpsrc_create), (gst_udpsrc_set_property),
|
|
|
|
(gst_udpsrc_get_property), (gst_udpsrc_start):
|
|
|
|
* gst/udp/gstudpsrc.h:
|
2006-07-27 10:05:27 +00:00
|
|
|
Added "buffer-size" property to control the kernel receive buffer size.
|
2006-07-27 09:04:51 +00:00
|
|
|
Update documentation.
|
|
|
|
Small cleanups. Fixes #348752.
|
2006-07-27 10:05:27 +00:00
|
|
|
API: buffer-size property
|
2006-07-27 09:04:51 +00:00
|
|
|
|
2006-07-26 16:36:59 +00:00
|
|
|
2006-07-26 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Kai Vehmanen <kv2004 at eca dot cx>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_flush),
|
|
|
|
(gst_rtp_pcma_pay_handle_buffer):
|
|
|
|
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_flush),
|
|
|
|
(gst_rtp_pcmu_pay_handle_buffer):
|
|
|
|
Fix timestamp calculation on outgoing RTP packets.
|
|
|
|
Fixes #348675.
|
|
|
|
|
2006-07-26 10:07:29 +00:00
|
|
|
2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Fix writing of comment frames (should be COMM not TCOM),
|
|
|
|
is still sub-optimal though, since we don't retain or
|
|
|
|
extract the comment descriptions properly (#334375,
|
|
|
|
also see #334375).
|
|
|
|
|
2006-07-26 09:02:56 +00:00
|
|
|
2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c:
|
|
|
|
#define 'fact' RIFF chunk if we are not compiling against
|
|
|
|
-base CVS (we don't want to depend on -base CVS for this
|
|
|
|
one define only, and also not for release order reasons).
|
|
|
|
|
2006-07-26 08:17:45 +00:00
|
|
|
2006-07-26 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Handle multiple tags of the same type properly. Re-inject
|
|
|
|
unparsed ID3v2 frames that we get as binary blobs from
|
|
|
|
id3demux into the tag again so we don't lose information
|
|
|
|
when retagging (#334375).
|
|
|
|
|
2006-07-25 17:54:25 +00:00
|
|
|
2006-07-25 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/ximage/gstximagesrc.c: (gst_ximage_src_class_init):
|
|
|
|
Document newly-added properties properly, so that there is a
|
|
|
|
'Since: 0.10.4' in the plugin docs. Convert some property
|
|
|
|
names into canonical GObject style (GObject will do that
|
|
|
|
internally anyway).
|
|
|
|
|
2006-07-25 16:47:04 +00:00
|
|
|
2006-07-25 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3tags.c:
|
|
|
|
(id3demux_add_id3v2_frame_blob_to_taglist):
|
|
|
|
Extract frames for ID3v2 versions prior to ID3v2.3.0 properly as
|
|
|
|
well, and add the version to the blob's buffer caps, since that
|
|
|
|
information will be needed for deserialisation later on (#348644).
|
|
|
|
|
2006-07-25 13:14:05 +00:00
|
|
|
2006-07-25 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_read_subindexes),
|
|
|
|
(gst_avi_demux_parse_stream):
|
|
|
|
Moved win32 variant of GST_DEBUG_CATEGORY_EXTERN to gstinfo.h. Fixed
|
|
|
|
indentation and spacing.
|
|
|
|
|
2006-07-24 15:25:49 +00:00
|
|
|
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
* docs/plugins/inspect/plugin-1394.xml:
|
|
|
|
* docs/plugins/inspect/plugin-aasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alpha.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alphacolor.xml:
|
|
|
|
* docs/plugins/inspect/plugin-annodex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-apetag.xml:
|
|
|
|
* docs/plugins/inspect/plugin-auparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-autodetect.xml:
|
|
|
|
* docs/plugins/inspect/plugin-avi.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cacasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cairo.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cdio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cutter.xml:
|
|
|
|
* docs/plugins/inspect/plugin-debug.xml:
|
|
|
|
* docs/plugins/inspect/plugin-dv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-efence.xml:
|
|
|
|
* docs/plugins/inspect/plugin-effectv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-esdsink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flac.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flxdec.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gconfelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gdkpixbuf.xml:
|
|
|
|
* docs/plugins/inspect/plugin-goom.xml:
|
|
|
|
* docs/plugins/inspect/plugin-halelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-icydemux.xml:
|
|
|
|
* docs/plugins/inspect/plugin-id3demux.xml:
|
|
|
|
* docs/plugins/inspect/plugin-jpeg.xml:
|
|
|
|
* docs/plugins/inspect/plugin-level.xml:
|
|
|
|
* docs/plugins/inspect/plugin-matroska.xml:
|
|
|
|
* docs/plugins/inspect/plugin-mulaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-multipart.xml:
|
|
|
|
* docs/plugins/inspect/plugin-navigationtest.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ossaudio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-png.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtsp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-shout2send.xml:
|
|
|
|
* docs/plugins/inspect/plugin-smpte.xml:
|
|
|
|
* docs/plugins/inspect/plugin-speex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-udp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videobalance.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videobox.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videoflip.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videomixer.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavenc.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ximagesrc.xml:
|
|
|
|
Update files to CVS/Prerelease version, add esdsink docs.
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c:
|
|
|
|
Add gtk-doc blurb.
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c:
|
|
|
|
Fix typo in element description.
|
|
|
|
|
2006-07-24 14:49:19 +00:00
|
|
|
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_open),
|
|
|
|
(gst_esdsink_factory_init):
|
|
|
|
Prevent libesd from auto-spawning a sound daemon if it
|
|
|
|
is not already running. Now that we don't do evil stuff
|
|
|
|
like that any longer we can give esdsink a rank so that
|
|
|
|
autoaudiosink will try it as well if all other audio
|
|
|
|
sinks fail (#343051).
|
|
|
|
|
2006-07-24 14:42:11 +00:00
|
|
|
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/esd/Makefile.am:
|
|
|
|
Oops, need to remove README from EXTRA_DIST as well.
|
|
|
|
|
2006-07-24 14:37:36 +00:00
|
|
|
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/esd/README:
|
|
|
|
Remove, it contains nothing useful anyway.
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_init), (gst_esdsink_prepare),
|
|
|
|
(gst_esdsink_delay):
|
|
|
|
Some small clean-ups; use GST_BOILERPLATE etc.
|
|
|
|
|
2006-07-24 14:16:06 +00:00
|
|
|
2006-07-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/law/alaw-decode.c: (alawdec_getcaps):
|
|
|
|
* gst/law/alaw-encode.c: (alawenc_getcaps), (gst_alawenc_chain):
|
|
|
|
* gst/law/mulaw-decode.c: (mulawdec_getcaps):
|
|
|
|
* gst/law/mulaw-encode.c: (mulawenc_getcaps):
|
|
|
|
Fix negotiation to deal with ANY/EMPTY caps instead of leaking.
|
|
|
|
|
2006-07-24 14:54:04 +00:00
|
|
|
2006-07-24 Stefan Kost <ensonic@users.sf.net>
|
2006-07-24 13:40:56 +00:00
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
|
|
|
|
(gst_wavparse_other), (gst_wavparse_perform_seek),
|
|
|
|
(gst_wavparse_get_upstream_size), (gst_wavparse_stream_headers),
|
|
|
|
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
|
|
|
|
(gst_wavparse_pad_query):
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
Use information from 'fact' chunk for length calculation of compressed
|
|
|
|
samples. Calculate bps if bogus value is found in wav header (embeded
|
|
|
|
mp2/mp3).
|
|
|
|
|
|
|
|
|
2006-07-24 11:48:03 +00:00
|
|
|
2006-07-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Based on patch by: Joni Valtanen <joni dot valtanen at movial fi>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* gst/udp/Makefile.am:
|
|
|
|
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_init),
|
|
|
|
(gst_dynudpsink_finalize), (gst_dynudpsink_close):
|
|
|
|
* gst/udp/gstdynudpsink.h:
|
|
|
|
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_init),
|
|
|
|
(gst_multiudpsink_finalize), (gst_multiudpsink_close):
|
|
|
|
* gst/udp/gstmultiudpsink.h:
|
|
|
|
* gst/udp/gstudp.c: (plugin_init):
|
|
|
|
* gst/udp/gstudpsink.h:
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create),
|
|
|
|
(gst_udpsrc_start), (gst_udpsrc_stop):
|
|
|
|
* gst/udp/gstudpsrc.h:
|
|
|
|
* gst/udp/gstudpnetutils.c: (gst_udp_net_utils_win32_inet_aton),
|
|
|
|
(gst_udp_net_utils_win32_wsa_startup):
|
|
|
|
* gst/udp/gstudpnetutils.h:
|
|
|
|
Port udp plugin to win32 (#345288).
|
|
|
|
|
2006-07-24 11:00:34 +00:00
|
|
|
2006-07-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_send):
|
|
|
|
Remove unwanted DEBUG line.
|
|
|
|
|
2006-07-23 11:33:54 +00:00
|
|
|
2006-07-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (plugin_init):
|
|
|
|
* gst/id3demux/id3tags.c:
|
|
|
|
(id3demux_add_id3v2_frame_blob_to_taglist):
|
|
|
|
* gst/id3demux/id3tags.h:
|
|
|
|
On second thought, it might be wiser and more efficient
|
|
|
|
not to do tag registration from a streaming thread.
|
|
|
|
|
2006-07-23 10:56:27 +00:00
|
|
|
2006-07-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3tags.c:
|
|
|
|
(id3demux_add_id3v2_frame_blob_to_taglist),
|
|
|
|
(id3demux_id3v2_frames_to_tag_list):
|
|
|
|
Put ID3v2 frames we can't parse as binary blobs into private
|
|
|
|
tags, so that they are not lost when retagging, at least once
|
|
|
|
id3v2mux has been taught to re-inject those frames again.
|
|
|
|
See bug #334375.
|
|
|
|
|
2006-07-21 10:57:00 +00:00
|
|
|
2006-07-21 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_process_next_entry):
|
|
|
|
Fix some leaks.
|
|
|
|
|
|
|
|
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
|
|
|
|
Don't use \n in debug lines.
|
|
|
|
|
2006-07-20 18:48:32 +00:00
|
|
|
2006-07-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
2006-08-14 11:14:43 +00:00
|
|
|
Add annodex and icydemux, cleanup the sections a bit
|
2006-07-20 18:48:32 +00:00
|
|
|
|
2006-07-19 11:52:53 +00:00
|
|
|
2006-07-19 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Alex Lancaster <alexl at users sourceforge net>
|
|
|
|
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Write GST_TAG_ENCODER and GST_TAG_ENCODER_VERSION as
|
|
|
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ID3v2 TSSE frames (#347898).
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|
2006-07-18 19:59:01 +00:00
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|
2006-07-18 Stefan Kost <ensonic@users.sf.net>
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* gst/avi/gstavimux.c: (gst_avi_mux_vidsink_set_caps):
|
2006-08-14 11:14:43 +00:00
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Respect mpegversion for "video/mpeg" and give message in case of
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unhandled versions.
|
2006-07-18 19:59:01 +00:00
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2006-07-17 10:22:54 +00:00
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2006-07-17 Wim Taymans <wim@fluendo.com>
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* ext/libpng/gstpngdec.c: (gst_pngdec_init), (buffer_clip),
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(gst_pngdec_caps_create_and_set), (gst_pngdec_task),
|
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(gst_pngdec_chain), (gst_pngdec_sink_event),
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(gst_pngdec_libpng_init), (gst_pngdec_change_state),
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(gst_pngdec_sink_activate_push):
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* ext/libpng/gstpngdec.h:
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Use statically allocated segment instead of leaking.
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Various cleanups.
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Fix flush and seek handling.
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gst/rtp/: Added simple generic mpeg4 depayloader.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init),
(gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init),
(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process),
(gst_rtp_mp4g_depay_set_property),
(gst_rtp_mp4g_depay_get_property),
(gst_rtp_mp4g_depay_change_state),
(gst_rtp_mp4g_depay_plugin_init):
* gst/rtp/gstrtpmp4gdepay.h:
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
(gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps),
(gst_rtp_mp4g_pay_flush):
Added simple generic mpeg4 depayloader.
Fix generic mpeg4 payloader.
2006-07-16 14:31:48 +00:00
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2006-07-16 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpmp4gdepay.c: (gst_rtp_mp4g_depay_base_init),
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(gst_rtp_mp4g_depay_class_init), (gst_rtp_mp4g_depay_init),
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(gst_rtp_mp4g_depay_setcaps), (gst_rtp_mp4g_depay_process),
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(gst_rtp_mp4g_depay_set_property),
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(gst_rtp_mp4g_depay_get_property),
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(gst_rtp_mp4g_depay_change_state),
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(gst_rtp_mp4g_depay_plugin_init):
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* gst/rtp/gstrtpmp4gdepay.h:
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
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(gst_rtp_mp4g_pay_parse_audio_config), (gst_rtp_mp4g_pay_setcaps),
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(gst_rtp_mp4g_pay_flush):
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Added simple generic mpeg4 depayloader.
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Fix generic mpeg4 payloader.
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2006-07-15 15:25:05 +00:00
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2006-07-15 Tim-Philipp Müller <tim at centricular dot net>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state):
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Don't try doing state changes on a NULL pointer.
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2006-07-14 13:33:54 +00:00
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2006-07-14 Wim Taymans <wim@fluendo.com>
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Patch by: Sebastien Cote <sebas642 at yahoo dot ca>
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_base_init),
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(gst_rtp_amr_depay_class_init), (gst_rtp_amr_depay_init),
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(gst_rtp_amr_depay_setcaps), (gst_rtp_amr_depay_process):
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* gst/rtp/gstrtpamrdepay.h:
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rtpamrdec isn't a subclass of GstBaseRtpDepayload.
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Fixes #321191
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2006-07-14 12:01:05 +00:00
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2006-07-14 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
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(gst_ximage_src_get_caps), (gst_ximage_src_class_init):
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Fix segfault when moving mouse pointer to the bottom right corner.
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2006-07-12 09:34:15 +00:00
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2006-07-12 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpmp2tdepay.c: (gst_rtp_mp2t_depay_base_init),
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(gst_rtp_mp2t_depay_class_init), (gst_rtp_mp2t_depay_init),
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(gst_rtp_mp2t_depay_setcaps), (gst_rtp_mp2t_depay_process),
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(gst_rtp_mp2t_depay_set_property),
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(gst_rtp_mp2t_depay_get_property),
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(gst_rtp_mp2t_depay_change_state),
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(gst_rtp_mp2t_depay_plugin_init):
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* gst/rtp/gstrtpmp2tdepay.h:
|
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Added mpeg2 TS depayloader. Closing #347234.
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2006-07-11 22:46:47 +00:00
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2006-07-11 Tim-Philipp Müller <tim at centricular dot net>
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* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_close):
|
2006-07-19 07:40:52 +00:00
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|
Remove g_assert that shouldn't be there and was triggered
|
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|
|
after trying to open a device that doesn't exist or can't
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|
|
be opened for some other reason (#347972).
|
2006-07-11 22:46:47 +00:00
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2006-07-10 20:11:34 +00:00
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2006-07-10 Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
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|
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(gst_avi_demux_stream_header), (push_tag_lists):
|
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|
* gst/avi/gstavidemux.h:
|
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|
|
Don't push tag events found by gst_riff_parse_info() before outputting
|
|
|
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GST_EVENT_NEWSEGMENT.
|
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2006-07-10 16:41:57 +00:00
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2006-07-10 Wim Taymans <wim@fluendo.com>
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|
* gst/rtsp/Makefile.am:
|
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|
|
* gst/rtsp/rtspconnection.c: (rtsp_connection_send),
|
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|
|
(rtsp_connection_close):
|
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|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
replaced closesocket and close in code with one CLOSE_SOCKET.
|
|
|
|
Some more cleanups. Fixes #345301.
|
|
|
|
|
2006-07-10 15:26:39 +00:00
|
|
|
2006-07-10 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
Fix example pipeline in docs.
|
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|
2006-07-10 10:21:57 +00:00
|
|
|
2006-07-10 Wim Taymans <wim@fluendo.com>
|
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|
|
Patch by: Rob Taylor <robtaylor at floopily dot org>
|
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|
|
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* gst/udp/gstmultiudpsink.c: (join_multicast),
|
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|
|
(gst_multiudpsink_init_send), (gst_multiudpsink_add):
|
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|
|
If a destination is added before the stream is set to PAUSED, the
|
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|
|
multicast group is not joined as the socket is not created yet.
|
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|
|
Also TTL and LOOP should also be set. Fixes #346921.
|
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|
2006-07-08 22:41:25 +00:00
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|
2006-07-09 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
|
|
|
|
(gst_ximage_src_set_property), (gst_ximage_src_get_property),
|
|
|
|
(gst_ximage_src_get_caps), (gst_ximage_src_class_init),
|
|
|
|
(gst_ximage_src_init):
|
|
|
|
* sys/ximage/gstximagesrc.h:
|
|
|
|
Fix use-damage property to actually work :)
|
|
|
|
Add startx, starty, endx, endy properties so screencasts other than full
|
|
|
|
screen ones can work.
|
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|
|
2006-07-08 19:03:54 +00:00
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|
|
2006-07-08 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get),
|
|
|
|
(gst_ximage_src_set_property), (gst_ximage_src_get_property),
|
|
|
|
(gst_ximage_src_class_init), (gst_ximage_src_init):
|
|
|
|
* sys/ximage/gstximagesrc.h:
|
|
|
|
Add use_damage property to offer ability to choose whether to use
|
|
|
|
XDamage or not.
|
|
|
|
|
2006-07-07 15:04:29 +00:00
|
|
|
2006-07-07 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/goom/filters.c: (zoomFilterSetResolution):
|
|
|
|
Avoid goom coredumping by clearing memory.
|
|
|
|
Fixes 345679.
|
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|
|
2006-07-05 20:21:02 +00:00
|
|
|
2006-07-05 Sebastien Moutte <sebastien@moutte.net>
|
|
|
|
|
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|
|
* win32/vs6/libgstid3demux.dsp:
|
|
|
|
Add a link to libgsttag-0.10.lib.
|
|
|
|
|
2006-07-05 14:52:13 +00:00
|
|
|
2006-07-05 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
|
|
|
|
(gst_tag_demux_read_range):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
|
|
|
|
(gst_id3demux_read_range):
|
|
|
|
Don't return FLOW_UNEXPECTED when a buffer is before
|
|
|
|
the start of the stream (which might happen with
|
|
|
|
large ID3v2 tags if the tag reading was done pullrange
|
|
|
|
based and we then switched to push mode later on).
|
|
|
|
Fixes regression introduced by commit from June 29th.
|
|
|
|
|
2006-07-05 10:14:16 +00:00
|
|
|
2006-07-05 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Make UTF-8 the default encoding when writing string
|
|
|
|
tags (before, our UTF-8 strings would automatically
|
|
|
|
be converted to ISO-8859-1 by taglib and written as
|
|
|
|
ISO-8859-1 fields if that was possible).
|
|
|
|
|
|
|
|
* tests/check/elements/id3v2mux.c: (utf8_string_in_buf),
|
|
|
|
(test_taglib_id3mux_check_tag_buffer), (identity_cb),
|
|
|
|
(test_taglib_id3mux_with_tags):
|
|
|
|
Add test case that makes sure our UTF-8 strings have
|
|
|
|
actually been written into the tag as UTF-8.
|
|
|
|
|
2006-07-04 16:00:26 +00:00
|
|
|
2006-07-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Let's try that again.
|
|
|
|
|
2006-07-04 15:40:47 +00:00
|
|
|
2006-07-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Disable monoscope plugin for now until it fulfills
|
|
|
|
all the requirements.
|
|
|
|
|
Port monoscope visualisation to 0.10.
Original commit message from CVS:
* configure.ac:
* gst/monoscope/Makefile.am:
* gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init),
(gst_monoscope_class_init), (gst_monoscope_init),
(gst_monoscope_finalize), (gst_monoscope_reset),
(gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps),
(gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain),
(gst_monoscope_sink_event), (gst_monoscope_src_event),
(gst_monoscope_change_state), (plugin_init):
* gst/monoscope/gstmonoscope.h:
Port monoscope visualisation to 0.10.
2006-07-03 20:35:45 +00:00
|
|
|
2006-07-03 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* gst/monoscope/Makefile.am:
|
|
|
|
* gst/monoscope/gstmonoscope.c: (gst_monoscope_base_init),
|
|
|
|
(gst_monoscope_class_init), (gst_monoscope_init),
|
|
|
|
(gst_monoscope_finalize), (gst_monoscope_reset),
|
|
|
|
(gst_monoscope_sink_setcaps), (gst_monoscope_src_setcaps),
|
|
|
|
(gst_monoscope_src_negotiate), (get_buffer), (gst_monoscope_chain),
|
|
|
|
(gst_monoscope_sink_event), (gst_monoscope_src_event),
|
|
|
|
(gst_monoscope_change_state), (plugin_init):
|
|
|
|
* gst/monoscope/gstmonoscope.h:
|
|
|
|
Port monoscope visualisation to 0.10.
|
|
|
|
|
2006-07-03 16:43:10 +00:00
|
|
|
2006-07-03 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
|
|
|
|
Return FLOW_UNEXPECTED when at the end of the file, not
|
|
|
|
FLOW_ERROR. Fixes 'internal stream error' errors that
|
|
|
|
would sometimes occur in totem when scrubbing to the
|
|
|
|
end of an ID3v1 tagged mp3 file.
|
|
|
|
|
2006-07-03 15:31:22 +00:00
|
|
|
2006-07-03 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_init), (user_info_callback),
|
|
|
|
(buffer_clip), (user_end_callback), (gst_pngdec_chain),
|
|
|
|
(gst_pngdec_sink_event), (gst_pngdec_change_state):
|
|
|
|
* ext/libpng/gstpngdec.h:
|
|
|
|
Implement buffer clipping/dropping using GstSegment.
|
|
|
|
This provides accurate seeking.
|
|
|
|
|
2006-07-03 15:28:48 +00:00
|
|
|
2006-07-03 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
|
|
|
|
(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_handle_seek), (gst_avi_demux_aggregated_flow),
|
|
|
|
(gst_avi_demux_process_next_entry), (push_tag_lists),
|
|
|
|
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
Proper aggregation of each stream's GstFlowReturn in order to figure out
|
|
|
|
whether the task should stop or not.
|
|
|
|
Don't send inline events before pushing out a NEW_SEGMENT, more
|
|
|
|
specifically for GST_TAG_EVENT.
|
|
|
|
Change a GST_ERROR to a GST_WARNING for a non-fatal situation in reading
|
|
|
|
sub-indexes.
|
|
|
|
|
2006-06-30 07:11:24 +00:00
|
|
|
2006-06-30 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_build_list):
|
|
|
|
Move "Monitor" slider to input tab so it works more like
|
|
|
|
sdtaudiocontrol, which is what people on Solaris are used
|
|
|
|
to using for their mixer program (#346259).
|
|
|
|
|
2006-06-29 14:50:18 +00:00
|
|
|
2006-06-29 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* tests/check/elements/level.c: (GST_START_TEST):
|
|
|
|
fix a leak, clean up at the end
|
|
|
|
|
2006-06-29 11:41:55 +00:00
|
|
|
2006-06-29 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
|
|
|
|
(gst_matroska_demux_send_event),
|
|
|
|
(gst_matroska_demux_loop_stream_parse_id):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Send tag event after newsegment event.
|
|
|
|
|
2006-06-29 11:11:50 +00:00
|
|
|
2006-06-29 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_trim_buffer),
|
|
|
|
(gst_id3demux_read_range):
|
|
|
|
Make sure we don't return GST_FLOW_OK with a NULL buffer in
|
|
|
|
certain cases where a read beyond the end of the file is
|
|
|
|
requested. Fixes #345930.
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_trim_buffer),
|
|
|
|
(gst_tag_demux_read_range):
|
|
|
|
Fix same issue here as well.
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2006-06-29 11:05:14 +00:00
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2006-06-29 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
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* sys/ximage/gstximagesrc.c: (gst_ximage_src_ximage_get):
|
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|
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|
Fix hypothetical crash.
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|
2006-06-28 08:36:30 +00:00
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2006-06-28 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
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|
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|
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* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_prepare):
|
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|
Do not modify the ports value. If the user has turned off the
|
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|
|
built-in speakers, then we should not reset it in the prepare
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|
|
function, since this causes the built-in speakers to turn
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|
|
back on anytime the user changes a track in totem, rhythmbox,
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|
|
|
etc. (#346066).
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|
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|
2006-06-23 09:35:45 +00:00
|
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|
2006-06-23 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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* gst/goom/gstgoom.c: (gst_goom_src_negotiate):
|
|
|
|
Fix double caps unref when negotiation fails.
|
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|
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|
2006-06-22 19:31:04 +00:00
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2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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* ext/annodex/gstcmmldec.c:
|
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* ext/annodex/gstcmmlenc.c:
|
|
|
|
* ext/annodex/gstcmmlparser.c:
|
|
|
|
* ext/dv/gstdvdec.c:
|
|
|
|
* ext/dv/gstdvdemux.c:
|
|
|
|
* ext/gdk_pixbuf/pixbufscale.c:
|
|
|
|
* ext/jpeg/gstjpegenc.c:
|
|
|
|
* ext/jpeg/gstsmokedec.c:
|
|
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|
* ext/jpeg/gstsmokeenc.c:
|
|
|
|
* ext/libpng/gstpngdec.c:
|
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|
|
* ext/libpng/gstpngenc.c:
|
|
|
|
* ext/speex/gstspeexenc.c:
|
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|
|
* gst/alpha/gstalphacolor.c:
|
|
|
|
* gst/cutter/gstcutter.c:
|
|
|
|
* gst/debug/gstnavigationtest.c:
|
|
|
|
* gst/icydemux/gsticydemux.c:
|
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|
|
* gst/level/gstlevel.c:
|
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|
|
* gst/multipart/multipart.c:
|
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|
* gst/rtp/gstrtpamrpay.c:
|
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* gst/rtp/gstrtpdepay.c:
|
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* gst/rtp/gstrtpilbcpay.c:
|
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* gst/rtp/gstrtpmp4gpay.c:
|
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|
* gst/rtp/gstrtpmp4vpay.c:
|
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|
* gst/rtsp/gstrtpdec.c:
|
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|
* gst/rtsp/gstrtspsrc.c:
|
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|
* gst/udp/gstdynudpsink.c:
|
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|
* gst/udp/gstmultiudpsink.c:
|
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|
|
* gst/udp/gstudpsrc.c:
|
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|
|
* gst/videobox/gstvideobox.c:
|
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|
|
* gst/videofilter/gstvideoflip.c:
|
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|
|
Use GST_DEBUG_CATEGORY_STATIC where possible (#342503)
|
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|
plus two minor macro fixes.
|
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|
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|
2006-06-22 16:27:03 +00:00
|
|
|
2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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* gst/matroska/matroska-demux.c:
|
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|
(gst_matroska_demux_check_subtitle_buffer),
|
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|
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
|
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|
|
(gst_matroska_demux_subtitle_caps):
|
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|
|
* gst/matroska/matroska-ids.c:
|
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|
|
(gst_matroska_track_init_subtitle_context):
|
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|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Try to fix up broken matroska files containing subtitle
|
|
|
|
streams with non-UTF8 character encodings (courtesy of
|
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|
|
mkvmerge) using either the encoding specified in the
|
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|
|
GST_SUBTITLE_ENCODING environment variable or the
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|
|
current locale's character set if it is non-UTF8.
|
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|
Fixes #337076.
|
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|
2006-06-22 12:17:13 +00:00
|
|
|
2006-06-22 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (parse_picture_frame):
|
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|
Set image type from APIC frame as "image-type" field
|
|
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|
of GST_TAG_IMAGE buffer caps (#344605).
|
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|
|
|
ext/flac/: Support chain-based operation, should make flac-over-DAAP work (#340492).
Original commit message from CVS:
* ext/flac/Makefile.am:
* ext/flac/gstflacdec.c: (gst_flac_dec_init),
(gst_flac_dec_reset_decoders),
(gst_flac_dec_setup_seekable_decoder),
(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize),
(gst_flac_dec_metadata_callback),
(gst_flac_dec_metadata_callback_seekable),
(gst_flac_dec_metadata_callback_stream),
(gst_flac_dec_error_callback),
(gst_flac_dec_error_callback_seekable),
(gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable),
(gst_flac_dec_read_stream), (gst_flac_dec_write),
(gst_flac_dec_write_seekable), (gst_flac_dec_write_stream),
(gst_flac_dec_loop), (gst_flac_dec_sink_event),
(gst_flac_dec_chain), (gst_flac_dec_convert_sink),
(gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query),
(gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
(gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate),
(gst_flac_dec_sink_activate_push),
(gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
* ext/flac/gstflacdec.h:
Support chain-based operation, should make flac-over-DAAP
work (#340492).
2006-06-20 19:40:29 +00:00
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|
2006-06-20 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
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* ext/flac/Makefile.am:
|
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|
* ext/flac/gstflacdec.c: (gst_flac_dec_init),
|
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|
(gst_flac_dec_reset_decoders),
|
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|
|
(gst_flac_dec_setup_seekable_decoder),
|
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|
|
(gst_flac_dec_setup_stream_decoder), (gst_flac_dec_finalize),
|
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|
(gst_flac_dec_metadata_callback),
|
|
|
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(gst_flac_dec_metadata_callback_seekable),
|
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|
|
(gst_flac_dec_metadata_callback_stream),
|
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|
(gst_flac_dec_error_callback),
|
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(gst_flac_dec_error_callback_seekable),
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|
|
(gst_flac_dec_error_callback_stream), (gst_flac_dec_read_seekable),
|
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|
|
(gst_flac_dec_read_stream), (gst_flac_dec_write),
|
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|
|
(gst_flac_dec_write_seekable), (gst_flac_dec_write_stream),
|
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|
|
(gst_flac_dec_loop), (gst_flac_dec_sink_event),
|
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|
|
(gst_flac_dec_chain), (gst_flac_dec_convert_sink),
|
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|
(gst_flac_dec_get_sink_query_types), (gst_flac_dec_sink_query),
|
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(gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
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(gst_flac_dec_handle_seek_event), (gst_flac_dec_sink_activate),
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(gst_flac_dec_sink_activate_push),
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(gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
|
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|
* ext/flac/gstflacdec.h:
|
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|
Support chain-based operation, should make flac-over-DAAP
|
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|
work (#340492).
|
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|
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|
2006-06-20 15:35:05 +00:00
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2006-06-20 Wim Taymans <wim@fluendo.com>
|
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|
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|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
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|
Doc updates, merge some unused symbols.
|
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2006-06-20 14:57:09 +00:00
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|
2006-06-20 Wim Taymans <wim@fluendo.com>
|
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* docs/plugins/Makefile.am:
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|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
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|
|
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
|
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|
* gst/rtsp/gstrtspsrc.c:
|
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|
|
* gst/rtsp/gstrtspsrc.h:
|
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|
|
Added documentation for the rtsp plugin. Fixes #345393.
|
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|
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|
2006-06-20 12:10:29 +00:00
|
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|
2006-06-20 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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|
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
|
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(rtsp_connection_close), (rtsp_connection_free):
|
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|
Use better G_OS_* macros. Fixes #345301 some more.
|
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|
|
|
sys/sunaudio/: Add a SunAudio source plugin.
Original commit message from CVS:
Patch by: Brian Cameron <brian dot cameron at sun dot com>
* sys/sunaudio/Makefile.am:
* sys/sunaudio/gstsunaudio.c: (plugin_init):
* sys/sunaudio/gstsunaudiomixerctrl.c:
(gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
(gst_sunaudiomixer_ctrl_list_tracks),
(gst_sunaudiomixer_ctrl_get_volume),
(gst_sunaudiomixer_ctrl_set_volume),
(gst_sunaudiomixer_ctrl_set_mute),
(gst_sunaudiomixer_ctrl_set_record):
* sys/sunaudio/gstsunaudiomixerctrl.h:
* sys/sunaudio/gstsunaudiomixertrack.c:
(gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
* sys/sunaudio/gstsunaudiomixertrack.h:
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
(gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
(gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
(gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
(gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
(gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
(gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
(gst_sunaudiosrc_reset):
* sys/sunaudio/gstsunaudiosrc.h:
Add a SunAudio source plugin.
Support stereo and right/left channel gain in the mixer plugin.
Support the RECORD flag so that you can switch between line-input and
microphone in gnome-volume-control.
Code cleanups like using an enumerator for track number instead of an
integer. Fixes #344923.
2006-06-20 10:35:48 +00:00
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|
2006-06-20 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
|
|
|
|
|
|
|
* sys/sunaudio/Makefile.am:
|
|
|
|
* sys/sunaudio/gstsunaudio.c: (plugin_init):
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_build_list), (gst_sunaudiomixer_ctrl_new),
|
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|
|
(gst_sunaudiomixer_ctrl_list_tracks),
|
|
|
|
(gst_sunaudiomixer_ctrl_get_volume),
|
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|
|
(gst_sunaudiomixer_ctrl_set_volume),
|
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|
|
(gst_sunaudiomixer_ctrl_set_mute),
|
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|
|
(gst_sunaudiomixer_ctrl_set_record):
|
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|
|
* sys/sunaudio/gstsunaudiomixerctrl.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.c:
|
|
|
|
(gst_sunaudiomixer_track_init), (gst_sunaudiomixer_track_new):
|
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|
|
* sys/sunaudio/gstsunaudiomixertrack.h:
|
|
|
|
* sys/sunaudio/gstsunaudiosrc.c: (gst_sunaudiosrc_dispose),
|
|
|
|
(gst_sunaudiosrc_base_init), (gst_sunaudiosrc_class_init),
|
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|
|
(gst_sunaudiosrc_init), (gst_sunaudiosrc_set_property),
|
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|
|
(gst_sunaudiosrc_get_property), (gst_sunaudiosrc_getcaps),
|
|
|
|
(gst_sunaudiosrc_open), (gst_sunaudiosrc_close),
|
|
|
|
(gst_sunaudiosrc_prepare), (gst_sunaudiosrc_unprepare),
|
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|
|
(gst_sunaudiosrc_read), (gst_sunaudiosrc_delay),
|
|
|
|
(gst_sunaudiosrc_reset):
|
|
|
|
* sys/sunaudio/gstsunaudiosrc.h:
|
|
|
|
Add a SunAudio source plugin.
|
|
|
|
Support stereo and right/left channel gain in the mixer plugin.
|
|
|
|
Support the RECORD flag so that you can switch between line-input and
|
|
|
|
microphone in gnome-volume-control.
|
|
|
|
Code cleanups like using an enumerator for track number instead of an
|
|
|
|
integer. Fixes #344923.
|
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|
|
|
2006-06-20 10:31:41 +00:00
|
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|
2006-06-20 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Joni Valtanen <joni dot valtanen at movial dot fi>
|
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|
|
|
|
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|
* gst/rtsp/rtspconnection.c: (inet_aton), (rtsp_connection_send),
|
|
|
|
(rtsp_connection_close):
|
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|
|
Make RTSP plugin compile on windows. Fixes #345301.
|
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|
|
Some changes to original patch to catch errors better.
|
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|
|
use ifdef WIN32 instead of ifndef.
|
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|
2006-06-19 10:00:18 +00:00
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|
2006-06-19 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
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|
|
|
|
|
|
* configure.ac:
|
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|
|
If we have libraw1394 >= 1.2.1, then we need libiec61883.
|
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|
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|
2006-06-18 14:00:19 +00:00
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|
2006-06-18 Edward Hervey <edward@fluendo.com>
|
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|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
|
|
|
|
After a failed buffer alloc, we need to abort the jpeg decoding (it
|
|
|
|
started when parsing headers to figure out how many bytes we need
|
|
|
|
to request downstream).
|
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|
|
|
2006-06-18 12:37:12 +00:00
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|
2006-06-18 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
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|
Patch by: Mark Nauwelaerts <manauw at skynet be>
|
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|
|
|
|
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|
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
|
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Make sure we don't read beyond the end of the file (#345232).
|
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|
|
|
2006-06-17 14:35:37 +00:00
|
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|
2006-06-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
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|
* configure.ac:
|
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|
Fix --disable-external (can't set conditionals conditionally,
|
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|
|
#343602).
|
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|
|
|
2006-06-16 10:56:24 +00:00
|
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|
2006-06-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
* configure.ac:
|
|
|
|
* docs/Makefile.am:
|
|
|
|
Use GST_PLUGIN_DOCS, --enable-plugin-docs etc.
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/inspect/plugin-taglib.xml:
|
|
|
|
Add/fix apev2mux docs.
|
|
|
|
|
2006-06-14 15:59:56 +00:00
|
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|
2006-06-14 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_class_init), (gst_dvdec_init),
|
|
|
|
(gst_dvdec_finalize), (gst_dvdec_sink_event),
|
|
|
|
(gst_dvdec_change_state):
|
|
|
|
Reset segment info on flush.
|
|
|
|
Alloc segment in _init, free in _finalize.
|
|
|
|
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_pull_seek):
|
|
|
|
Don't send segments twice.
|
|
|
|
|
2006-06-14 15:07:22 +00:00
|
|
|
2006-06-14 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
|
|
|
|
Respect segment.stop. Fixes #342592.
|
|
|
|
|
2006-06-14 11:28:41 +00:00
|
|
|
2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream):
|
|
|
|
No language specified means the implied language is English
|
|
|
|
according to the matroska spec (partially fixes #344708);
|
|
|
|
add some more debug output.
|
|
|
|
|
2006-06-14 08:06:43 +00:00
|
|
|
2006-06-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_peek_chunk_info),
|
|
|
|
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
|
|
|
|
(gst_wavparse_chain):
|
|
|
|
When operating chain-based, don't make any assumptions about the
|
|
|
|
chunking of the incoming data and make streaming work on days other
|
|
|
|
than the second Thursday after a full moon. Also fix up debug
|
|
|
|
messages here and there and make use of the most excellent new
|
|
|
|
gst_pad_query_peer_duration() utility function.
|
|
|
|
Skip any 'bext' chunks in front of the 'fmt ' chunk. Fixes #343837.
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
Remove trailing comma after last enum value, some compilers don't
|
|
|
|
like that.
|
|
|
|
|
2006-06-13 17:05:25 +00:00
|
|
|
2006-06-13 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_data):
|
|
|
|
Handle premature EOS gracefully.
|
|
|
|
|
2006-06-13 09:54:26 +00:00
|
|
|
2006-06-13 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
|
|
|
|
Prevent out of bounds array access when scrubbing towards
|
|
|
|
the end of the file between the last index entry and the
|
|
|
|
end. Fixes occasional 'start <= stop' newsegment event
|
|
|
|
assertions when scrubbing in MJPEG files.
|
|
|
|
|
2006-06-12 11:13:39 +00:00
|
|
|
2006-06-12 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* tests/check/elements/.cvsignore:
|
|
|
|
And another one.
|
|
|
|
|
2006-06-11 19:31:10 +00:00
|
|
|
2006-06-11 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(scan_encoded_string), (parse_picture_frame):
|
|
|
|
Extract images from ID3v2 tags (APIC frames). Fixes #339704.
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Require core >= 0.10.8 (for GST_TAG_IMAGE and
|
|
|
|
GST_TAG_PPEVIEW_IMAGE used in the patch above).
|
|
|
|
|
2006-06-11 18:52:19 +00:00
|
|
|
2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_discover_avc_node):
|
|
|
|
gratuitous comment changes
|
|
|
|
* tests/check/elements/level.c: (GST_START_TEST):
|
|
|
|
fix level test leaks
|
|
|
|
|
2006-06-11 18:20:39 +00:00
|
|
|
2006-06-11 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size):
|
|
|
|
* gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size):
|
|
|
|
Use gst_pad_query_peer_duration() utility function here.
|
|
|
|
|
2006-06-11 13:57:19 +00:00
|
|
|
2006-06-11 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
require am17
|
|
|
|
* configure.ac:
|
|
|
|
* ext/annodex/Makefile.am:
|
|
|
|
* ext/cdio/Makefile.am:
|
|
|
|
* ext/dv/Makefile.am:
|
|
|
|
* ext/esd/Makefile.am:
|
|
|
|
* ext/flac/Makefile.am:
|
|
|
|
* ext/gdk_pixbuf/Makefile.am:
|
|
|
|
* ext/ladspa/Makefile.am:
|
|
|
|
* ext/libcaca/Makefile.am:
|
|
|
|
* ext/speex/Makefile.am:
|
|
|
|
* ext/taglib/Makefile.am:
|
|
|
|
* sys/oss/Makefile.am:
|
|
|
|
* sys/sunaudio/Makefile.am:
|
|
|
|
* sys/ximage/Makefile.am:
|
|
|
|
clean up build further
|
|
|
|
|
2006-06-09 20:36:00 +00:00
|
|
|
2006-06-09 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gconf/Makefile.am:
|
|
|
|
Honour --disable-schemas-install configure option. Fixes #344100.
|
|
|
|
|
2006-06-09 18:33:01 +00:00
|
|
|
2006-06-09 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* tests/examples/level/Makefile.am:
|
|
|
|
Add -lm to LIBS for pow() function, don't assume one of our
|
|
|
|
dependencies (such as libxml-2.0) drags it in automatically
|
|
|
|
(#343603).
|
|
|
|
|
2006-06-09 18:17:23 +00:00
|
|
|
2006-06-09 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Peter Kjellerstedt <pkj at axis dot com>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
We should use $SED and not $(SED) in configure.ac (#343678).
|
|
|
|
|
2006-06-09 17:29:08 +00:00
|
|
|
2006-06-09 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_open), (gst_sunaudiomixer_ctrl_build_list),
|
|
|
|
(gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_set_volume),
|
|
|
|
(gst_sunaudiomixer_ctrl_set_mute):
|
|
|
|
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init),
|
|
|
|
(gst_sunaudiosink_init), (gst_sunaudiosink_prepare),
|
|
|
|
(gst_sunaudiosink_write):
|
2006-08-14 11:14:43 +00:00
|
|
|
Attached find a patch that fixes a number of bugs with the SunAudio
|
|
|
|
mixer plugin and fixes #344101:
|
|
|
|
1. The gst_sunaudiomixer_ctrl_build_list kept appending the same 3
|
|
|
|
tracks onto the tracklist causing gnome-volume-control's preferences
|
|
|
|
dialog to be messed up and would core dump if you checked/unchecked
|
|
|
|
any item.
|
|
|
|
2. We weren't previously setting the MUTE flag properly. Fixing this
|
|
|
|
makes gnome-volume-control work better.
|
|
|
|
3. Now we properly define the input track to be GST_MIXER_TRACK_INPUT
|
|
|
|
and the monitor to be GST_MIXER_TRACK_OUTPUT, so that makes
|
|
|
|
gnome-volume-control look better.
|
2006-06-09 17:29:08 +00:00
|
|
|
Also some minor cleanup in gstsunaudiosink.c.
|
|
|
|
|
2006-06-09 17:12:52 +00:00
|
|
|
2006-06-09 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_idct_method_get_type),
|
|
|
|
(gst_jpeg_dec_class_init), (gst_jpeg_dec_init),
|
|
|
|
(gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_decode_direct),
|
|
|
|
(gst_jpeg_dec_chain), (gst_jpeg_dec_sink_event),
|
|
|
|
(gst_jpeg_dec_set_property), (gst_jpeg_dec_get_property):
|
|
|
|
* ext/jpeg/gstjpegdec.h:
|
|
|
|
API: Added IDCT method property
|
|
|
|
Small cleanups.
|
|
|
|
Avoid dynamic allocation of trivial fixed structure.
|
|
|
|
Allocate enough space for temp 4:4:4 YUV buffers. Fixes #343661.
|
|
|
|
|
2006-06-07 09:25:16 +00:00
|
|
|
2006-06-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
We now require libraw1394 >= 1.1.0 and that version onwards all
|
|
|
|
have .pc files.
|
|
|
|
|
2006-06-02 15:02:54 +00:00
|
|
|
2006-06-02 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/law/alaw-decode.c: (alawdec_getcaps):
|
|
|
|
Trying to get items from an ANY or EMPTY caps is ... stupid.
|
|
|
|
|
2006-06-02 11:33:18 +00:00
|
|
|
2006-06-02 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_event),
|
|
|
|
(gst_dvdec_chain), (gst_dvdec_change_state):
|
|
|
|
* ext/dv/gstdvdec.h:
|
|
|
|
Added GstSegment handling, now implements dropping/clipping.
|
|
|
|
|
2006-06-01 21:07:26 +00:00
|
|
|
2006-06-01 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.h:
|
|
|
|
* ext/annodex/gstcmmldec.h:
|
|
|
|
* ext/cairo/gsttimeoverlay.h:
|
|
|
|
* ext/dv/gstdvdec.h:
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
* ext/esd/esdmon.h:
|
|
|
|
* ext/esd/esdsink.h:
|
|
|
|
* ext/flac/gstflacenc.h:
|
|
|
|
* ext/gconf/gstgconfaudiosink.h:
|
|
|
|
* ext/gconf/gstgconfaudiosrc.h:
|
|
|
|
* ext/gconf/gstgconfvideosink.h:
|
|
|
|
* ext/gconf/gstgconfvideosrc.h:
|
|
|
|
* ext/gdk_pixbuf/gstgdkanimation.h:
|
|
|
|
* ext/gdk_pixbuf/pixbufscale.h:
|
|
|
|
* ext/hal/gsthalaudiosink.h:
|
|
|
|
* ext/hal/gsthalaudiosrc.h:
|
|
|
|
* ext/jpeg/gstjpegenc.h:
|
|
|
|
* ext/jpeg/gstsmokedec.h:
|
|
|
|
* ext/jpeg/gstsmokeenc.h:
|
|
|
|
* ext/libcaca/gstcacasink.h:
|
|
|
|
* ext/libmng/gstmngdec.h:
|
|
|
|
* ext/libmng/gstmngenc.h:
|
|
|
|
* ext/libpng/gstpngdec.h:
|
|
|
|
* ext/libpng/gstpngenc.h:
|
|
|
|
* ext/raw1394/gstdv1394src.h:
|
|
|
|
* ext/speex/gstspeexenc.h:
|
|
|
|
* gst/autodetect/gstautoaudiosink.h:
|
|
|
|
* gst/autodetect/gstautovideosink.h:
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
* gst/cutter/gstcutter.h:
|
|
|
|
* gst/debug/efence.h:
|
|
|
|
* gst/debug/gstnavigationtest.h:
|
|
|
|
* gst/debug/gstnavseek.h:
|
|
|
|
* gst/flx/gstflxdec.h:
|
|
|
|
* gst/goom/gstgoom.h:
|
|
|
|
* gst/icydemux/gsticydemux.h:
|
|
|
|
* gst/id3demux/gstid3demux.h:
|
|
|
|
* gst/law/alaw-decode.h:
|
|
|
|
* gst/law/alaw-encode.h:
|
|
|
|
* gst/law/mulaw-decode.h:
|
|
|
|
* gst/law/mulaw-encode.h:
|
|
|
|
* gst/matroska/matroska-mux.h:
|
|
|
|
* gst/median/gstmedian.h:
|
|
|
|
* gst/oldcore/gstaggregator.h:
|
|
|
|
* gst/oldcore/gstfdsink.h:
|
|
|
|
* gst/oldcore/gstmd5sink.h:
|
|
|
|
* gst/oldcore/gstmultifilesrc.h:
|
|
|
|
* gst/oldcore/gstpipefilter.h:
|
|
|
|
* gst/oldcore/gstshaper.h:
|
|
|
|
* gst/oldcore/gststatistics.h:
|
|
|
|
* gst/rtp/gstasteriskh263.h:
|
|
|
|
* gst/rtp/gstrtpL16depay.h:
|
|
|
|
* gst/rtp/gstrtpL16pay.h:
|
|
|
|
* gst/rtp/gstrtpamrdepay.h:
|
|
|
|
* gst/rtp/gstrtpamrpay.h:
|
|
|
|
* gst/rtp/gstrtpdepay.h:
|
|
|
|
* gst/rtp/gstrtpgsmdepay.h:
|
|
|
|
* gst/rtp/gstrtpgsmpay.h:
|
|
|
|
* gst/rtp/gstrtph263pay.h:
|
|
|
|
* gst/rtp/gstrtph263pdepay.h:
|
|
|
|
* gst/rtp/gstrtph263ppay.h:
|
|
|
|
* gst/rtp/gstrtpmp4gpay.h:
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.h:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.h:
|
|
|
|
* gst/rtp/gstrtpmpadepay.h:
|
|
|
|
* gst/rtp/gstrtpmpapay.h:
|
|
|
|
* gst/rtp/gstrtppcmadepay.h:
|
|
|
|
* gst/rtp/gstrtppcmapay.h:
|
|
|
|
* gst/rtp/gstrtppcmudepay.h:
|
|
|
|
* gst/rtp/gstrtppcmupay.h:
|
|
|
|
* gst/rtp/gstrtpspeexdepay.h:
|
|
|
|
* gst/rtp/gstrtpspeexpay.h:
|
|
|
|
* gst/rtsp/gstrtpdec.h:
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
* gst/smpte/gstsmpte.h:
|
|
|
|
* gst/udp/gstdynudpsink.h:
|
|
|
|
* gst/udp/gstmultiudpsink.h:
|
|
|
|
* gst/udp/gstudpsink.h:
|
|
|
|
* gst/udp/gstudpsrc.h:
|
|
|
|
* gst/videofilter/gstvideobalance.h:
|
|
|
|
* gst/videofilter/gstvideoflip.h:
|
|
|
|
* sys/oss/gstossdmabuffer.h:
|
|
|
|
* sys/oss/gstossmixerelement.h:
|
|
|
|
* sys/oss/gstosssink.h:
|
|
|
|
* sys/oss/gstosssrc.h:
|
|
|
|
* sys/osxvideo/osxvideosink.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixer.h:
|
|
|
|
* sys/sunaudio/gstsunaudiosink.h:
|
|
|
|
* sys/ximage/gstximagesrc.h:
|
|
|
|
Fix more gobject macros: obj<->klass, GstXXX<->GstXXXClass
|
|
|
|
|
gst/goom/gstgoom.*: Handle QoS.
Original commit message from CVS:
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
(gst_goom_finalize), (gst_goom_reset), (gst_goom_sink_setcaps),
(gst_goom_src_setcaps), (gst_goom_src_event),
(gst_goom_sink_event), (get_buffer), (gst_goom_chain),
(gst_goom_change_state):
* gst/goom/gstgoom.h:
Handle QoS.
Handle flushing, discont and events.
Fix timestamps and various other cleanups.
2006-05-31 16:23:54 +00:00
|
|
|
2006-05-31 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
|
|
|
|
(gst_goom_finalize), (gst_goom_reset), (gst_goom_sink_setcaps),
|
|
|
|
(gst_goom_src_setcaps), (gst_goom_src_event),
|
|
|
|
(gst_goom_sink_event), (get_buffer), (gst_goom_chain),
|
|
|
|
(gst_goom_change_state):
|
|
|
|
* gst/goom/gstgoom.h:
|
|
|
|
Handle QoS.
|
|
|
|
Handle flushing, discont and events.
|
|
|
|
Fix timestamps and various other cleanups.
|
|
|
|
|
2006-05-31 15:37:16 +00:00
|
|
|
2006-05-31 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_bus_reset):
|
|
|
|
Fix bus reset when using libiec61883
|
|
|
|
|
|
|
|
2006-05-31 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Detect libiec61883 and set necessary CFLAGS and LIBS for dv1394.
|
|
|
|
* ext/raw1394/Makefile.am:
|
|
|
|
Add CFLAGS.
|
|
|
|
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_iec61883_receive),
|
|
|
|
New method, to receive using libiec61883.
|
|
|
|
(gst_dv1394src_iso_receive),
|
|
|
|
#ifdef'd out if libiec61883 is present.
|
|
|
|
(gst_dv1394src_bus_reset),
|
|
|
|
Get userdata correctly if using libiec61883.
|
|
|
|
(gst_dv1394src_create),
|
|
|
|
When using libiec61883, only poll one fd and no need to read.
|
|
|
|
(gst_dv1394src_discover_avc_node),
|
|
|
|
Replace g_warnings.
|
|
|
|
(gst_dv1394src_start),
|
|
|
|
Create new handle when we know which dv port. More reliable
|
|
|
|
than setting port on an existing handle. Initialise libiec61883.
|
|
|
|
(gst_dv1394src_stop):
|
|
|
|
If using libiec61883, then cleanup its handle properly.
|
|
|
|
* ext/raw1394/gstdv1394src.h:
|
|
|
|
Add libiec61883 handle.
|
|
|
|
|
2006-05-30 21:07:38 +00:00
|
|
|
2006-05-30 Sebastien Moutte <sebastien@moutte.net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c:
|
|
|
|
add an explicit dll imported declaration for GST_CAT_EVENT+WIN32
|
|
|
|
* win32/MANIFEST:
|
|
|
|
sort file listing
|
|
|
|
* win32/vs6/libgstavi.dsp:
|
|
|
|
add gstavimux.c to the project
|
|
|
|
* win32/vs6/libgstid3demux.dsp:
|
|
|
|
add link to zlib library
|
|
|
|
* win32/vs6/libgstmatroska.dsp:
|
|
|
|
add matroska-ids.c to the project
|
|
|
|
|
2006-05-30 14:35:18 +00:00
|
|
|
2006-05-30 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Sebastian Dröge <mail at slomosnail de >
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* ext/taglib/Makefile.am:
|
|
|
|
* ext/taglib/gstapev2mux.cc:
|
|
|
|
* ext/taglib/gstapev2mux.h:
|
|
|
|
* ext/taglib/gstid3v2mux.cc:
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* ext/taglib/gsttaglibmux.c: (plugin_init):
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* ext/taglib/gsttaglibmux.h:
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Add apev2mux element (#343122).
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* tests/check/Makefile.am:
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* tests/check/elements/apev2mux.c:
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(test_taglib_apev2mux_create_tags),
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(test_taglib_apev2mux_check_tags), (fill_mp3_buffer), (got_buffer),
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(demux_pad_added), (test_taglib_apev2mux_check_output_buffer),
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(test_taglib_apev2mux_with_tags), (GST_START_TEST),
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(apev2mux_suite), (main):
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Add unit test for apev2mux element.
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2006-05-28 17:33:13 +00:00
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2006-05-28 Tim-Philipp Müller <tim at centricular dot net>
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
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* gst/debug/negotiation.c: (gst_negotiation_update_caps):
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* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
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GST_PTR_FORMAT should be used to print caps in debug statements.
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2006-05-28 14:38:11 +00:00
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2006-05-28 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Sebastian Dröge <slomo at ubuntu dot com>
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* gst/apetag/gstapedemux.c: (ape_demux_get_gst_tag_from_tag),
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(ape_demux_parse_tags):
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Some clean-ups and additions: map APE 'file' tag to
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GST_TAG_LOCATION (#343123); add support for extracting
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the track count and clean up parsing a bit (#343127).
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2006-05-28 13:49:12 +00:00
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2006-05-28 Edward Hervey <edward@fluendo.com>
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_change_state):
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Initialize segment to GST_FORMAT_UNDEFINED in READY->PAUSED.
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2006-05-28 13:30:13 +00:00
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2006-05-28 Edward Hervey <edward@fluendo.com>
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_finalize),
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(gst_jpeg_dec_init), (gst_jpeg_dec_chain),
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(gst_jpeg_dec_sink_event), (gst_jpeg_dec_change_state):
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* ext/jpeg/gstjpegdec.h:
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|
Clip outgoing buffers according to currently configured segment.
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2006-05-28 10:39:00 +00:00
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2006-05-28 Tim-Philipp Müller <tim at centricular dot net>
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* ext/taglib/gstid3v2mux.cc:
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Handle writing of track-count or album-volume-count without
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track-number or albume-volume-number (in this case the number
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will just be set to 0).
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* tests/check/elements/id3v2mux.c: (test_taglib_id3mux_check_tags):
|
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|
It would be nice if we actually checked the values received for
|
|
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|
track/album-volume number/count in _check_tags(), rather than
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setting them again ...
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|
2006-05-28 10:05:47 +00:00
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2006-05-28 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
|
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A track/volume number or count of 0 does not make sense,
|
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|
just ignore it along with negative numbers (a tag might
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|
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|
only contain a track count without a track number).
|
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|
2006-05-27 13:11:37 +00:00
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2006-05-27 Edward Hervey <edward@fluendo.com>
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
|
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(gst_jpeg_dec_sink_event):
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Abort decompression when receiving FLUSH_STOP. This should avoid
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issues when interrupting decoding with flushes.
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2006-05-27 12:10:50 +00:00
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2006-05-27 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflac.c:
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Don't #include file we don't dist any longer.
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2006-05-27 11:27:59 +00:00
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2006-05-27 Tim-Philipp Müller <tim at centricular dot net>
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* README:
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Replace current README (containing the release notes from
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some 0.9.x version) with a proper README taken from the core.
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2006-05-24 16:03:40 +00:00
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2006-05-24 Wim Taymans <wim@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_loop):
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Implement EOS correctly by either posting
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SEGMENT_DONE or pushing an EOS message depending
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|
on the seek type. Fixes #342592
|
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2006-05-24 10:00:50 +00:00
|
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2006-05-24 Wim Taymans <wim@fluendo.com>
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* gst/law/alaw-decode.c: (gst_alawdec_chain):
|
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* gst/law/alaw-decode.h:
|
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|
* gst/law/alaw-encode.c: (gst_alawenc_chain):
|
|
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|
* gst/law/alaw-encode.h:
|
|
|
|
* gst/law/mulaw-decode.c: (gst_mulawdec_chain):
|
|
|
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* gst/law/mulaw-decode.h:
|
|
|
|
* gst/law/mulaw-encode.c: (gst_mulawenc_chain):
|
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|
|
* gst/law/mulaw-encode.h:
|
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|
|
Some cleanups in the chain functions.
|
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|
Remove some GStreamer 0.0.2 bits.
|
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|
2006-05-23 20:15:04 +00:00
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2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
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|
Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_change_state):
|
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|
gst_collect_pads_stop() needs to be called before chaining up
|
|
|
|
to the parent class (#342734).
|
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|
2006-05-23 16:45:22 +00:00
|
|
|
2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
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|
|
* ext/flac/Makefile.am:
|
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|
|
* ext/flac/flac_compat.h:
|
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|
|
* ext/flac/gstflac.c:
|
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|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_init):
|
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|
|
* ext/flac/gstflacenc.c:
|
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|
|
Remove backwards compatibility cruft for dealing with FLAC API
|
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|
|
changes in the 1.0.x series - we require 1.1.1 or newer these days.
|
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|
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|
gst/matroska/: Add support for muxing/demuxing theora video (#342448; too bad none of the usual linux players can act...
Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
(gst_matroska_demux_push_xiph_codec_priv_data),
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
* gst/matroska/matroska-ids.h:
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
(gst_matroska_mux_video_pad_setcaps),
(xiph3_streamheader_to_codecdata),
(vorbis_streamheader_to_codecdata),
(theora_streamheader_to_codecdata),
(gst_matroska_mux_audio_pad_setcaps),
(gst_matroska_mux_write_data):
Add support for muxing/demuxing theora video (#342448; too bad
none of the usual linux players can actually play this). Playback
in GStreamer will require additional changes to theoradec in -base.
Refactor streamheaders <=> CodecPrivateData code a bit; some small
cleanups.
2006-05-23 13:44:11 +00:00
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2006-05-23 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
|
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|
(gst_matroska_demux_push_xiph_codec_priv_data),
|
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|
|
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
|
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|
(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
|
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* gst/matroska/matroska-ids.h:
|
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|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init),
|
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|
(gst_matroska_mux_video_pad_setcaps),
|
|
|
|
(xiph3_streamheader_to_codecdata),
|
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|
(vorbis_streamheader_to_codecdata),
|
|
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|
(theora_streamheader_to_codecdata),
|
|
|
|
(gst_matroska_mux_audio_pad_setcaps),
|
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|
(gst_matroska_mux_write_data):
|
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|
Add support for muxing/demuxing theora video (#342448; too bad
|
|
|
|
none of the usual linux players can actually play this). Playback
|
|
|
|
in GStreamer will require additional changes to theoradec in -base.
|
|
|
|
Refactor streamheaders <=> CodecPrivateData code a bit; some small
|
|
|
|
cleanups.
|
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|
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|
2006-05-22 15:23:05 +00:00
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2006-05-22 Tim-Philipp Müller <tim at centricular dot net>
|
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* ext/jpeg/gstjpegdec.c: (hresamplecpy1),
|
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(gst_jpeg_dec_decode_indirect), (gst_jpeg_dec_chain):
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Fix crashes when the horizontal subsampling is 1.
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Fixes #342097.
|
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|
gst/avi/gstavimux.*: - add odml (large file) index support
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet be>
* gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_write_avix_index),
(gst_avi_mux_add_index), (gst_avi_mux_bigfile),
(gst_avi_mux_start_file), (gst_avi_mux_stop_file),
(gst_avi_mux_handle_event), (gst_avi_mux_do_audio_buffer),
(gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
Some enhancements for avimux (#342526):
- add odml (large file) index support
- store codec init data (e.g. huffyuv)
- miscellaneous other fixes/cleanups
2006-05-22 13:53:18 +00:00
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2006-05-22 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
|
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* gst/avi/gstavimux.c: (gst_avi_mux_finalize), (gst_avi_mux_init),
|
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|
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
|
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|
|
(gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
|
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|
|
(gst_avi_mux_riff_get_avix_header), (gst_avi_mux_write_avix_index),
|
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|
|
(gst_avi_mux_add_index), (gst_avi_mux_bigfile),
|
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|
(gst_avi_mux_start_file), (gst_avi_mux_stop_file),
|
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|
(gst_avi_mux_handle_event), (gst_avi_mux_do_audio_buffer),
|
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|
(gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
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(gst_avi_mux_change_state):
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|
* gst/avi/gstavimux.h:
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|
Some enhancements for avimux (#342526):
|
|
|
|
- add odml (large file) index support
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|
|
- store codec init data (e.g. huffyuv)
|
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|
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- miscellaneous other fixes/cleanups
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|
|
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2006-05-19 14:05:53 +00:00
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2006-05-19 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
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Don't output any tag when we encounter a negative track number - the
|
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|
tag type is uint, so we end up outputting huge positive numbers
|
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|
instead. (Fixes: #342029)
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2006-05-18 23:04:59 +00:00
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2006-05-19 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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|
|
update for new GSTPB_PLUGINS_DIR
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2006-05-18 14:45:33 +00:00
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|
2006-05-18 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
|
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* rtp/gst/gstrtph263pay.c:
|
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|
|
Properly set static caps for H263 at 34.
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|
2006-05-18 12:46:08 +00:00
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|
2006-05-18 Tim-Philipp Müller <tim at centricular dot net>
|
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|
Patch by: James "Doc" Livingston <doclivingston gmail com>
|
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* ext/taglib/gsttaglibmux.c: (gst_tag_lib_mux_render_tag):
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Merge event tags and tag setter tags correctly (#339918). Also,
|
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|
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don't leak taglist in case of an error.
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2006-05-17 12:36:26 +00:00
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2006-05-17 Edward Hervey <edward@fluendo.com>
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* gst/law/mulaw-decode.c: (mulawdec_getcaps):
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We can only do caps intersection if the othercaps are non-empty and not
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|
|
ANY. Else we return the pad template (base_caps).
|
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2006-05-17 11:20:44 +00:00
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2006-05-17 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
|
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|
|
Fix crash when outputting debugging information for certain
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|
|
pictures (always good to use the right struct member for
|
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|
|
the number of records in an array).
|
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|
|
|
gst/matroska/ebml-read.c: Don't create unnecessary sub-buffers all the time. Dramatically improves performance with m...
Original commit message from CVS:
Patch by: Jindrich Makovicka <jindrich.makivicka at itonis tv>
* gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
(gst_ebml_read_pull_bytes), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_buffer),
(gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint),
(gst_ebml_read_float), (gst_ebml_read_ascii),
(gst_ebml_read_binary):
Don't create unnecessary sub-buffers all the time. Dramatically
improves performance with multiple concurrently running
matroskademux instances (#341818) (and avoids doing
unnecessarily inefficient things in the general case).
2006-05-17 08:10:31 +00:00
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|
2006-05-16 Tim-Philipp Müller <tim at centricular dot net>
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|
|
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|
Patch by: Jindrich Makovicka <jindrich.makivicka at itonis tv>
|
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|
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|
|
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* gst/matroska/ebml-read.c: (gst_ebml_read_peek_bytes),
|
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|
|
(gst_ebml_read_pull_bytes), (gst_ebml_read_element_id),
|
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|
|
(gst_ebml_read_element_length), (gst_ebml_read_buffer),
|
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|
|
(gst_ebml_read_bytes), (gst_ebml_read_uint), (gst_ebml_read_sint),
|
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|
|
(gst_ebml_read_float), (gst_ebml_read_ascii),
|
|
|
|
(gst_ebml_read_binary):
|
|
|
|
Don't create unnecessary sub-buffers all the time. Dramatically
|
|
|
|
improves performance with multiple concurrently running
|
|
|
|
matroskademux instances (#341818) (and avoids doing
|
|
|
|
unnecessarily inefficient things in the general case).
|
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|
|
|
2006-05-16 17:20:04 +00:00
|
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|
2006-05-16 Edward Hervey <edward@fluendo.com>
|
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|
|
|
|
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|
* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
|
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|
|
In snapshot mode, we always return GST_FLOW_UNEXPECTED whatever the
|
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|
|
return value of gst_pad_push_event().
|
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|
2006-05-16 14:07:29 +00:00
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|
2006-05-16 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
|
|
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* gst/autodetect/gstautoaudiosink.c:
|
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|
|
(gst_auto_audio_sink_find_best):
|
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|
|
* gst/autodetect/gstautovideosink.c:
|
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(gst_auto_video_sink_find_best):
|
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|
Make the name of the child element be based on the name of the
|
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|
|
parent, so that debug output is more useful.
|
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* gst/id3demux/id3v2frames.c: (find_utf16_bom),
|
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|
(parse_insert_string_field), (parse_split_strings):
|
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|
Rework string parsing to always walk over BOM markers in UTF16
|
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|
|
strings, using the endianness indicated by the innermost one,
|
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|
|
then trying the opposite endianness if that fails to convert
|
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|
|
to valid UTF-8. Fixes #341774
|
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2006-05-16 13:31:02 +00:00
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2006-05-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
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|
Patch from: Matthieu <matthieu at fluendo dot com>
|
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* ext/libpng/Makefile.am:
|
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Add LIBPNG_CFLAGS.
|
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2006-05-15 09:00:42 +00:00
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2006-05-15 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
* ext/taglib/gstid3v2mux.cc:
|
|
|
|
Add support for writing images (APIC frames) into ID3v2
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|
|
tags (picture type always set to 'other' for now though).
|
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|
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2006-05-14 12:50:07 +00:00
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|
2006-05-14 Michael Smith <msmith@fluendo.com>
|
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* gst/wavparse/gstwavparse.c:
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Update docs; wavparse implements push and pull modes.
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2006-05-12 18:10:36 +00:00
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2006-05-12 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_index_next),
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(gst_avi_demux_parse_index), (gst_avi_demux_massage_index),
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(gst_avi_demux_handle_seek), (gst_avi_demux_loop):
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Ooops, bitten by the copy-and-paste design paradigm, fixes
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seek again.
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gst/avi/gstavidemux.*: Some cleanups, prepare to use GstSegment.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_index_next), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_subindex),
(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
(gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index),
(gst_avi_demux_calculate_durations_from_index),
(gst_avi_demux_push_event), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Some cleanups, prepare to use GstSegment.
Fix error in entry walking code.
Fix VBR detection.
Smarter timestamp calculation code.
Uniform error/eos handling.
2006-05-12 18:04:22 +00:00
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2006-05-12 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
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(gst_avi_demux_index_next), (gst_avi_demux_handle_src_query),
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(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_subindex),
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(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
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(gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
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(gst_avi_demux_massage_index),
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(gst_avi_demux_calculate_durations_from_index),
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(gst_avi_demux_push_event), (gst_avi_demux_stream_header),
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(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
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(gst_avi_demux_loop):
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* gst/avi/gstavidemux.h:
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Some cleanups, prepare to use GstSegment.
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Fix error in entry walking code.
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Fix VBR detection.
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Smarter timestamp calculation code.
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Uniform error/eos handling.
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2006-05-12 17:44:15 +00:00
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2006-05-12 Michael Smith <msmith@fluendo.com>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt),
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(gst_wavparse_perform_seek), (gst_wavparse_stream_headers):
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Fix use of uninitialised values if we're NOT seeking in ready.
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Fix typos.
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2006-05-12 08:23:18 +00:00
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2006-05-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavparse/Makefile.am:
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Add CFLAGS and LIBS for libgstbase, fixes build on
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Cygwin (#341489).
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2006-05-12 08:21:37 +00:00
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2006-05-10 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3v2frames.c: (parse_insert_string_field):
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Some more debug info. No need to check whether the string
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returned by g_convert() is really UTF-8 - either it is or
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we get NULL returned.
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2006-05-10 13:51:01 +00:00
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2006-05-10 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
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Fix parsing of numeric genre strings some more, by ensuring that
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we only try and parse strings that a) Start with '(' and b) Consist
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only of digits.
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Also, when finding an escaping '((' sequence, bust it back to '(' by
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swallowing the first parenthesis
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2006-05-10 11:17:31 +00:00
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2006-05-10 Tim-Philipp Müller <tim at centricular dot net>
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* ext/esd/esdsink.c: (gst_esdsink_finalize), (gst_esdsink_getcaps),
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(gst_esdsink_open), (gst_esdsink_close):
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* ext/esd/esdsink.h:
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Move the esd_get_server_info() into gst_esdsink_open() and fail
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with a decent error message on errors.
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2006-05-10 10:29:54 +00:00
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2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
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* ext/esd/esdmon.c: (gst_esdmon_depths_get_type),
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(gst_esdmon_channels_get_type):
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_profile_get_type):
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* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_method_get_type):
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* ext/libcaca/gstcacasink.c: (gst_cacasink_dither_get_type):
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* ext/shout2/gstshout2.c: (gst_shout2send_protocol_get_type):
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* gst/alpha/gstalpha.c: (gst_alpha_method_get_type):
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* gst/rtp/gstrtpilbcdepay.c: (gst_ilbc_mode_get_type):
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* gst/rtsp/gstrtspsrc.c: (gst_rtsp_proto_get_type):
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* gst/videobox/gstvideobox.c: (gst_video_box_fill_get_type):
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* gst/videofilter/gstvideoflip.c: (gst_video_flip_method_get_type):
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* gst/videomixer/videomixer.c:
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(gst_video_mixer_background_get_type):
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Const-ify GEnumValue arrays.
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2006-05-09 14:08:15 +00:00
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2006-05-09 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet bet>
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* gst/avi/gstavimux.c: (gst_avi_mux_do_audio_buffer),
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(gst_avi_mux_do_video_buffer):
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Work around gst_buffer_make_metadata_writable() bug that
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results in avimux marking all frames in the index as
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keyframes (#340859).
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2006-05-08 14:35:20 +00:00
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2006-05-08 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/rtspurl.c: (rtsp_url_parse):
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Make parsing of urls suck slightly less.
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2006-05-08 11:53:03 +00:00
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2006-05-08 Edward Hervey <edward@fluendo.com>
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* autogen.sh: (CONFIGURE_DEF_OPT):
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libtoolize on Darwin/MacOSX is called glibtoolize.
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2006-05-08 10:59:05 +00:00
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2006-05-08 Wim Taymans <wim@fluendo.com>
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Patch by: Jens Granseuer <jensgr at gmx dot net>
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* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_init):
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* gst/wavparse/gstwavparse.c: (gst_wavparse_dispose):
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C89 compliance fixes. Fixes #340980
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2006-05-06 09:01:34 +00:00
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2006-05-06 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
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* ext/flac/gstflacdec.h:
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Handle segment seeks that include the end of the file as stop point
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properly: when the decoder hits EOS we want to send a SEGMENT_DONE
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message instead of an EOS event in case we're in segment seek
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mode (fixes #340699).
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2006-05-06 00:14:09 +00:00
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2006-05-05 Maciej Katafiasz <mathrick@freedesktop.org>
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* ext/cairo/gsttextoverlay.c:
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* ext/flac/gstflacdec.c:
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* ext/gdk_pixbuf/pixbufscale.c:
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* gst/apetag/gstapedemux.c:
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* gst/debug/breakmydata.c:
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* gst/debug/testplugin.c:
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* gst/matroska/ebml-write.c:
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* gst/multipart/multipartdemux.c:
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* sys/osxaudio/gstosxaudiosink.c:
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* sys/osxaudio/gstosxaudiosrc.c:
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Add semicolons after GST_BOILERPLATE[_FULL] so that
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indent doesn't mess up following lines.
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2006-05-04 17:27:27 +00:00
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2006-05-04 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Michal Benes <michal dot benes at xeris dot cz>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_reset):
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Don't leak caps when freeing the stream context (#340623).
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2006-05-04 15:40:18 +00:00
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2006-05-04 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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Back to CVS
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2006-05-04 15:05:00 +00:00
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=== release 0.10.3 ===
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2006-05-04 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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releasing 0.10.3, "Desplazado"
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2006-05-03 18:41:47 +00:00
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2006-05-03 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_stream_is_vorbis_header),
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(gst_matroska_mux_write_data):
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Don't strcmp() NULL strings.
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Only start new clusters on video keyframes, not on any
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random audio buffer that doesn't have the DELTA_UNIT
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flag set (fixes 'make check' again).
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2006-05-03 14:51:50 +00:00
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2006-05-03 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet be>
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad),
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(gst_matroska_mux_stream_is_vorbis_header),
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(gst_matroska_mux_write_data):
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Don't misinterpret GST_CLOCK_TIME_NONE as very high timestamp
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value and then dead-lock when muxing vorbis audio streams
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(the three vorbis header buffers carry no timestamp, and it
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would try to mux these after all video buffers). Fixes #340346.
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Improve clustering: start a new cluster also whenever we get
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a keyframe.
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2006-05-03 14:28:57 +00:00
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2006-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
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* win32/common/config.h:
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* win32/MANIFEST
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add the generated file as well
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2006-05-03 14:26:51 +00:00
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2006-05-03 Thomas Vander Stichele <thomas at apestaart dot org>
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* Makefile.am:
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* configure.ac:
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* win32/common/config.h.in:
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add win32 stuff
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2006-05-02 22:34:52 +00:00
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2006-05-03 Michael Smith <msmith@fluendo.com>
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* ext/cairo/gsttimeoverlay.c: (gst_cairo_time_overlay_transform):
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Fix timeoverlay for non-multiple-of-4 widths. This fourcc crap
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SUCKS.
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2006-05-02 18:50:23 +00:00
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2006-05-02 Stefan Kost <ensonic@users.sf.net>
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps):
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* gst/debug/negotiation.c: (gst_negotiation_update_caps):
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* gst/rtp/gstrtpilbcdepay.c: (gst_rtp_ilbc_depay_setcaps):
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don't leak caps-string
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2006-05-02 15:46:02 +00:00
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2006-05-02 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
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(gst_id3demux_sink_activate):
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Let core insert default error message for TYPE_NOT_FOUND
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errors, it's just as good as our own and has the added
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bonus of being translated.
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2006-05-02 15:40:15 +00:00
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2006-05-02 Tim-Philipp Müller <tim at centricular dot net>
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_init),
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(gst_tag_demux_sink_event):
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* gst/id3demux/gstid3demux.c: (gst_id3demux_init),
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(gst_id3demux_sink_event):
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Post an error message when we get an EOS event and were not
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able to find out the type of stream.
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* tests/check/elements/id3v2mux.c: (fill_mp3_buffer), (got_buffer),
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(test_taglib_id3mux_with_tags):
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Decrease num-buffers to 16 per iteration again, otherwise the
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many memcpy()s and reallocations in the test will hammer slow
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CPUs completely and make the test timeout.
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2006-05-02 13:24:38 +00:00
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2006-05-02 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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figure out where plugins-base plugins are
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* tests/check/Makefile.am:
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use plugins-base plugins, so we have typefind functions
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* tests/check/elements/id3v2mux.c: (test_taglib_id3mux_with_tags):
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increase num-buffers, this makes sure the test errors out instead
|
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of timing out when no typefind functions are present
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2006-05-01 11:46:33 +00:00
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2006-05-01 Thomas Vander Stichele <thomas at apestaart dot org>
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* docs/plugins/Makefile.am:
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also check .cc files for gtk-doc markup
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* configure.ac:
|
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|
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
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|
* tests/check/Makefile.am:
|
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* tests/check/elements/id3v2mux.c: (id3v2mux_suite), (main):
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* ext/Makefile.am:
|
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|
* ext/taglib/Makefile.am:
|
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|
|
* ext/taglib/gstid3v2mux.h:
|
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* ext/taglib/gsttaglibmux.c:
|
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* ext/taglib/gsttaglibmux.h:
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move taglib-based id3v2muxer to -good. Fixes #336110.
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2006-04-29 16:14:20 +00:00
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2006-04-29 Tim-Philipp Müller <tim at centricular dot net>
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
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... and fix multichannel/WAVFORMATEX support again.
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gst/wavparse/gstwavparse.*: Add push (streaming) mode to wavparse (fixes #337625)
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl),
(gst_wavparse_parse_cues), (gst_wavparse_parse_file_header),
(gst_wavparse_stream_init), (gst_wavparse_perform_seek),
(gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk),
(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data), (gst_wavparse_loop),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state), (plugin_init):
* gst/wavparse/gstwavparse.h:
Add push (streaming) mode to wavparse (fixes #337625)
2006-04-28 23:09:17 +00:00
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2006-04-29 Stefan Kost <ensonic@users.sf.net>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
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(gst_wavparse_class_init), (gst_wavparse_dispose),
|
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|
|
(gst_wavparse_reset), (gst_wavparse_init),
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(gst_wavparse_create_sourcepad), (gst_wavparse_parse_adtl),
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(gst_wavparse_parse_cues), (gst_wavparse_parse_file_header),
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(gst_wavparse_stream_init), (gst_wavparse_perform_seek),
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(gst_wavparse_peek_chunk_info), (gst_wavparse_peek_chunk),
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(gst_wavparse_stream_headers), (gst_wavparse_parse_stream_init),
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(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
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(gst_wavparse_stream_data), (gst_wavparse_loop),
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(gst_wavparse_chain), (gst_wavparse_srcpad_event),
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(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
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(gst_wavparse_change_state), (plugin_init):
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* gst/wavparse/gstwavparse.h:
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Add push (streaming) mode to wavparse (fixes #337625)
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2006-04-28 19:15:08 +00:00
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2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* tests/Makefile.am:
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add ximagesrc icles test
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2006-04-28 18:57:09 +00:00
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2006-04-28 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init),
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(gst_cmml_enc_push_clip):
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* sys/Makefile.am:
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* sys/ximage/Makefile.am:
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* sys/ximage/gstximagesrc.c:
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Move ximagesrc plug-in to good after review. Fixes #336756.
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gst/icydemux/gsticydemux.*: Fix event handling: cache events when typefinding and forward later.
Original commit message from CVS:
* gst/icydemux/gsticydemux.c: (gst_icydemux_reset), (gst_icydemux_init), (gst_icydemux_sink_setcaps),
(gst_icydemux_add_srcpad), (gst_icydemux_parse_and_send_tags),
(gst_icydemux_handle_event), (gst_icydemux_send_cached_events),
(gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
(gst_icydemux_chain), (gst_icydemux_send_tag_event):
* gst/icydemux/gsticydemux.h:
Fix event handling: cache events when typefinding and forward later.
2006-04-28 14:57:57 +00:00
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2006-04-28 Michael Smith <msmith@fluendo.com>
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* gst/icydemux/gsticydemux.c: (gst_icydemux_reset),
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(gst_icydemux_init), (gst_icydemux_sink_setcaps),
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(gst_icydemux_add_srcpad), (gst_icydemux_parse_and_send_tags),
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(gst_icydemux_handle_event), (gst_icydemux_send_cached_events),
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(gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
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(gst_icydemux_chain), (gst_icydemux_send_tag_event):
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* gst/icydemux/gsticydemux.h:
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Fix event handling: cache events when typefinding and forward later.
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2006-04-28 14:55:20 +00:00
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2006-04-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* sys/osxaudio/gstosxaudiosink.c:
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(plugin_init):
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Register osxaudiosrc to the plugin.
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* sys/osxaudio/gstosxaudiosrc.c:
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(gst_osx_audio_src_osxelement_do_init),
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(gst_osx_audio_src_base_init), (gst_osx_audio_src_class_init),
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(gst_osx_audio_src_init), (gst_osx_audio_src_set_property),
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(gst_osx_audio_src_get_property),
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(gst_osx_audio_src_create_ringbuffer), (gst_osx_audio_src_io_proc),
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(gst_osx_audio_src_osxelement_init):
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* sys/osxaudio/gstosxaudiosrc.h:
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Port of osxaudiosrc to 0.10.
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* sys/osxaudio/Makefile.am:
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Add osxaudiosrc
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2006-04-28 12:00:39 +00:00
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2006-04-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* sys/osxaudio/gstosxringbuffer.c:
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* sys/osxaudio/gstosxringbuffer.h:
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Forgot to commit earlier, part of the OSX audio plugin port
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2006-04-28 11:37:22 +00:00
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2006-04-28 Tim-Philipp Müller <tim at centricular dot net>
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* gst/id3demux/id3v2frames.c: (has_utf16_bom),
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(parse_split_strings):
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Recognise and skip any byte order marker (BOM) in
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UTF-16 strings.
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2006-04-27 16:05:54 +00:00
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2006-04-27 Tim-Philipp Müller <tim at centricular dot net>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* docs/plugins/inspect/plugin-avi.xml:
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* gst/avi/gstavidemux.c:
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* gst/avi/gstavimux.c:
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Add docs for both avidemux and avimux.
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gst/avi/: Port AVI muxer to GStreamer-0.10 (#332031).
Original commit message from CVS:
Patch by: Mark Nauwelaerts <manauw at skynet dot be>
* gst/avi/Makefile.am:
* gst/avi/gstavi.c: (plugin_init):
* gst/avi/gstavimux.c: (gst_avi_mux_get_type),
(gst_avi_mux_base_init), (gst_avi_mux_finalize),
(gst_avi_mux_class_init), (gst_avi_mux_init),
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
(gst_avi_mux_pad_link), (gst_avi_mux_pad_unlink),
(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
(gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
(gst_avi_mux_riff_get_avix_header),
(gst_avi_mux_riff_get_video_header),
(gst_avi_mux_riff_get_audio_header), (gst_avi_mux_add_index),
(gst_avi_mux_write_index), (gst_avi_mux_bigfile),
(gst_avi_mux_start_file), (gst_avi_mux_stop_file),
(gst_avi_mux_restart_file), (gst_avi_mux_handle_event),
(gst_avi_mux_fill_queue), (gst_avi_mux_send_pad_data),
(gst_avi_mux_strip_buffer), (gst_avi_mux_do_audio_buffer),
(gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
(gst_avi_mux_loop), (gst_avi_mux_collect_pads),
(gst_avi_mux_get_property), (gst_avi_mux_set_property),
(gst_avi_mux_change_state):
* gst/avi/gstavimux.h:
Port AVI muxer to GStreamer-0.10 (#332031).
* tests/check/Makefile.am:
* tests/check/elements/avimux.c:
* tests/check/elements/.cvsignore:
Add unit test for AVI muxer.
2006-04-27 14:51:06 +00:00
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2006-04-27 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet dot be>
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* gst/avi/Makefile.am:
|
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* gst/avi/gstavi.c: (plugin_init):
|
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* gst/avi/gstavimux.c: (gst_avi_mux_get_type),
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(gst_avi_mux_base_init), (gst_avi_mux_finalize),
|
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(gst_avi_mux_class_init), (gst_avi_mux_init),
|
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|
|
(gst_avi_mux_vidsink_set_caps), (gst_avi_mux_audsink_set_caps),
|
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(gst_avi_mux_pad_link), (gst_avi_mux_pad_unlink),
|
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(gst_avi_mux_request_new_pad), (gst_avi_mux_release_pad),
|
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(gst_avi_mux_write_tag), (gst_avi_mux_riff_get_avi_header),
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(gst_avi_mux_riff_get_avix_header),
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(gst_avi_mux_riff_get_video_header),
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(gst_avi_mux_riff_get_audio_header), (gst_avi_mux_add_index),
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(gst_avi_mux_write_index), (gst_avi_mux_bigfile),
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(gst_avi_mux_start_file), (gst_avi_mux_stop_file),
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(gst_avi_mux_restart_file), (gst_avi_mux_handle_event),
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(gst_avi_mux_fill_queue), (gst_avi_mux_send_pad_data),
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(gst_avi_mux_strip_buffer), (gst_avi_mux_do_audio_buffer),
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(gst_avi_mux_do_video_buffer), (gst_avi_mux_do_one_buffer),
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(gst_avi_mux_loop), (gst_avi_mux_collect_pads),
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(gst_avi_mux_get_property), (gst_avi_mux_set_property),
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(gst_avi_mux_change_state):
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* gst/avi/gstavimux.h:
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Port AVI muxer to GStreamer-0.10 (#332031).
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* tests/check/Makefile.am:
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* tests/check/elements/avimux.c:
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* tests/check/elements/.cvsignore:
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Add unit test for AVI muxer.
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gst/wavparse/gstwavparse.*: reverted patch #337625 for the price of 1 hour sleep
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_reset),
(gst_wavparse_init), (gst_wavparse_create_sourcepad),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
(gst_wavparse_stream_data), (gst_wavparse_loop),
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate),
(gst_wavparse_sink_activate_pull), (gst_wavparse_change_state),
(plugin_init):
* gst/wavparse/gstwavparse.h:
reverted patch #337625 for the price of 1 hour sleep
2006-04-26 21:29:45 +00:00
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|
2006-04-27 Stefan Kost <ensonic@users.sf.net>
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|
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
|
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|
(gst_wavparse_class_init), (gst_wavparse_reset),
|
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|
|
(gst_wavparse_init), (gst_wavparse_create_sourcepad),
|
|
|
|
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
|
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|
|
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
|
|
|
|
(gst_wavparse_send_event), (gst_wavparse_add_src_pad),
|
|
|
|
(gst_wavparse_stream_data), (gst_wavparse_loop),
|
|
|
|
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate),
|
|
|
|
(gst_wavparse_sink_activate_pull), (gst_wavparse_change_state),
|
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(plugin_init):
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|
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* gst/wavparse/gstwavparse.h:
|
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|
reverted patch #337625 for the price of 1 hour sleep
|
|
|
|
|
gst/wavparse/gstwavparse.*: correct partial implementation of push mode (from my last commit)
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_reset),
(gst_wavparse_init), (gst_wavparse_create_sourcepad),
(gst_wavparse_parse_adtl), (gst_wavparse_parse_cues),
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
(gst_wavparse_stream_data), (gst_wavparse_loop),
(gst_wavparse_chain), (plugin_init):
* gst/wavparse/gstwavparse.h:
correct partial implementation of push mode
(from my last commit)
2006-04-26 20:11:18 +00:00
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|
2006-04-26 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
|
|
|
|
(gst_wavparse_class_init), (gst_wavparse_reset),
|
|
|
|
(gst_wavparse_init), (gst_wavparse_create_sourcepad),
|
|
|
|
(gst_wavparse_parse_adtl), (gst_wavparse_parse_cues),
|
|
|
|
(gst_wavparse_parse_file_header), (gst_wavparse_stream_init),
|
|
|
|
(gst_wavparse_perform_seek), (gst_wavparse_stream_headers),
|
|
|
|
(gst_wavparse_stream_data), (gst_wavparse_loop),
|
|
|
|
(gst_wavparse_chain), (plugin_init):
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
correct partial implementation of push mode
|
|
|
|
(from my last commit)
|
|
|
|
|
2006-04-26 17:37:10 +00:00
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|
|
2006-04-26 Wim Taymans <wim@fluendo.com>
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|
|
|
|
|
|
* ext/esd/esdsink.c:
|
|
|
|
Fix compile problem by defining ESD_MAX_WRITE_SIZE if
|
|
|
|
it is not in esd.h
|
|
|
|
|
gst/auparse/gstauparse.*: Rewrite auparse to suck a little bit less: make source pad dynamic, so decodebin/playbin wo...
Original commit message from CVS:
* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
(gst_au_parse_class_init), (gst_au_parse_init),
(gst_au_parse_reset), (gst_au_parse_add_srcpad),
(gst_au_parse_remove_srcpad), (gst_au_parse_parse_header),
(gst_au_parse_chain), (gst_au_parse_src_convert),
(gst_au_parse_src_query), (gst_au_parse_handle_seek),
(gst_au_parse_sink_event), (gst_au_parse_src_event),
(gst_au_parse_change_state):
* gst/auparse/gstauparse.h:
Rewrite auparse to suck a little bit less: make source pad
dynamic, so decodebin/playbin work with non-raw formats
like alaw/mulaw; add query function for duration/position
queries; check whether we have enough data before attempting
to parse the header (instead of crashing when that is not the
case); work around audioconvert sucking by swapping endianness
to the native endianness ourselves for float formats; send
initial newsegment event. Fixes #161712.
2006-04-26 17:08:24 +00:00
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|
2006-04-26 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
|
|
|
|
(gst_au_parse_class_init), (gst_au_parse_init),
|
|
|
|
(gst_au_parse_reset), (gst_au_parse_add_srcpad),
|
|
|
|
(gst_au_parse_remove_srcpad), (gst_au_parse_parse_header),
|
|
|
|
(gst_au_parse_chain), (gst_au_parse_src_convert),
|
|
|
|
(gst_au_parse_src_query), (gst_au_parse_handle_seek),
|
|
|
|
(gst_au_parse_sink_event), (gst_au_parse_src_event),
|
|
|
|
(gst_au_parse_change_state):
|
|
|
|
* gst/auparse/gstauparse.h:
|
|
|
|
Rewrite auparse to suck a little bit less: make source pad
|
|
|
|
dynamic, so decodebin/playbin work with non-raw formats
|
|
|
|
like alaw/mulaw; add query function for duration/position
|
|
|
|
queries; check whether we have enough data before attempting
|
|
|
|
to parse the header (instead of crashing when that is not the
|
|
|
|
case); work around audioconvert sucking by swapping endianness
|
|
|
|
to the native endianness ourselves for float formats; send
|
|
|
|
initial newsegment event. Fixes #161712.
|
|
|
|
|
sys/osxaudio/: Port of osxaudiosink to 0.10
Original commit message from CVS:
2006-04-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/osxaudio/Makefile.am:
* sys/osxaudio/gstosxaudioelement.c:
(gst_osx_audio_element_get_type),
(gst_osx_audio_element_class_init):
* sys/osxaudio/gstosxaudioelement.h:
* sys/osxaudio/gstosxaudiosink.c:
(gst_osx_audio_sink_osxelement_do_init),
(gst_osx_audio_sink_base_init), (gst_osx_audio_sink_class_init),
(gst_osx_audio_sink_init), (gst_osx_audio_sink_set_property),
(gst_osx_audio_sink_get_property), (gst_osx_audio_sink_getcaps),
(gst_osx_audio_sink_create_ringbuffer),
(gst_osx_audio_sink_io_proc), (gst_osx_audio_sink_osxelement_init),
(plugin_init):
* sys/osxaudio/gstosxaudiosink.h:
Port of osxaudiosink to 0.10
2006-04-26 16:29:38 +00:00
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|
|
2006-04-26 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* sys/osxaudio/Makefile.am:
|
|
|
|
* sys/osxaudio/gstosxaudioelement.c:
|
|
|
|
(gst_osx_audio_element_get_type),
|
|
|
|
(gst_osx_audio_element_class_init):
|
|
|
|
* sys/osxaudio/gstosxaudioelement.h:
|
|
|
|
* sys/osxaudio/gstosxaudiosink.c:
|
|
|
|
(gst_osx_audio_sink_osxelement_do_init),
|
|
|
|
(gst_osx_audio_sink_base_init), (gst_osx_audio_sink_class_init),
|
|
|
|
(gst_osx_audio_sink_init), (gst_osx_audio_sink_set_property),
|
|
|
|
(gst_osx_audio_sink_get_property), (gst_osx_audio_sink_getcaps),
|
|
|
|
(gst_osx_audio_sink_create_ringbuffer),
|
|
|
|
(gst_osx_audio_sink_io_proc), (gst_osx_audio_sink_osxelement_init),
|
|
|
|
(plugin_init):
|
|
|
|
* sys/osxaudio/gstosxaudiosink.h:
|
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|
|
Port of osxaudiosink to 0.10
|
|
|
|
|
2006-04-26 08:55:27 +00:00
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|
2006-04-26 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_prepare), (gst_esdsink_delay):
|
|
|
|
Always write ESD_BUF_SIZE bytes and use ESD_MAX_WRITE_SIZE as
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the size of the ringbuffer. This should fix hangs with older
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esd sound servers.
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Define GstElementDetails as const and also static (when defined as global)
Original commit message from CVS:
* ext/aalib/gstaasink.c:
* ext/annodex/gstcmmldec.c:
* ext/annodex/gstcmmlenc.c:
* ext/cairo/gsttextoverlay.c:
* ext/cairo/gsttimeoverlay.c:
* ext/cdio/gstcdiocddasrc.c:
* ext/dv/gstdvdec.c:
* ext/dv/gstdvdemux.c:
* ext/esd/esdmon.c:
* ext/esd/esdsink.c:
* ext/flac/gstflacenc.c:
* ext/flac/gstflactag.c:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
* ext/gdk_pixbuf/pixbufscale.c:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
* ext/jpeg/gstjpegdec.c:
* ext/jpeg/gstjpegenc.c:
* ext/jpeg/gstsmokedec.c:
* ext/jpeg/gstsmokeenc.c:
* ext/libcaca/gstcacasink.c:
* ext/libmng/gstmngdec.c:
* ext/libmng/gstmngenc.c:
* ext/libpng/gstpngdec.c:
* ext/libpng/gstpngenc.c:
* ext/mikmod/gstmikmod.c:
* ext/raw1394/gstdv1394src.c:
* ext/shout2/gstshout2.c: (gst_shout2send_init):
* ext/shout2/gstshout2.h:
* ext/speex/gstspeexdec.c:
* ext/speex/gstspeexenc.c:
* gst/alpha/gstalpha.c:
* gst/alpha/gstalphacolor.c:
* gst/apetag/gstapedemux.c:
* gst/auparse/gstauparse.c:
* gst/autodetect/gstautoaudiosink.c:
(gst_auto_audio_sink_base_init):
* gst/autodetect/gstautovideosink.c:
(gst_auto_video_sink_base_init):
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
* gst/avi/gstavimux.c: (gst_avimux_base_init):
* gst/cutter/gstcutter.c:
* gst/debug/breakmydata.c:
* gst/debug/efence.c:
* gst/debug/gstnavigationtest.c:
* gst/debug/gstnavseek.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/testplugin.c:
* gst/effectv/gstaging.c:
* gst/effectv/gstdice.c:
* gst/effectv/gstedge.c:
* gst/effectv/gstquark.c:
* gst/effectv/gstrev.c:
* gst/effectv/gstshagadelic.c:
* gst/effectv/gstvertigo.c:
* gst/effectv/gstwarp.c:
* gst/flx/gstflxdec.c:
* gst/goom/gstgoom.c:
* gst/icydemux/gsticydemux.c:
* gst/id3demux/gstid3demux.c:
* gst/interleave/deinterleave.c:
* gst/interleave/interleave.c:
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
* gst/level/gstlevel.c:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
* gst/median/gstmedian.c:
* gst/monoscope/gstmonoscope.c:
* gst/multipart/multipartdemux.c:
* gst/multipart/multipartmux.c:
* gst/oldcore/gstaggregator.c:
* gst/oldcore/gstfdsink.c:
* gst/oldcore/gstmd5sink.c:
* gst/oldcore/gstmultifilesrc.c:
* gst/oldcore/gstpipefilter.c:
* gst/oldcore/gstshaper.c:
* gst/oldcore/gststatistics.c:
* gst/rtp/gstasteriskh263.c:
* gst/rtp/gstrtpL16depay.c:
* gst/rtp/gstrtpL16pay.c:
* gst/rtp/gstrtpamrdepay.c:
* gst/rtp/gstrtpamrpay.c:
* gst/rtp/gstrtpdepay.c:
* gst/rtp/gstrtpgsmpay.c:
* gst/rtp/gstrtph263pay.c:
* gst/rtp/gstrtph263pdepay.c:
* gst/rtp/gstrtph263ppay.c:
* gst/rtp/gstrtpilbcdepay.c:
* gst/rtp/gstrtpmp4gpay.c:
* gst/rtp/gstrtpmp4vdepay.c:
* gst/rtp/gstrtpmp4vpay.c:
* gst/rtp/gstrtpmpadepay.c:
* gst/rtp/gstrtpmpapay.c:
* gst/rtp/gstrtppcmadepay.c:
* gst/rtp/gstrtppcmapay.c:
* gst/rtp/gstrtppcmudepay.c:
* gst/rtp/gstrtppcmupay.c:
* gst/rtp/gstrtpspeexdepay.c:
* gst/rtp/gstrtpspeexpay.c:
* gst/rtsp/gstrtpdec.c:
* gst/rtsp/gstrtspsrc.c:
* gst/smpte/gstsmpte.c:
* gst/udp/gstdynudpsink.c:
* gst/udp/gstmultiudpsink.c:
* gst/udp/gstudpsink.c:
* gst/udp/gstudpsrc.c:
* gst/videobox/gstvideobox.c:
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideotemplate.c:
(gst_videotemplate_base_init):
* gst/videomixer/videomixer.c:
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
(gst_wavparse_class_init), (gst_wavparse_dispose),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
(gst_wavparse_chain), (gst_wavparse_srcpad_event),
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
* sys/oss/gstossmixerelement.c:
* sys/oss/gstosssink.c:
* sys/oss/gstosssrc.c:
* sys/osxaudio/gstosxaudioelement.c:
* sys/osxaudio/gstosxaudiosink.c:
* sys/osxaudio/gstosxaudiosrc.c:
* sys/sunaudio/gstsunaudiomixer.c:
* sys/sunaudio/gstsunaudiosink.c:
Define GstElementDetails as const and also static (when defined as
global)
2006-04-25 21:39:46 +00:00
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2006-04-26 Stefan Kost <ensonic@users.sf.net>
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* ext/aalib/gstaasink.c:
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* ext/annodex/gstcmmldec.c:
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* ext/annodex/gstcmmlenc.c:
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* ext/cairo/gsttextoverlay.c:
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* ext/cairo/gsttimeoverlay.c:
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* ext/cdio/gstcdiocddasrc.c:
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* ext/dv/gstdvdec.c:
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* ext/dv/gstdvdemux.c:
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* ext/esd/esdmon.c:
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* ext/esd/esdsink.c:
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* ext/flac/gstflacenc.c:
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* ext/flac/gstflactag.c:
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
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* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
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* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
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* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
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* ext/gdk_pixbuf/pixbufscale.c:
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* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
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* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
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* ext/jpeg/gstjpegdec.c:
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* ext/jpeg/gstjpegenc.c:
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* ext/jpeg/gstsmokedec.c:
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* ext/jpeg/gstsmokeenc.c:
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* ext/libcaca/gstcacasink.c:
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* ext/libmng/gstmngdec.c:
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* ext/libmng/gstmngenc.c:
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* ext/libpng/gstpngdec.c:
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* ext/libpng/gstpngenc.c:
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* ext/mikmod/gstmikmod.c:
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* ext/raw1394/gstdv1394src.c:
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* ext/shout2/gstshout2.c: (gst_shout2send_init):
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* ext/shout2/gstshout2.h:
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* ext/speex/gstspeexdec.c:
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* ext/speex/gstspeexenc.c:
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* gst/alpha/gstalpha.c:
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* gst/alpha/gstalphacolor.c:
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* gst/apetag/gstapedemux.c:
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* gst/auparse/gstauparse.c:
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* gst/autodetect/gstautoaudiosink.c:
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(gst_auto_audio_sink_base_init):
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* gst/autodetect/gstautovideosink.c:
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(gst_auto_video_sink_base_init):
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* gst/avi/gstavidemux.c: (gst_avi_demux_base_init):
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* gst/avi/gstavimux.c: (gst_avimux_base_init):
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* gst/cutter/gstcutter.c:
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* gst/debug/breakmydata.c:
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* gst/debug/efence.c:
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* gst/debug/gstnavigationtest.c:
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* gst/debug/gstnavseek.c:
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* gst/debug/negotiation.c:
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* gst/debug/progressreport.c:
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* gst/debug/testplugin.c:
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* gst/effectv/gstaging.c:
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* gst/effectv/gstdice.c:
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* gst/effectv/gstedge.c:
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* gst/effectv/gstquark.c:
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* gst/effectv/gstrev.c:
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* gst/effectv/gstshagadelic.c:
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* gst/effectv/gstvertigo.c:
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* gst/effectv/gstwarp.c:
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* gst/flx/gstflxdec.c:
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* gst/goom/gstgoom.c:
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* gst/icydemux/gsticydemux.c:
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* gst/id3demux/gstid3demux.c:
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* gst/interleave/deinterleave.c:
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* gst/interleave/interleave.c:
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* gst/law/alaw-decode.c: (gst_alawdec_base_init):
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* gst/law/alaw-encode.c: (gst_alawenc_base_init):
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* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
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* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
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* gst/level/gstlevel.c:
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
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* gst/median/gstmedian.c:
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* gst/monoscope/gstmonoscope.c:
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* gst/multipart/multipartdemux.c:
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* gst/multipart/multipartmux.c:
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* gst/oldcore/gstaggregator.c:
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* gst/oldcore/gstfdsink.c:
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* gst/oldcore/gstmd5sink.c:
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* gst/oldcore/gstmultifilesrc.c:
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* gst/oldcore/gstpipefilter.c:
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* gst/oldcore/gstshaper.c:
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* gst/oldcore/gststatistics.c:
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* gst/rtp/gstasteriskh263.c:
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* gst/rtp/gstrtpL16depay.c:
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* gst/rtp/gstrtpL16pay.c:
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* gst/rtp/gstrtpamrdepay.c:
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* gst/rtp/gstrtpamrpay.c:
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* gst/rtp/gstrtpdepay.c:
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* gst/rtp/gstrtpgsmpay.c:
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* gst/rtp/gstrtph263pay.c:
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* gst/rtp/gstrtph263pdepay.c:
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* gst/rtp/gstrtph263ppay.c:
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* gst/rtp/gstrtpilbcdepay.c:
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* gst/rtp/gstrtpmp4gpay.c:
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* gst/rtp/gstrtpmp4vdepay.c:
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* gst/rtp/gstrtpmp4vpay.c:
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* gst/rtp/gstrtpmpadepay.c:
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* gst/rtp/gstrtpmpapay.c:
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* gst/rtp/gstrtppcmadepay.c:
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* gst/rtp/gstrtppcmapay.c:
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* gst/rtp/gstrtppcmudepay.c:
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* gst/rtp/gstrtppcmupay.c:
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* gst/rtp/gstrtpspeexdepay.c:
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* gst/rtp/gstrtpspeexpay.c:
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* gst/rtsp/gstrtpdec.c:
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* gst/rtsp/gstrtspsrc.c:
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* gst/smpte/gstsmpte.c:
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* gst/udp/gstdynudpsink.c:
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* gst/udp/gstmultiudpsink.c:
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* gst/udp/gstudpsink.c:
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* gst/udp/gstudpsrc.c:
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* gst/videobox/gstvideobox.c:
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* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
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* gst/videofilter/gstvideobalance.c:
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* gst/videofilter/gstvideoflip.c:
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* gst/videofilter/gstvideotemplate.c:
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(gst_videotemplate_base_init):
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* gst/videomixer/videomixer.c:
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* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
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(gst_wavparse_class_init), (gst_wavparse_dispose),
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(gst_wavparse_reset), (gst_wavparse_init),
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(gst_wavparse_perform_seek), (gst_wavparse_peek_chunk_info),
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(gst_wavparse_peek_chunk), (gst_wavparse_stream_headers),
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(gst_wavparse_parse_stream_init), (gst_wavparse_send_event),
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(gst_wavparse_add_src_pad), (gst_wavparse_stream_data),
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(gst_wavparse_chain), (gst_wavparse_srcpad_event),
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(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull),
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(gst_wavparse_change_state):
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* gst/wavparse/gstwavparse.h:
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* sys/oss/gstossmixerelement.c:
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* sys/oss/gstosssink.c:
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* sys/oss/gstosssrc.c:
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* sys/osxaudio/gstosxaudioelement.c:
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* sys/osxaudio/gstosxaudiosink.c:
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* sys/osxaudio/gstosxaudiosrc.c:
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* sys/sunaudio/gstsunaudiomixer.c:
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* sys/sunaudio/gstsunaudiosink.c:
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Define GstElementDetails as const and also static (when defined as
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global)
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2006-04-25 17:57:23 +00:00
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2006-04-25 Tim-Philipp Müller <tim at centricular dot net>
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
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Source pad has fixed caps. If we don't set this, bad
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things happen when the window is resized.
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2006-04-25 16:38:50 +00:00
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2006-04-25 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/Makefile.am:
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
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(gst_matroska_demux_handle_src_event):
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* gst/matroska/matroska-ids.c:
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(gst_matroska_track_init_video_context),
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(gst_matroska_track_init_audio_context),
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(gst_matroska_track_init_subtitle_context),
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(gst_matroska_track_init_complex_context):
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* gst/matroska/matroska-ids.h:
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Handle case where the TrackType ebml chunk does not come before the
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TrackInfoAudio or TrackInfoVideo ebml chunk (#339446). Ignore QoS
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events.
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2006-04-25 16:09:55 +00:00
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2006-04-25 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_setcaps):
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* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
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It's codec_data, not codec_info.
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2006-04-25 11:45:00 +00:00
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2006-04-25 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet dot be>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
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Handle codec_data for VfW compatibility codec IDs (#339451)
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_video_pad_setcaps):
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Same here, handle codec_data and add additional caps we can handle
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now to the pad template (huffyuv, dv and h263 video) (#339451)
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2006-04-25 11:09:24 +00:00
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2006-04-25 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Josef Zlomek <josef dot zlomek at itonis dot tv>
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_create_buffer_header),
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(gst_matroska_mux_write_data):
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Fix timestamping of B-frames, use signed integers, do
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some rounding (#339678).
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2006-04-24 18:30:55 +00:00
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2006-04-24 Edgard Lima <edgard.lima@indt.org.br>
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* ext/annodex/gstcmmlparser.c: (gst_cmml_parser_generic_error):
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just make it compile with --disable-gst-debug.
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2006-04-23 15:55:30 +00:00
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2006-04-23 Sebastien Moutte <sebastien@moutte.net>
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* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
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Fix a bad conversion using gst_guint64_to_gdouble.
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fabs ((gdouble) demux->index[entry].time - (gdouble) seek_pos) can not be
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replaced by fabs (gst_guint64_to_gdouble (demux->index[entry].time - seek_pos)) as the
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difference could be negative. fabs (gst_guint64_to_gdouble (demux->index[entry].time) -
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gst_guint64_to_gdouble (seek_pos)) is the good solution. Thanks to Tim who has seen my
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mistake.
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2006-04-22 15:32:48 +00:00
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2006-04-21 Sebastien Moutte <sebastien@moutte.net>
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* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
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Use gst_guint64_to_gdouble for conversions
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* win32/vs6/gst_plugins_good.dsw:
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* win32/vs6/libgsticydemux.dsp:
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Add a project file for icydemux
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2006-04-21 18:07:10 +00:00
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2006-04-21 Wim Taymans <wim@fluendo.com>
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Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
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* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
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(gst_avi_demux_parse_index), (gst_avi_demux_massage_index):
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When splitting audio chunks, the block alignment is not taken in
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consideration, so the smaller chunks could be of size which is
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not a multiple of the block alignment. Fixes #336904
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2006-04-21 17:59:03 +00:00
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2006-04-21 Wim Taymans <wim@fluendo.com>
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* ext/raw1394/gstdv1394src.c: (gst_dv1394src_convert):
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Use scale functions
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2006-04-21 17:27:40 +00:00
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2006-04-21 Tim-Philipp Müller <tim at centricular dot net>
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* ext/dv/gstdv.c: (plugin_init):
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Fix build.
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2006-04-21 17:15:29 +00:00
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2006-04-21 Tim-Philipp Müller <tim at centricular dot net>
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* gst/debug/progressreport.c: (gst_progress_report_finalize),
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(gst_progress_report_class_init), (gst_progress_report_init),
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(gst_progress_report_do_query), (gst_progress_report_report),
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(gst_progress_report_set_property),
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(gst_progress_report_get_property):
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Add 'format' property to force querying to a particular format.
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2006-04-21 15:50:28 +00:00
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2006-04-21 Andy Wingo <wingo@pobox.com>
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* ext/dv/gstdv.c (plugin_init): libdv is a marginal decoder, at
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best, on big endian systems. Drop its rank in that case. OTOH on
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x86 it's quite fine. See changes from today in gst-ffmpeg as well.
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Add icydemux, and tests.
Original commit message from CVS:
* configure.ac:
* gst/icydemux/Makefile.am:
* gst/icydemux/gsticydemux.c: (gst_icydemux_get_type),
(gst_icydemux_base_init), (gst_icydemux_class_init),
(gst_icydemux_reset), (gst_icydemux_init),
(gst_icydemux_sink_setcaps), (gst_icydemux_dispose),
(gst_icydemux_add_srcpad), (gst_icydemux_remove_srcpad),
(unicodify), (gst_icydemux_unicodify),
(gst_icydemux_parse_and_send_tags),
(gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
(gst_icydemux_chain), (gst_icydemux_change_state),
(gst_icydemux_send_tag_event), (plugin_init):
* gst/icydemux/gsticydemux.h:
* tests/check/Makefile.am:
* tests/check/elements/icydemux.c: (typefind_succeed),
(plugin_init), (icydemux_found_pad), (create_icydemux),
(cleanup_icydemux), (push_data), (GST_START_TEST),
(icydemux_suite), (main):
Add icydemux, and tests.
2006-04-21 09:27:11 +00:00
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2006-04-21 Michael Smith <msmith@fluendo.com>
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* configure.ac:
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* gst/icydemux/Makefile.am:
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* gst/icydemux/gsticydemux.c: (gst_icydemux_get_type),
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(gst_icydemux_base_init), (gst_icydemux_class_init),
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(gst_icydemux_reset), (gst_icydemux_init),
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(gst_icydemux_sink_setcaps), (gst_icydemux_dispose),
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(gst_icydemux_add_srcpad), (gst_icydemux_remove_srcpad),
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(unicodify), (gst_icydemux_unicodify),
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(gst_icydemux_parse_and_send_tags),
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(gst_icydemux_typefind_or_forward), (gst_icydemux_add_meta),
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(gst_icydemux_chain), (gst_icydemux_change_state),
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(gst_icydemux_send_tag_event), (plugin_init):
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* gst/icydemux/gsticydemux.h:
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* tests/check/Makefile.am:
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* tests/check/elements/icydemux.c: (typefind_succeed),
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(plugin_init), (icydemux_found_pad), (create_icydemux),
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(cleanup_icydemux), (push_data), (GST_START_TEST),
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(icydemux_suite), (main):
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Add icydemux, and tests.
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2006-04-20 17:48:29 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (gst_flac_dec_loop):
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Post SEGMENT_DONE message in TIME format.
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2006-04-20 16:33:55 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
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* gst/avi/gstavidemux.c: (gst_avi_demux_peek_tag),
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(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan):
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Fix index creation when we have to scan the file to create
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an index. There may be other types of RIFF 'LIST' chunks than
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'movi' and we need to skip them properly as well or we'll end up
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reading garbage (#336889). Some other cosmetic changes.
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2006-04-20 14:21:42 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (gst_flac_dec_loop),
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(gst_flac_dec_handle_seek_event):
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Add support for segment seeks (fixes #338290). Also demote
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some recurring debug message from DEBUG to LOG level.
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2006-04-20 13:23:40 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
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(gst_matroskademux_do_index_seek),
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(gst_matroska_demux_handle_seek_event),
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(gst_matroska_demux_parse_blockgroup_or_simpleblock):
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* gst/matroska/matroska-ids.h:
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Set DISCONT flag on first buffer after a discontinuity.
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Fix newsegment events sent when seeking and honour KEY_UNIT
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seek flag. Create pad with bogus caps if we don't recognise
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the stream codec id.
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* gst/matroska/matroska-demux.h:
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Fix GObject macros.
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2006-04-20 11:00:16 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Mark Nauwelaerts <manauw at skynet dot be>
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* gst/matroska/matroska-demux.c:
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(gst_matroska_demux_handle_seek_event), (gst_matroska_demux_loop):
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Handle end of segment properly when set; don't dead-lock when
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posting start of segment message when doing a segment seek.
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Fixes #338810.
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2006-04-20 09:11:22 +00:00
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2006-04-20 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_audio_caps),
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(gst_matroska_demux_plugin_init):
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Make mpeg2 aac audio work: create artificial private codec data
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chunk which faad2 seems to require, just as we do for mpeg4 aac.
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Also call gst_riff_init(). Partially fixes #338767.
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gst/wavenc/gstwavenc.*: Set caps on first outgoing buffer, so that it doesn't error out immediately with a non-negoti...
Original commit message from CVS:
* gst/wavenc/gstwavenc.c: (gst_wavenc_base_init),
(gst_wavenc_class_init), (gst_wavenc_init),
(gst_wavenc_create_header_buf), (gst_wavenc_push_header),
(gst_wavenc_sink_setcaps), (get_id_from_name), (gst_wavenc_event),
(gst_wavenc_chain), (gst_wavenc_change_state):
* gst/wavenc/gstwavenc.h:
Set caps on first outgoing buffer, so that it doesn't error out
immediately with a non-negotiated error (#338716). Rewrite and
clean up a bit; fix setcaps function to parse things properly;
fix sink caps (8bit audio is unsigned and doesn't have depth);
use boilerplate macros; remove unused properties stuff.
2006-04-19 15:16:33 +00:00
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2006-04-19 Tim-Philipp Müller <tim at centricular dot net>
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* gst/wavenc/gstwavenc.c: (gst_wavenc_base_init),
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(gst_wavenc_class_init), (gst_wavenc_init),
|
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(gst_wavenc_create_header_buf), (gst_wavenc_push_header),
|
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(gst_wavenc_sink_setcaps), (get_id_from_name), (gst_wavenc_event),
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(gst_wavenc_chain), (gst_wavenc_change_state):
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* gst/wavenc/gstwavenc.h:
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Set caps on first outgoing buffer, so that it doesn't error out
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immediately with a non-negotiated error (#338716). Rewrite and
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clean up a bit; fix setcaps function to parse things properly;
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fix sink caps (8bit audio is unsigned and doesn't have depth);
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use boilerplate macros; remove unused properties stuff.
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2006-04-18 18:14:34 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gdk_pixbuf/gstgdkpixbuf.c:
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Leave JPEG decoding to our jpegdec plugin. gdkpixbufdec cannot
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handle MJPEG streams and might be autoplugged for those if the
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user doesn't have jpegdec installed (resulting in a cryptic error
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message about huffman tables). Better to disable JPEG decoding here
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and let the user figure out that she needs to install jpegdec.
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2006-04-18 18:04:48 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
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(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init),
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(gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_chain):
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* ext/gdk_pixbuf/gstgdkpixbuf.h:
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Make work with packetised/framed input (e.g. png-in-quicktime). Use
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GST_ELEMENT_ERROR when we return GST_FLOW_ERROR. Add some
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GST_DEBUG_FUNCPTR here and there. Use GST_LOG for recurring
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debug messages. Fix boilerplate macros.
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2006-04-18 17:29:42 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_get_capslist),
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(gst_gdk_pixbuf_set_property), (gst_gdk_pixbuf_get_property):
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No need to special-case for Gdk-2.0 any longer, we require
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Gdk 2.2 or newer; minor clean-ups.
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Rewrite a bit: use GstBaseSink::start and stop instead of a state change function; use GST_ELEMENT_ERROR for error re...
Original commit message from CVS:
* ext/shout2/gstshout2.c: (gst_shout2send_base_init),
(gst_shout2send_class_init), (gst_shout2send_init),
(set_shout_metadata), (gst_shout2send_set_metadata),
(gst_shout2send_event), (gst_shout2send_start),
(gst_shout2send_connect), (gst_shout2send_stop),
(gst_shout2send_render), (gst_shout2send_set_property),
(gst_shout2send_get_property), (gst_shout2send_setcaps),
(plugin_init):
* ext/shout2/gstshout2.h:
* po/POTFILES.in:
Rewrite a bit: use GstBaseSink::start and stop instead of a state
change function; use GST_ELEMENT_ERROR for error reporting, not
g_error() or GST_ERROR(); don't unref caps in setcaps function,
will cause crashes or assertion failures; remove (unused) "sync"
property, basesink already has such a property; misc. other
minor fixes and cleanups.
2006-04-18 17:17:55 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* ext/shout2/gstshout2.c: (gst_shout2send_base_init),
|
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(gst_shout2send_class_init), (gst_shout2send_init),
|
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|
|
(set_shout_metadata), (gst_shout2send_set_metadata),
|
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|
(gst_shout2send_event), (gst_shout2send_start),
|
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(gst_shout2send_connect), (gst_shout2send_stop),
|
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(gst_shout2send_render), (gst_shout2send_set_property),
|
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(gst_shout2send_get_property), (gst_shout2send_setcaps),
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(plugin_init):
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* ext/shout2/gstshout2.h:
|
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* po/POTFILES.in:
|
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Rewrite a bit: use GstBaseSink::start and stop instead of a state
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|
change function; use GST_ELEMENT_ERROR for error reporting, not
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g_error() or GST_ERROR(); don't unref caps in setcaps function,
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will cause crashes or assertion failures; remove (unused) "sync"
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property, basesink already has such a property; misc. other
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minor fixes and cleanups.
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2006-04-18 14:15:33 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* ext/esd/esdsink.c: (gst_esdsink_open), (gst_esdsink_prepare):
|
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|
* ext/esd/gstesd.c: (plugin_init):
|
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* po/POTFILES.in:
|
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Add translatable error message for when we cannot
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connect to the sound server, as "Cannot open resource
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for writing" isn't really an acceptable message to show
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to the user in this case.
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2006-04-18 13:32:29 +00:00
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2006-04-18 Tim-Philipp Müller <tim at centricular dot net>
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* sys/oss/gst-i18n-plugin.h:
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Remove bogus file that doesn't belong here.
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2006-04-17 19:57:10 +00:00
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2006-04-17 Tim-Philipp Müller <tim at centricular dot net>
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Patch by: Philippe Valembois
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* ext/shout2/gstshout2.c: (gst_shout2send_init),
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(gst_shout2send_set_metadata), (gst_shout2send_event),
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(gst_shout2send_render), (gst_shout2send_change_state):
|
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* ext/shout2/gstshout2.h:
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Handle tags being received before the connection to
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the server is established properly (see #338636).
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2006-04-17 19:12:36 +00:00
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2006-04-17 Tim-Philipp Müller <tim at centricular dot net>
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* ext/shout2/gstshout2.c: (gst_shout2send_render):
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Don't crash in case the connection to the server fails:
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don't set pointer to NULL by assigning FALSE; error out
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properly by using GST_ELEMENT_ERROR and returning
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GST_FLOW_ERROR (fixes #338636). Lastly, free connection
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before resetting the pointer.
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2006-04-17 10:01:51 +00:00
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|
2006-04-17 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3tags.c:
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Recognise TCO (Genre) tags in ID3v2.2. Patch by Alex Lancaster
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(Fixes #338713)
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2006-04-12 09:42:10 +00:00
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2006-04-12 Wim Taymans <wim@fluendo.com>
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* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
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(gst_gdk_pixbuf_get_capslist), (gst_gdk_pixbuf_sink_getcaps),
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(gst_gdk_pixbuf_class_init), (gst_gdk_pixbuf_init),
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(gst_gdk_pixbuf_flush), (gst_gdk_pixbuf_sink_event),
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(gst_gdk_pixbuf_chain):
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Some cleanups.
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Added RGBA as a possible output format.
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Correctly free the supported mimetypes.
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deprecate silent arg, it's not used.
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Return result from _alloc_buffer to peer.
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2006-04-11 18:03:36 +00:00
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2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_process):
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Don't leak memory allocated by gst_buffer_new_and_alloc() by
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overwriting GST_BUFFER_MALLOCDATA.
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ext/libpng/gstpngdec.*: Handle more than one frame if the content is framed, like with png-in-quicktime (#331917).
Original commit message from CVS:
* ext/libpng/gstpngdec.c: (gst_pngdec_init),
(user_endrow_callback), (user_end_callback),
(gst_pngdec_caps_create_and_set), (gst_pngdec_chain),
(gst_pngdec_sink_setcaps), (gst_pngdec_sink_event),
(gst_pngdec_libpng_clear), (gst_pngdec_change_state):
* ext/libpng/gstpngdec.h:
Handle more than one frame if the content is framed,
like with png-in-quicktime (#331917).
2006-04-11 09:35:45 +00:00
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2006-04-11 Tim-Philipp Müller <tim at centricular dot net>
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* ext/libpng/gstpngdec.c: (gst_pngdec_init),
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(user_endrow_callback), (user_end_callback),
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(gst_pngdec_caps_create_and_set), (gst_pngdec_chain),
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(gst_pngdec_sink_setcaps), (gst_pngdec_sink_event),
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(gst_pngdec_libpng_clear), (gst_pngdec_change_state):
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* ext/libpng/gstpngdec.h:
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Handle more than one frame if the content is framed,
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like with png-in-quicktime (#331917).
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2006-04-10 19:55:31 +00:00
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2006-04-10 Thomas Vander Stichele <thomas at apestaart dot org>
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* sys/oss/Makefile.am:
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* sys/oss/common.h:
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* sys/oss/gstosssink.c: (gst_oss_sink_init), (gst_oss_sink_open),
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(gst_oss_sink_prepare), (gst_oss_sink_unprepare):
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* sys/oss/gstosssrc.c: (gst_oss_src_prepare),
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(gst_oss_src_unprepare):
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- the user-visible error strings were in the wrong category
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- and the messages were not marked for translation
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- which is actually a good thing, because they were exactly
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the kind of message you would never want anyone to see
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- the macros were using variables that didn't exist in the macro
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arguments
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- and they were obviously copied from each other and then modified
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- so a common header makes sense
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2006-04-10 16:09:03 +00:00
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2006-04-10 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/ebml-read.c: (gst_ebml_read_sint):
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Don't try to modify read-only data.
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* gst/matroska/matroska-demux.c:
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(gst_matroska_demux_parse_blockgroup_or_simpleblock):
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Fix comment (won't crash any longer now).
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2006-04-10 15:48:55 +00:00
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2006-04-10 Michael Smith <msmith@fluendo.com>
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* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_set_header_on_caps):
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Use copies of header buffers for caps to avoid circular refcounting
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problems (as in theoradec, vorbisdec).
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* tests/check/elements/cmmldec.c: (GST_START_TEST):
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Fix a typo in test that meant it was testing the wrong thing.
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* tests/check/elements/cmmlenc.c: (check_headers):
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Fix refcount checks now that we use buffer-copies for caps.
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2006-04-10 15:43:54 +00:00
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2006-04-10 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init),
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(gst_matroska_demux_handle_seek_event),
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(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps),
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(gst_matroska_demux_subtitle_caps),
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(gst_matroska_demux_plugin_init):
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Use static pad templates with ANY caps for audio and video
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source pads and get rid of a lot of unnecessary (and partially
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broken) code for the template caps. Clean up caps finding
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functions. Fixes playback of audio files/streams that do not
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contain the sample rate and/or number of channels in the audio
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context (happens a lot with vorbis/mp3 .mka files it seems).
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Fixes #337183.
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Also add myself to copyright holders.
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2006-04-10 15:29:21 +00:00
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2006-04-10 Michael Smith <msmith@fluendo.com>
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* ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_del_clip):
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Use g_list_delete_link () instead of g_list_remove_link () so that
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we free the link as well as the contained data.
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2006-04-10 10:10:55 +00:00
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2006-04-10 Wim Taymans <wim@fluendo.com>
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Patch by: Ryan Lortie (desrt) <desrt at destr dot ca>
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* gst/avi/gstavidemux.c: (gst_avi_demux_parse_superindex),
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(gst_avi_demux_parse_stream), (gst_avi_demux_parse_index),
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(gst_avi_demux_stream_header):
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Fix some crashers with empty chunks. (Fixes #337749)
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2006-04-09 17:31:37 +00:00
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2006-04-09 Sebastien Moutte <sebastien@moutte.net>
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* gst/level/gstlevel.c: (gst_level_set_caps),(gst_level_transform_ip):
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use G_GINT64_CONSTANT for INT64 constants
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* gst/videofilter/gstvideobalance.c:
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define rint for WIN32 #define rint(x) (floor((x)+0.5))
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* win32/vs6/libgstavi.dsp:
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|
|
add missing libraries for the link and remove avimux.c from
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the project as it isn't ported to 0.10 yet
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2006-04-09 14:00:32 +00:00
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2006-04-09 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/ebml-read.c: (gst_ebml_read_sint):
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Even better would be if we actually did the right thing
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here (also, G_GUINT64_CONSTANT only exists since GLib-2.10).
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2006-04-09 13:52:03 +00:00
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2006-04-08 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/ebml-read.c: (gst_ebml_read_sint):
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Can't just replace 1LL with 1L here just because MSVC doesn't
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|
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support it, as it might lead to incorrect results when doing the
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bitshifting here. Using GLib's G_GUINT64_CONSTANT() macro to
|
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force a 64-bit constant in a way that all compilers are happy with.
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2006-04-08 21:21:45 +00:00
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|
2006-04-08 Stefan Kost <ensonic@users.sf.net>
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* ext/aalib/gstaasink.c: (gst_aasink_class_init):
|
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|
* ext/esd/esdsink.c: (gst_esdsink_class_init):
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* ext/flac/gstflactag.c: (gst_flac_tag_class_init):
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* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_class_init):
|
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|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_class_init):
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* ext/jpeg/gstsmokedec.c: (gst_smokedec_class_init):
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_class_init):
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* ext/libcaca/gstcacasink.c: (gst_cacasink_class_init):
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* ext/libmng/gstmngdec.c: (gst_mngdec_class_init):
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* ext/libmng/gstmngenc.c: (gst_mngenc_class_init):
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|
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init):
|
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|
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init):
|
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|
* ext/mikmod/gstmikmod.c: (gst_mikmod_class_init):
|
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* ext/shout2/gstshout2.c: (gst_shout2send_class_init):
|
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|
|
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init):
|
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|
|
* gst/alpha/gstalpha.c: (gst_alpha_class_init):
|
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|
|
* gst/avi/gstavimux.c: (gst_avimux_class_init):
|
|
|
|
* gst/debug/efence.c: (gst_efence_class_init):
|
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|
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* gst/debug/negotiation.c: (gst_negotiation_class_init):
|
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|
|
* gst/flx/gstflxdec.c: (gst_flxdec_class_init):
|
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|
|
* gst/goom/gstgoom.c: (gst_goom_class_init):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_class_init):
|
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|
|
* gst/interleave/deinterleave.c: (deinterleave_class_init):
|
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|
|
* gst/interleave/interleave.c: (interleave_class_init):
|
|
|
|
* gst/law/alaw-decode.c: (gst_alawdec_class_init):
|
|
|
|
* gst/law/alaw-encode.c: (gst_alawenc_class_init):
|
|
|
|
* gst/law/mulaw-encode.c: (gst_mulawenc_class_init):
|
|
|
|
* gst/median/gstmedian.c: (gst_median_class_init):
|
|
|
|
* gst/monoscope/gstmonoscope.c: (gst_monoscope_class_init):
|
|
|
|
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init):
|
|
|
|
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_class_init):
|
|
|
|
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_class_init):
|
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|
|
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_class_init):
|
|
|
|
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_class_init):
|
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|
|
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init):
|
|
|
|
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_class_init):
|
|
|
|
* gst/rtp/gstrtpgsmdepay.c: (gst_rtp_gsm_depay_class_init):
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|
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_class_init):
|
|
|
|
* gst/rtp/gstrtph263pay.c: (gst_rtp_h263_pay_class_init):
|
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|
|
* gst/rtp/gstrtph263pdepay.c: (gst_rtp_h263p_depay_class_init):
|
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|
|
* gst/rtp/gstrtph263ppay.c: (gst_rtp_h263p_pay_class_init):
|
|
|
|
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init):
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_class_init):
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init):
|
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|
|
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_class_init):
|
|
|
|
* gst/rtp/gstrtpmpapay.c: (gst_rtp_mpa_pay_class_init):
|
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|
|
* gst/rtp/gstrtppcmadepay.c: (gst_rtp_pcma_depay_class_init):
|
|
|
|
* gst/rtp/gstrtppcmapay.c: (gst_rtp_pcma_pay_class_init):
|
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|
|
* gst/rtp/gstrtppcmudepay.c: (gst_rtp_pcmu_depay_class_init):
|
|
|
|
* gst/rtp/gstrtppcmupay.c: (gst_rtp_pcmu_pay_class_init):
|
|
|
|
* gst/rtp/gstrtpspeexdepay.c: (gst_rtp_speex_depay_class_init):
|
|
|
|
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_class_init):
|
|
|
|
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_class_init):
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_class_init):
|
|
|
|
* gst/smpte/gstsmpte.c: (gst_smpte_class_init):
|
|
|
|
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
|
|
|
|
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init):
|
|
|
|
* gst/udp/gstudpsink.c: (gst_udpsink_class_init):
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_class_init):
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
|
|
|
|
* sys/oss/gstossdmabuffer.c: (gst_ossdmabuffer_class_init):
|
|
|
|
* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
|
|
|
|
* sys/osxaudio/gstosxaudioelement.c:
|
|
|
|
(gst_osxaudioelement_class_init):
|
|
|
|
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_class_init):
|
|
|
|
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_class_init):
|
|
|
|
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_class_init):
|
|
|
|
Fix #337365 (g_type_class_ref <-> g_type_class_peek_parent)
|
|
|
|
|
2006-04-08 19:06:25 +00:00
|
|
|
2006-04-08 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* ext/mikmod/gstmikmod.h:
|
|
|
|
* gst/level/gstlevel.h:
|
|
|
|
Fix more broken GObject macros
|
|
|
|
|
2006-04-08 18:25:55 +00:00
|
|
|
2006-04-08 Stefan Kost <ensonic@users.sf.net>
|
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|
|
|
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|
|
* ext/annodex/gstcmmldec.h:
|
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|
|
* ext/annodex/gstcmmlenc.h:
|
|
|
|
* ext/annodex/gstcmmltag.h:
|
|
|
|
* ext/cairo/gsttextoverlay.h:
|
|
|
|
* ext/ladspa/gstsignalprocessor.h:
|
|
|
|
* gst/matroska/ebml-read.h:
|
|
|
|
* gst/matroska/ebml-write.h:
|
|
|
|
* sys/osxaudio/gstosxaudioelement.h:
|
|
|
|
Fix broken GObject macros
|
|
|
|
|
2006-04-08 13:09:50 +00:00
|
|
|
2006-04-08 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event):
|
|
|
|
Don't try to seek beyond the end of the file (would
|
|
|
|
occasionally display error dialogs in totem when seeking
|
|
|
|
to the end) (#335869). Will still throw an error though
|
|
|
|
if the file is truncated and the total_samples value in
|
|
|
|
the stream header is wrong.
|
|
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|
|
2006-04-07 18:15:08 +00:00
|
|
|
2006-04-07 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_calculate_crc8),
|
|
|
|
(gst_flac_dec_scan_got_frame), (gst_flac_dec_scan_for_last_block),
|
|
|
|
(gst_flac_dec_metadata_callback):
|
|
|
|
* ext/flac/gstflacdec.h:
|
|
|
|
If the stream header doesn't contain the total number of samples,
|
|
|
|
search for the last flac frame at the end of the file and calculate
|
|
|
|
the total duration from that frame's offset (fixes #337609).
|
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|
|
2006-04-07 10:48:19 +00:00
|
|
|
2006-04-07 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream):
|
|
|
|
Don't unref the GstPadTemplate returned by
|
|
|
|
gst_element_class_get_pad_template().
|
|
|
|
|
2006-04-06 09:14:30 +00:00
|
|
|
2006-04-06 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
|
Patch by: Brian Cameron <brian dot cameron at sun dot com>
|
|
|
|
|
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|
|
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init),
|
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|
|
(gst_sunaudiosink_prepare), (gst_sunaudiosink_write):
|
|
|
|
* sys/sunaudio/gstsunaudiosink.h:
|
|
|
|
Use spec->segsize and spec->segtotal in the prepare function
|
|
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|
to initialise the ring buffer instead of using the buffer-time
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property (#337421).
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2006-04-06 08:52:51 +00:00
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2006-04-06 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Bump core requirements to CVS for gst_pad_query_peer_duration()
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which is used by speexdec.
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ext/speex/: Fix seeking and duration queries (#337033); clean up and refactor a bit.
Original commit message from CVS:
* ext/speex/gstspeex.c: (plugin_init):
* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init),
(gst_speex_dec_reset), (gst_speex_dec_init), (speex_dec_convert),
(speex_get_sink_query_types), (speex_dec_sink_query),
(speex_get_src_query_types), (speex_dec_src_query),
(speex_dec_src_event), (speex_dec_sink_event),
(speex_dec_chain_parse_header), (speex_dec_chain_parse_comments),
(speex_dec_chain_parse_data), (speex_dec_chain),
(gst_speex_dec_get_property), (gst_speex_dec_set_property),
(speex_dec_change_state):
* ext/speex/gstspeexdec.h:
Fix seeking and duration queries (#337033); clean up and
refactor a bit.
2006-04-05 18:27:22 +00:00
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2006-04-05 Tim-Philipp Müller <tim at centricular dot net>
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* ext/speex/gstspeex.c: (plugin_init):
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* ext/speex/gstspeexdec.c: (gst_speex_dec_class_init),
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(gst_speex_dec_reset), (gst_speex_dec_init), (speex_dec_convert),
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(speex_get_sink_query_types), (speex_dec_sink_query),
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(speex_get_src_query_types), (speex_dec_src_query),
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(speex_dec_src_event), (speex_dec_sink_event),
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(speex_dec_chain_parse_header), (speex_dec_chain_parse_comments),
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(speex_dec_chain_parse_data), (speex_dec_chain),
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(gst_speex_dec_get_property), (gst_speex_dec_set_property),
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(speex_dec_change_state):
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* ext/speex/gstspeexdec.h:
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Fix seeking and duration queries (#337033); clean up and
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refactor a bit.
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2006-04-05 12:41:14 +00:00
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2006-04-05 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/raw1394/gstdv1394src.c:
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distinguish between device not found and could not open for
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reading
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2006-04-03 09:02:29 +00:00
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2006-04-03 Thomas Vander Stichele <thomas at apestaart dot org>
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* Makefile.am:
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* configure.ac:
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* pkgconfig/.cvsignore:
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* pkgconfig/Makefile.am:
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* pkgconfig/gstreamer-plugins-good-uninstalled.pc.in:
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add a .pc file so other modules can use good plugins in tests
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2006-04-01 09:56:45 +00:00
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2006-04-01 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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clean up, use AS_VERSION and AS_NANO
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
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use PACKAGE_VERSION define
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* po/af.po:
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* po/az.po:
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* po/cs.po:
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* po/en_GB.po:
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* po/hu.po:
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* po/it.po:
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* po/nb.po:
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* po/nl.po:
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* po/or.po:
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* po/sq.po:
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* po/sr.po:
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* po/sv.po:
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* po/uk.po:
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* po/vi.po:
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updated
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2006-03-30 23:37:16 +00:00
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2006-03-31 Sebastien Moutte <sebastien@moutte.net>
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* ext\jpeg\smokecodec.c:
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use of GST_DEBUG instead of DEBUG(a...) for WIN32
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* ext\speex\gstspeexenc.c: (gst_speexenc_set_header_on_caps):
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move first instruction after all variables declarations
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* gst\alpha\gstalpha.c:
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* gst\effectv\gstshagadelic.c:
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* gst\smpte\paint.c:
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* gst\videofilter\gstvideobalance.c:
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define M_PI if it's not defined (it's not defined on WIN32)
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* gst\cutter\gstcutter.c: (gst_cutter_chain):
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* gst\id3demux\id3v2frames.c: (parse_relative_volume_adjustment_two):
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* gst\level\gstlevel.c: (gst_level_set_property), (gst_level_transform_ip):
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* gst\matroska\matroska-demux.c: (gst_matroska_demux_parse_info),
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(gst_matroska_demux_video_caps):
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* gst\matroska\matroska-mux.c: (gst_matroska_mux_start), (gst_matroska_mux_finish):
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* gst\wavparse\gstwavparse.c: (gst_wavparse_stream_data):
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use gst_guint64_to_gdouble for conversions
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* gst\goom\filters.c: (setPixelRGB_):
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fix a debug which was using undefined variable
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* gst\level\gstlevel.c: (gst_level_set_caps), (gst_level_transform_ip):
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* gst\matroska\ebml-read.c: (gst_ebml_read_sint):
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replace LL suffix with L suffix (LL isn't supported by MSVC6.0)
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* win32/vs6:
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add vs6 projects files for most of plugins-good
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2006-03-30 15:37:05 +00:00
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2006-03-30 j^ <j@bootlab.org>
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* ext/aalib/gstaasink.c:
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* ext/annodex/gstcmmldec.c:
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* ext/annodex/gstcmmlenc.c:
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* ext/cairo/gsttextoverlay.c:
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* ext/cairo/gsttimeoverlay.c:
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* ext/cdio/gstcdiocddasrc.c:
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* ext/dv/gstdvdec.c:
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* ext/esd/esdmon.c:
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* ext/esd/esdsink.c:
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* ext/flac/gstflacdec.c:
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* ext/flac/gstflacenc.c:
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* ext/flac/gstflactag.c:
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_base_init):
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* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init):
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* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_base_init):
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* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init):
|
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* ext/gdk_pixbuf/gstgdkpixbuf.c:
|
|
|
|
* ext/gdk_pixbuf/pixbufscale.c:
|
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|
|
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init):
|
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|
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init):
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* ext/jpeg/gstjpegdec.c:
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|
* ext/jpeg/gstjpegenc.c:
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* ext/jpeg/gstsmokedec.c:
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* ext/jpeg/gstsmokeenc.c:
|
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* ext/libcaca/gstcacasink.c:
|
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|
|
* ext/libmng/gstmngdec.c:
|
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* ext/libmng/gstmngenc.c:
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* ext/libpng/gstpngdec.c:
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* ext/libpng/gstpngenc.c:
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|
* ext/mikmod/gstmikmod.c:
|
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|
|
* ext/raw1394/gstdv1394src.c:
|
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|
|
* ext/shout2/gstshout2.c:
|
|
|
|
* ext/speex/gstspeexdec.c:
|
|
|
|
* ext/speex/gstspeexenc.c:
|
|
|
|
* gst/alpha/gstalpha.c:
|
|
|
|
* gst/alpha/gstalphacolor.c:
|
|
|
|
* gst/auparse/gstauparse.c:
|
|
|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
(gst_auto_audio_sink_base_init):
|
|
|
|
* gst/autodetect/gstautovideosink.c:
|
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|
|
(gst_auto_video_sink_base_init):
|
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|
|
* gst/avi/gstavimux.c: (gst_avimux_base_init):
|
|
|
|
* gst/cutter/gstcutter.c:
|
|
|
|
* gst/debug/breakmydata.c:
|
|
|
|
* gst/debug/efence.c:
|
|
|
|
* gst/debug/gstnavigationtest.c:
|
|
|
|
* gst/debug/negotiation.c:
|
|
|
|
* gst/debug/progressreport.c:
|
|
|
|
* gst/debug/testplugin.c:
|
|
|
|
* gst/effectv/gstaging.c:
|
|
|
|
* gst/effectv/gstdice.c:
|
|
|
|
* gst/effectv/gstedge.c:
|
|
|
|
* gst/effectv/gstquark.c:
|
|
|
|
* gst/effectv/gstrev.c:
|
|
|
|
* gst/effectv/gstvertigo.c:
|
|
|
|
* gst/effectv/gstwarp.c:
|
|
|
|
* gst/flx/gstflxdec.c:
|
|
|
|
* gst/goom/gstgoom.c:
|
|
|
|
* gst/interleave/deinterleave.c:
|
|
|
|
* gst/interleave/interleave.c:
|
|
|
|
* gst/law/alaw-decode.c: (gst_alawdec_base_init):
|
|
|
|
* gst/law/alaw-encode.c: (gst_alawenc_base_init):
|
|
|
|
* gst/law/mulaw-decode.c: (gst_mulawdec_base_init):
|
|
|
|
* gst/law/mulaw-encode.c: (gst_mulawenc_base_init):
|
|
|
|
* gst/level/gstlevel.c:
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_base_init):
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_base_init):
|
|
|
|
* gst/median/gstmedian.c:
|
|
|
|
* gst/monoscope/gstmonoscope.c:
|
|
|
|
* gst/multipart/multipartdemux.c:
|
|
|
|
* gst/multipart/multipartmux.c:
|
|
|
|
* gst/oldcore/gstmd5sink.c:
|
|
|
|
* gst/oldcore/gstmultifilesrc.c:
|
|
|
|
* gst/oldcore/gstpipefilter.c:
|
|
|
|
* gst/oldcore/gstshaper.c:
|
|
|
|
* gst/oldcore/gststatistics.c:
|
|
|
|
* gst/rtp/gstasteriskh263.c:
|
|
|
|
* gst/rtp/gstrtpL16depay.c:
|
|
|
|
* gst/rtp/gstrtpL16pay.c:
|
|
|
|
* gst/rtp/gstrtpamrdepay.c:
|
|
|
|
* gst/rtp/gstrtpamrpay.c:
|
|
|
|
* gst/rtp/gstrtpdepay.c:
|
|
|
|
* gst/rtp/gstrtpgsmpay.c:
|
|
|
|
* gst/rtp/gstrtph263pay.c:
|
|
|
|
* gst/rtp/gstrtph263pdepay.c:
|
|
|
|
* gst/rtp/gstrtph263ppay.c:
|
|
|
|
* gst/rtp/gstrtpmp4gpay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c:
|
|
|
|
* gst/rtp/gstrtpmpadepay.c:
|
|
|
|
* gst/rtp/gstrtpmpapay.c:
|
|
|
|
* gst/rtp/gstrtppcmadepay.c:
|
|
|
|
* gst/rtp/gstrtppcmapay.c:
|
|
|
|
* gst/rtp/gstrtppcmudepay.c:
|
|
|
|
* gst/rtp/gstrtppcmupay.c:
|
|
|
|
* gst/rtp/gstrtpspeexdepay.c:
|
|
|
|
* gst/rtp/gstrtpspeexpay.c:
|
|
|
|
* gst/rtsp/gstrtpdec.c:
|
|
|
|
* gst/smpte/gstsmpte.c:
|
|
|
|
* gst/videobox/gstvideobox.c:
|
|
|
|
* gst/videofilter/gstgamma.c: (gst_gamma_base_init):
|
|
|
|
* gst/videofilter/gstvideobalance.c:
|
|
|
|
* gst/videofilter/gstvideoflip.c:
|
|
|
|
* gst/videofilter/gstvideotemplate.c:
|
|
|
|
(gst_videotemplate_base_init):
|
|
|
|
* gst/videomixer/videomixer.c:
|
|
|
|
* gst/wavenc/gstwavenc.c:
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init):
|
|
|
|
better/unified long descriptions
|
|
|
|
Fixed #336602
|
|
|
|
Some cleanups to auparse, don't send multiple newsegments.
|
|
|
|
|
2006-03-29 16:06:50 +00:00
|
|
|
2006-03-29 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
From a patch by: Michael Dominic K. <mdk at mdk dot org dot pl>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_class_init),
|
|
|
|
(gst_dvdemux_reset), (gst_dvdemux_src_convert),
|
|
|
|
(gst_dvdemux_send_event), (gst_dvdemux_flush), (gst_dvdemux_loop),
|
|
|
|
(gst_dvdemux_sink_activate_pull), (gst_dvdemux_change_state):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
Seek in READY patch. Only works for pull based mode.
|
|
|
|
Fixes #323880
|
|
|
|
|
2006-03-27 14:09:18 +00:00
|
|
|
2006-03-27 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_flush),
|
|
|
|
(gst_gdk_pixbuf_event):
|
|
|
|
Fix two crashers: don't unref the same caps twice, and
|
|
|
|
set pixbuf loader to NULL after freeing it.
|
|
|
|
|
2006-03-27 14:00:02 +00:00
|
|
|
2006-03-27 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/speex/gstspeexenc.c: (gst_speexenc_class_init),
|
|
|
|
(gst_speexenc_finalize), (gst_speexenc_sink_setcaps),
|
|
|
|
(gst_speexenc_chain):
|
|
|
|
* ext/speex/gstspeexenc.h:
|
|
|
|
Don't leak adapter.
|
|
|
|
A push *always* takes ownership of the buffer, even on
|
|
|
|
errors.
|
|
|
|
Small cleanups.
|
|
|
|
|
2006-03-26 12:24:56 +00:00
|
|
|
2006-03-26 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
|
|
|
|
Create source pad without leaking.
|
|
|
|
|
2006-03-24 19:41:03 +00:00
|
|
|
2006-03-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_handle_seek_event):
|
|
|
|
* ext/flac/gstflacdec.h:
|
|
|
|
* ext/flac/gstflacenc.h:
|
|
|
|
Spifify a bit.
|
|
|
|
Fix deadly lock order error in seeking code, STREAM_LOCK
|
|
|
|
cannot be taken within LOCK and the streaming variables are
|
|
|
|
protected with the STREAM_LOCK anyway.
|
|
|
|
|
2006-03-24 18:56:16 +00:00
|
|
|
2006-03-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_index),
|
|
|
|
(gst_avi_demux_stream_index), (gst_avi_demux_stream_scan),
|
|
|
|
(gst_avi_demux_massage_index), (gst_avi_demux_handle_seek):
|
|
|
|
this patch combines the global init_frames with the stream
|
|
|
|
init_frames. Rationale being that the global delay should
|
|
|
|
be subtracted from any stream delay.
|
|
|
|
Fixes #335858.
|
|
|
|
|
2006-03-24 17:11:56 +00:00
|
|
|
2006-03-24 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_init):
|
|
|
|
* gst/multipart/multipartmux.c: (gst_multipart_mux_init):
|
|
|
|
* gst/smpte/gstsmpte.c: (gst_smpte_init):
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_init):
|
|
|
|
use DEBUG_FUNCPTR for collectpads
|
|
|
|
|
2006-03-24 09:54:00 +00:00
|
|
|
2006-03-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init), (gst_jpegenc_chain):
|
|
|
|
Don't crash when encoding images where the number of rows isn't
|
|
|
|
a multiple of 2*DCTSIZE. Add some GST_DEBUG_FUNCPTR.
|
|
|
|
|
2006-03-23 21:28:06 +00:00
|
|
|
2006-03-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/speex/gstspeexdec.c: (speex_dec_change_state):
|
|
|
|
* gst/interleave/deinterleave.c: (deinterleave_change_state):
|
|
|
|
* gst/interleave/interleave.c: (interleave_change_state):
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_change_state):
|
|
|
|
More state change function fixes.
|
|
|
|
|
2006-03-23 20:12:47 +00:00
|
|
|
2006-03-23 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_class_init),
|
|
|
|
(gst_esdsink_getcaps), (gst_esdsink_open), (gst_esdsink_close),
|
|
|
|
(gst_esdsink_prepare), (gst_esdsink_unprepare),
|
|
|
|
(gst_esdsink_delay), (gst_esdsink_reset):
|
|
|
|
* ext/esd/esdsink.h:
|
|
|
|
Fix esd choppy playback by configuring audiosink
|
|
|
|
correctly. Fixes #325191
|
|
|
|
|
2006-03-23 19:57:34 +00:00
|
|
|
2006-03-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_change_state):
|
|
|
|
Make state change function thread-safe.
|
|
|
|
|
2006-03-23 16:50:32 +00:00
|
|
|
2006-03-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_get_upstream_size),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_stream_data):
|
|
|
|
Don't try to read beyond the end of the file just because
|
|
|
|
the header claims a bigger size (like with truncated files).
|
|
|
|
|
2006-03-23 15:36:27 +00:00
|
|
|
2006-03-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_add_src_pad),
|
|
|
|
(gst_wavparse_stream_data), (gst_wavparse_loop):
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
Delay source pad creation until we have the first chunk of
|
|
|
|
media data, so the we can examine the data and adjust the
|
|
|
|
caps accordingly if required. This makes playback of .wav
|
|
|
|
files with DTS-declared-as-PCM content work (#313266).
|
|
|
|
|
2006-03-22 13:00:34 +00:00
|
|
|
2006-03-22 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain):
|
|
|
|
Don't attempt typefinding on too-short buffers that have been
|
2006-03-22 13:02:11 +00:00
|
|
|
completely trimmed away. (Fixes #330239)
|
2006-03-22 13:00:34 +00:00
|
|
|
|
|
|
|
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
|
|
|
|
Improve the debug output
|
|
|
|
|
2006-03-21 18:12:59 +00:00
|
|
|
2006-03-21 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/esd/esdsink.c: (gst_esdsink_class_init), (gst_esdsink_init),
|
|
|
|
(gst_esdsink_finalize), (gst_esdsink_getcaps), (gst_esdsink_open),
|
|
|
|
(gst_esdsink_close), (gst_esdsink_prepare), (gst_esdsink_write),
|
|
|
|
(gst_esdsink_set_property), (gst_esdsink_get_property):
|
|
|
|
Some cleanups.
|
|
|
|
Reset fd to -1 when we close them.
|
|
|
|
|
2006-03-21 16:19:37 +00:00
|
|
|
2006-03-21 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_open):
|
|
|
|
the OPTIONS request result is optional so don't
|
|
|
|
fail on it.
|
|
|
|
|
2006-03-21 14:53:36 +00:00
|
|
|
2006-03-21 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_reset):
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_reset):
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_create_sourcepad),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_send_event),
|
|
|
|
(gst_wavparse_change_state):
|
|
|
|
gcc 4.1 unreferenced pointer fixes.
|
|
|
|
|
2006-03-21 13:07:31 +00:00
|
|
|
2006-03-21 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
Patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek):
|
|
|
|
Fix block alignment calculation. Alignment should be done before
|
|
|
|
adding the byte offset where the data starts (#335231).
|
|
|
|
|
2006-03-20 18:34:21 +00:00
|
|
|
2006-03-20 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/matroska/ebml-write.c: (gst_ebml_write_element_push):
|
|
|
|
Ensure that we set correct caps on buffers that are transferred
|
|
|
|
direct from the input.
|
|
|
|
|
2006-03-20 17:38:48 +00:00
|
|
|
2006-03-20 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/goom/filters.c: (zoomFilterDestroy):
|
|
|
|
* gst/goom/goom_core.c: (goom_close):
|
|
|
|
Free filter data when cleaning up. (Fixes: #334995)
|
|
|
|
|
2006-03-17 15:33:08 +00:00
|
|
|
2006-03-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Don't compile udp and rtsp plugins on win32 (mingw) or other
|
|
|
|
systems that don't have <sys/socket.h> for some reason (#316203).
|
|
|
|
|
2006-03-16 17:28:07 +00:00
|
|
|
2006-03-16 Zaheer Abbas Merali <zaheerabbas at merali dot org>
|
|
|
|
|
|
|
|
* ext/raw1394/gstdv1394src.c: (gst_dv1394src_bus_reset),
|
|
|
|
(gst_dv1394src_discover_avc_node), (gst_dv1394src_start):
|
|
|
|
* ext/raw1394/gstdv1394src.h:
|
|
|
|
Change bus reset handler so it reports useful information such as
|
|
|
|
whether the device being used connected or disconnected
|
|
|
|
|
2006-03-16 16:06:22 +00:00
|
|
|
2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c:
|
|
|
|
(parse_relative_volume_adjustment_two):
|
|
|
|
We only care about gain and peak data for the master volume.
|
|
|
|
|
2006-03-16 13:22:28 +00:00
|
|
|
2006-03-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(parse_id_string), (parse_unique_file_identifier),
|
|
|
|
(parse_relative_volume_adjustment_two), (id3v2_tag_to_taglist):
|
|
|
|
Read replay gain tags (#323721).
|
|
|
|
|
2006-03-15 23:19:30 +00:00
|
|
|
2006-03-15 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Bump requirements to gst-plugins-base CVS because
|
|
|
|
of buggy gst_tag_from_id3_user_tag() in 0.10.5.
|
|
|
|
|
2006-03-15 22:30:24 +00:00
|
|
|
2006-03-15 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
|
|
|
|
|
|
|
|
* rtp/gst/gstrtppcmadepay.c:
|
|
|
|
Fixed one of the caps in the code from mulaw to alaw.
|
|
|
|
|
2006-03-15 16:21:38 +00:00
|
|
|
2006-03-15 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain):
|
|
|
|
Ensure that we set caps on the buffers we pass.
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
|
|
|
|
(gst_id3demux_sink_activate):
|
|
|
|
Ensure that we set caps on the buffers we pass.
|
|
|
|
|
|
|
|
Use STREAM, TYPE_NOT_FOUND as the error class when
|
|
|
|
typefinding fails.
|
|
|
|
|
2006-03-15 16:17:12 +00:00
|
|
|
2006-03-15 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_init):
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_init), (gst_dvdemux_add_pads):
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init),
|
|
|
|
(gst_jpeg_dec_setcaps):
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_init):
|
|
|
|
* ext/jpeg/gstsmokedec.c: (gst_smokedec_init):
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_init):
|
|
|
|
* ext/libmng/gstmngdec.c: (gst_mngdec_init),
|
|
|
|
(gst_mngdec_src_getcaps):
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_init),
|
|
|
|
(gst_pngdec_caps_create_and_set):
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_init):
|
|
|
|
* ext/mikmod/gstmikmod.c: (gst_mikmod_init):
|
|
|
|
* ext/speex/gstspeexdec.c: (gst_speex_dec_init):
|
|
|
|
* gst/alpha/gstalpha.c: (gst_alpha_init):
|
|
|
|
* gst/auparse/gstauparse.c: (gst_au_parse_init):
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_init),
|
|
|
|
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream):
|
|
|
|
* gst/cutter/gstcutter.c: (gst_cutter_init):
|
|
|
|
* gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
|
|
|
|
(gst_efence_checkgetrange):
|
|
|
|
* gst/debug/negotiation.c: (gst_negotiation_init):
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_init):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_init):
|
|
|
|
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_init):
|
|
|
|
* gst/rtp/gstrtpL16depay.c: (gst_rtp_L16depay_init):
|
|
|
|
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_init):
|
|
|
|
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_init):
|
|
|
|
* gst/rtp/gstrtpdepay.c: (gst_rtp_depay_init):
|
|
|
|
* gst/rtp/gstrtpmpadepay.c: (gst_rtp_mpa_depay_init):
|
|
|
|
* gst/rtsp/gstrtpdec.c: (gst_rtpdec_init):
|
|
|
|
* gst/smpte/gstsmpte.c: (gst_smpte_init):
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_init),
|
|
|
|
(gst_wavparse_create_sourcepad):
|
|
|
|
Fix memleak with gst_static_pad_template_get().
|
|
|
|
This uses gst_pad_new_from_static_template() instead.
|
|
|
|
Fixes #333512
|
|
|
|
|
2006-03-14 17:56:02 +00:00
|
|
|
2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Bump -base requirement to 0.10.5 for gst_tag_from_id3_user_tag(),
|
|
|
|
used by id3demux.
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (plugin_init):
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(parse_user_text_identification_frame),
|
|
|
|
(parse_unique_file_identifier):
|
|
|
|
Add support for UFID and TXXX frames and extract musicbrainz tags.
|
|
|
|
|
2006-03-14 15:28:00 +00:00
|
|
|
2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
|
|
|
|
Catch short reads, like they might happen with truncated
|
|
|
|
files (see #305279); remove unnecessary indentation.
|
|
|
|
|
2006-03-14 14:18:16 +00:00
|
|
|
2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
|
|
|
|
Fix DIB image inversion for pictures with a
|
|
|
|
depth != 8 (#305279).
|
|
|
|
|
2006-03-14 09:23:09 +00:00
|
|
|
2006-03-14 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_class_init),
|
|
|
|
(gst_jpeg_dec_chain), (gst_jpeg_dec_change_state):
|
|
|
|
* ext/jpeg/gstjpegdec.h:
|
|
|
|
Fix durations on outgoing buffers after seeking
|
|
|
|
in MJPEG files (#334083); some minor clean-ups.
|
|
|
|
|
2006-03-13 18:28:18 +00:00
|
|
|
2006-03-13 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
|
|
|
|
(gst_wavparse_change_state):
|
|
|
|
Implement seek in READY (re-fixes #327658)
|
|
|
|
|
close #333784 unref the result of gst_pad_get_parent() by: Christophe Fergeau.
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps):
* ext/esd/esdmon.c: (gst_esdmon_get):
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
(gst_gdk_pixbuf_sink_getcaps):
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps),
(gst_jpegenc_setcaps):
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
(gst_smokeenc_setcaps):
* ext/libmng/gstmngdec.c: (gst_mngdec_sinklink),
(gst_mngdec_src_getcaps):
* ext/libmng/gstmngenc.c: (gst_mngenc_sinklink),
(gst_mngenc_chain):
* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps):
* ext/mikmod/gstmikmod.c: (gst_mikmod_srclink):
* ext/speex/gstspeexdec.c: (speex_dec_convert),
(speex_dec_src_event), (speex_dec_chain):
* gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect),
(gst_avimux_audsinkconnect), (gst_avimux_handle_event):
* gst/debug/negotiation.c: (gst_negotiation_getcaps),
(gst_negotiation_pad_link), (gst_negotiation_chain):
* gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler),
(gst_flxdec_chain):
* gst/interleave/deinterleave.c: (deinterleave_sink_link),
(deinterleave_chain):
* gst/law/mulaw-encode.c: (mulawenc_setcaps):
* gst/median/gstmedian.c: (gst_median_link):
* gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect),
(gst_monoscope_chain):
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect):
* gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps):
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain):
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get):
close #333784 unref the result of gst_pad_get_parent()
by: Christophe Fergeau.
2006-03-13 15:49:08 +00:00
|
|
|
2006-03-13 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_setcaps):
|
|
|
|
* ext/esd/esdmon.c: (gst_esdmon_get):
|
|
|
|
* ext/flac/gstflactag.c: (gst_flac_tag_chain):
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_setcaps),
|
|
|
|
(gst_gdk_pixbuf_sink_getcaps):
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_getcaps),
|
|
|
|
(gst_jpegenc_setcaps):
|
|
|
|
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_getcaps),
|
|
|
|
(gst_smokeenc_setcaps):
|
|
|
|
* ext/libmng/gstmngdec.c: (gst_mngdec_sinklink),
|
|
|
|
(gst_mngdec_src_getcaps):
|
|
|
|
* ext/libmng/gstmngenc.c: (gst_mngenc_sinklink),
|
|
|
|
(gst_mngenc_chain):
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps):
|
|
|
|
* ext/mikmod/gstmikmod.c: (gst_mikmod_srclink):
|
|
|
|
* ext/speex/gstspeexdec.c: (speex_dec_convert),
|
|
|
|
(speex_dec_src_event), (speex_dec_chain):
|
|
|
|
* gst/avi/gstavimux.c: (gst_avimux_vidsinkconnect),
|
|
|
|
(gst_avimux_audsinkconnect), (gst_avimux_handle_event):
|
|
|
|
* gst/debug/negotiation.c: (gst_negotiation_getcaps),
|
|
|
|
(gst_negotiation_pad_link), (gst_negotiation_chain):
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler),
|
|
|
|
(gst_flxdec_chain):
|
|
|
|
* gst/interleave/deinterleave.c: (deinterleave_sink_link),
|
|
|
|
(deinterleave_chain):
|
|
|
|
* gst/law/mulaw-encode.c: (mulawenc_setcaps):
|
|
|
|
* gst/median/gstmedian.c: (gst_median_link):
|
|
|
|
* gst/monoscope/gstmonoscope.c: (gst_monoscope_srcconnect),
|
|
|
|
(gst_monoscope_chain):
|
|
|
|
* gst/rtp/gstrtpL16pay.c: (gst_rtpL16pay_sinkconnect):
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_sink_setcaps):
|
|
|
|
* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_chain):
|
|
|
|
* sys/osxaudio/gstosxaudiosrc.c: (gst_osxaudiosrc_get):
|
|
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|
close #333784 unref the result of gst_pad_get_parent()
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|
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|
by: Christophe Fergeau.
|
|
|
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|
2006-03-09 11:47:32 +00:00
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|
2006-03-09 Wim Taymans <wim@fluendo.com>
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|
|
|
|
|
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* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event),
|
|
|
|
(gst_dvdemux_convert_segment), (gst_dvdemux_demux_frame):
|
|
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|
Handle events in push mode better, can now do non-flushing
|
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|
seeks in push mode as well.
|
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|
2006-03-07 17:31:03 +00:00
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|
2006-03-07 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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* gst/udp/gstdynudpsink.c: (gst_dynudpsink_class_init):
|
|
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|
Applied patch from Kai Vehmanen, fixes #333624.
|
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|
|
|
2006-03-06 22:22:45 +00:00
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|
|
2006-03-06 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
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|
* ext/libpng/gstpngdec.c: (gst_pngdec_caps_create_and_set):
|
|
|
|
Implement paletted and grayscale png files handling.
|
|
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|
(#150363).
|
|
|
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|
2006-03-06 00:10:29 +00:00
|
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|
2006-03-06 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
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|
* ext/speex/gstspeexenc.c: (gst_speexenc_set_header_on_caps),
|
|
|
|
(gst_speexenc_chain):
|
|
|
|
fix a tag list assert
|
|
|
|
follow gst-plugins-base/ext/ogg/README; set OFFSET
|
|
|
|
and OFFSET_END. Muxes correctly with gst-plugins-base
|
|
|
|
> 0.9.3
|
|
|
|
|
2006-03-04 20:11:35 +00:00
|
|
|
2006-03-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/Makefile.am:
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad),
|
|
|
|
(gst_id3demux_chain), (gst_id3demux_sink_activate):
|
|
|
|
Use new typefind helper functions here as well, and
|
|
|
|
do typefinding in pull-mode if upstream supports that.
|
|
|
|
|
2006-03-04 18:57:37 +00:00
|
|
|
2006-03-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_get_volume),
|
|
|
|
(gst_sunaudiomixer_ctrl_set_volume):
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.c:
|
|
|
|
(gst_sunaudiomixer_track_new):
|
|
|
|
Remove unused variables, breaks build from CVS
|
|
|
|
with -Werror (#333392, patch by: Benjamin Pineau)
|
|
|
|
|
docs/plugins/: Added wavparse docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Added wavparse docs.
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
(gst_wavparse_reset), (gst_wavparse_init),
(gst_wavparse_create_sourcepad), (gst_wavparse_parse_file_header),
(gst_wavparse_stream_init), (gst_wavparse_perform_seek),
(gst_wavparse_stream_headers), (gst_wavparse_send_event),
(gst_wavparse_stream_data), (gst_wavparse_loop),
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
Implement seek in READY (fixes #327658)
Added docs and did some cleanups.
2006-03-03 18:36:53 +00:00
|
|
|
2006-03-03 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
Added wavparse docs.
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_class_init),
|
|
|
|
(gst_wavparse_reset), (gst_wavparse_init),
|
|
|
|
(gst_wavparse_create_sourcepad), (gst_wavparse_parse_file_header),
|
|
|
|
(gst_wavparse_stream_init), (gst_wavparse_perform_seek),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_send_event),
|
|
|
|
(gst_wavparse_stream_data), (gst_wavparse_loop),
|
|
|
|
(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull),
|
|
|
|
(gst_wavparse_change_state):
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
Implement seek in READY (fixes #327658)
|
|
|
|
Added docs and did some cleanups.
|
|
|
|
|
2006-03-03 17:51:16 +00:00
|
|
|
2006-03-03 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
|
|
|
|
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_calculate_durations_from_index),
|
|
|
|
(gst_avi_demux_stream_header):
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
If we have an index, use a duration based on the index instead
|
|
|
|
of blindly trusting the information in the stream headers
|
|
|
|
(fixes #331817).
|
|
|
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|
2006-03-03 15:50:40 +00:00
|
|
|
2006-03-03 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
|
|
|
Added smoke and jpeg to the docs.
|
|
|
|
|
|
|
|
* ext/jpeg/Makefile.am:
|
|
|
|
* ext/jpeg/gstjpeg.c: (plugin_init):
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
|
|
|
|
* ext/jpeg/gstjpegenc.h:
|
|
|
|
* ext/jpeg/gstsmokedec.c: (gst_smokedec_init),
|
|
|
|
(gst_smokedec_chain):
|
|
|
|
* ext/jpeg/gstsmokedec.h:
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
|
|
|
|
* ext/jpeg/gstsmokeenc.h:
|
|
|
|
* ext/jpeg/smokecodec.h:
|
|
|
|
Port smokedec (fixes #331905).
|
|
|
|
Added some docs.
|
|
|
|
Some cleanups.
|
|
|
|
|
docs/plugins/: Added videobalance and videoflip to the docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
Added videobalance and videoflip to the docs.
* gst/videofilter/Makefile.am:
* gst/videofilter/gstvideobalance.c:
(gst_video_balance_update_tables_planar411),
(gst_video_balance_is_passthrough),
(gst_video_balance_update_properties), (oil_tablelookup_u8),
(gst_video_balance_planar411_ip), (gst_video_balance_set_caps),
(gst_video_balance_transform_ip), (gst_video_balance_base_init),
(gst_video_balance_finalize), (gst_video_balance_class_init),
(gst_video_balance_init), (gst_video_balance_interface_supported),
(gst_video_balance_interface_init),
(gst_video_balance_colorbalance_list_channels),
(gst_video_balance_colorbalance_set_value),
(gst_video_balance_colorbalance_get_value),
(gst_video_balance_colorbalance_init),
(gst_video_balance_set_property), (gst_video_balance_get_property),
(gst_video_balance_get_type), (plugin_init):
* gst/videofilter/gstvideobalance.h:
Ported to 0.10. (Fixes #326160)
Added docs.
* gst/videofilter/gstvideoflip.c:
* gst/videofilter/gstvideoflip.h:
Added docs.
2006-03-03 14:39:55 +00:00
|
|
|
2006-03-03 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
|
|
|
Added videobalance and videoflip to the docs.
|
|
|
|
|
|
|
|
* gst/videofilter/Makefile.am:
|
|
|
|
* gst/videofilter/gstvideobalance.c:
|
|
|
|
(gst_video_balance_update_tables_planar411),
|
|
|
|
(gst_video_balance_is_passthrough),
|
|
|
|
(gst_video_balance_update_properties), (oil_tablelookup_u8),
|
|
|
|
(gst_video_balance_planar411_ip), (gst_video_balance_set_caps),
|
|
|
|
(gst_video_balance_transform_ip), (gst_video_balance_base_init),
|
|
|
|
(gst_video_balance_finalize), (gst_video_balance_class_init),
|
|
|
|
(gst_video_balance_init), (gst_video_balance_interface_supported),
|
|
|
|
(gst_video_balance_interface_init),
|
|
|
|
(gst_video_balance_colorbalance_list_channels),
|
|
|
|
(gst_video_balance_colorbalance_set_value),
|
|
|
|
(gst_video_balance_colorbalance_get_value),
|
|
|
|
(gst_video_balance_colorbalance_init),
|
|
|
|
(gst_video_balance_set_property), (gst_video_balance_get_property),
|
|
|
|
(gst_video_balance_get_type), (plugin_init):
|
|
|
|
* gst/videofilter/gstvideobalance.h:
|
|
|
|
Ported to 0.10. (Fixes #326160)
|
|
|
|
Added docs.
|
|
|
|
|
|
|
|
* gst/videofilter/gstvideoflip.c:
|
|
|
|
* gst/videofilter/gstvideoflip.h:
|
|
|
|
Added docs.
|
|
|
|
|
2006-03-02 15:14:22 +00:00
|
|
|
2006-03-02 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Bump requirements to current core and -base CVS
|
|
|
|
(core for new typefind helper API, and -base for the
|
|
|
|
WAVFORMATEX support that was added to libgstriff and
|
|
|
|
is needed by wavparse).
|
|
|
|
|
|
|
|
* gst/apetag/Makefile.am:
|
|
|
|
* gst/apetag/gsttagdemux.c: (gst_tag_demux_chain),
|
|
|
|
(gst_tag_demux_sink_activate):
|
|
|
|
Use new typefind helpers for typefinding instead of our
|
|
|
|
home-grown stuff; also, do typefinding in pull-mode if
|
|
|
|
upstream supports that.
|
|
|
|
|
2006-02-28 10:39:08 +00:00
|
|
|
2006-02-28 Jürg Billeter <j (at) bitron.ch>
|
|
|
|
|
|
|
|
Reviewed by: Christian Schaller <christian@fluendo.com>
|
|
|
|
|
|
|
|
This patch fixes bug: 329107
|
|
|
|
|
|
|
|
This Changelog entry is for a commit done on February 17
|
|
|
|
|
|
|
|
* ext/gconf/gconf.c
|
|
|
|
* ext/gconf/gconf.h
|
|
|
|
* ext/gconf/gstgconfaudiosink.c
|
|
|
|
* ext/gconf/gstgconfaudiosink.h
|
|
|
|
* gconf/gstreamer.schemas.in
|
|
|
|
|
2006-02-28 10:29:16 +00:00
|
|
|
2006-02-28 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_seek),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
|
|
|
|
(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event),
|
|
|
|
(gst_wavparse_sink_activate), (gst_wavparse_sink_activate_pull):
|
|
|
|
Use DEBUG_OBJECT more.
|
|
|
|
|
docs/plugins/: Added dvdec and dvdemux to docs.
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
Added dvdec and dvdemux to docs.
* ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_chain):
Added docs.
Check frame sizes so we don't crash when don't have enough
data.
Send nice error messages on error.
* ext/dv/gstdvdemux.c: (gst_dvdemux_base_init),
(gst_dvdemux_class_init), (gst_dvdemux_init),
(gst_dvdemux_finalize), (gst_dvdemux_reset),
(gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
(gst_dvdemux_src_query), (gst_dvdemux_sink_query),
(gst_dvdemux_push_event), (gst_dvdemux_handle_sink_event),
(gst_dvdemux_convert_src_pair), (gst_dvdemux_convert_sink_pair),
(gst_dvdemux_convert_src_to_sink), (gst_dvdemux_handle_push_seek),
(gst_dvdemux_do_seek), (gst_dvdemux_handle_pull_seek),
(gst_dvdemux_handle_src_event), (gst_dvdemux_demux_audio),
(gst_dvdemux_demux_video), (gst_dvdemux_demux_frame),
(gst_dvdemux_flush), (gst_dvdemux_chain), (gst_dvdemux_loop),
(gst_dvdemux_sink_activate_push), (gst_dvdemux_sink_activate_pull),
(gst_dvdemux_sink_activate), (gst_dvdemux_change_state):
* ext/dv/gstdvdemux.h:
Added docs.
Implement pull mode.
Fix memleaks.
Reduce memcpy for the video demuxing.
2006-02-28 10:22:11 +00:00
|
|
|
2006-02-28 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
Added dvdec and dvdemux to docs.
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_chain):
|
|
|
|
Added docs.
|
|
|
|
Check frame sizes so we don't crash when don't have enough
|
|
|
|
data.
|
|
|
|
Send nice error messages on error.
|
|
|
|
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_base_init),
|
|
|
|
(gst_dvdemux_class_init), (gst_dvdemux_init),
|
|
|
|
(gst_dvdemux_finalize), (gst_dvdemux_reset),
|
|
|
|
(gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
|
|
|
|
(gst_dvdemux_src_query), (gst_dvdemux_sink_query),
|
|
|
|
(gst_dvdemux_push_event), (gst_dvdemux_handle_sink_event),
|
|
|
|
(gst_dvdemux_convert_src_pair), (gst_dvdemux_convert_sink_pair),
|
|
|
|
(gst_dvdemux_convert_src_to_sink), (gst_dvdemux_handle_push_seek),
|
|
|
|
(gst_dvdemux_do_seek), (gst_dvdemux_handle_pull_seek),
|
|
|
|
(gst_dvdemux_handle_src_event), (gst_dvdemux_demux_audio),
|
|
|
|
(gst_dvdemux_demux_video), (gst_dvdemux_demux_frame),
|
|
|
|
(gst_dvdemux_flush), (gst_dvdemux_chain), (gst_dvdemux_loop),
|
|
|
|
(gst_dvdemux_sink_activate_push), (gst_dvdemux_sink_activate_pull),
|
|
|
|
(gst_dvdemux_sink_activate), (gst_dvdemux_change_state):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
Added docs.
|
|
|
|
Implement pull mode.
|
|
|
|
Fix memleaks.
|
|
|
|
Reduce memcpy for the video demuxing.
|
|
|
|
|
2006-02-28 09:21:27 +00:00
|
|
|
2006-02-28 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/annodex/gstcmmldec.c: (gst_cmml_dec_sink_event),
|
|
|
|
(gst_cmml_dec_new_buffer), (gst_cmml_dec_parse_preamble),
|
|
|
|
(gst_cmml_dec_parse_head), (gst_cmml_dec_push_clip):
|
|
|
|
* ext/annodex/gstcmmlparser.c: (gst_cmml_parser_parse_chunk):
|
|
|
|
Add a little extra debug. Make the decoder not return NOT_LINKED,
|
|
|
|
as we want to continue decoding all CMML and emitting tags.
|
|
|
|
|
2006-02-27 14:00:18 +00:00
|
|
|
2006-02-27 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
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|
* ext/annodex/gstskeltag.c:
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* ext/annodex/gstskeltag.h:
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Deleted; these files aren't used any more either.
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2006-02-25 20:37:29 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* ext/Makefile.am: Fix dist-check.
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2006-02-25 19:36:24 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_push_clip): Fix another
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memleak.
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2006-02-25 19:07:41 +00:00
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2006-02-25 Alessandro Decina <alessandro@nnva.org>
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* ext/annodex/Makefile.am:
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* ext/annodex/gstannodex.c:
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* ext/annodex/gstcmmldec.c:
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* ext/annodex/gstcmmlenc.c:
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* ext/annodex/gstcmmlparser.c:
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* ext/annodex/gstcmmlparser.h:
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* ext/annodex/gstcmmlutils.c:
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* tests/check/elements/cmmldec.c:
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* tests/check/elements/cmmlenc.c:
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Fix a memleak in gst_cmml_track_list_add_clip.
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Handle overflows in clip's start and end times.
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Add the "encoded" parameter to cmmldec and cmmlenc caps.
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Do not parse junk at the end of a CMML preamble buffer.
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Register a libxml error handler to not print stuff on stderr.
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Check for bad clip start and end times in the testsuites.
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2006-02-25 11:37:10 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* ext/annodex/gstcmmldec.c: (gst_cmml_dec_class_init),
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(gst_cmml_dec_finalize), (gst_cmml_dec_change_state):
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* ext/annodex/gstcmmlenc.c: (gst_cmml_enc_class_init),
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(gst_cmml_enc_finalize), (gst_cmml_enc_change_state):
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* ext/annodex/gstcmmlutils.c: (gst_cmml_track_list_destroy): Fix
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possible memleaks.
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2006-02-24 23:52:28 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* tests/check/Makefile.am:
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* tests/check/elements/cmmldec.c:
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* tests/check/elements/cmmlenc.c: Fix tests so that they use
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the plugins-base tags.
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2006-02-24 23:36:58 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* ext/Makefile.am: Re-enable module.
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|
2006-02-24 23:32:14 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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|
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* tests/check/Makefile.am: Forgot to remove that test.
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2006-02-24 23:31:08 +00:00
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2006-02-25 Julien MOUTTE <julien@moutte.net>
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* ext/annodex/Makefile.am:
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* ext/annodex/gstannodex.c: (plugin_init):
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* ext/annodex/gstcmmldec.c:
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* ext/annodex/gstskeldec.c:
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* ext/annodex/gstskeldec.h:
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* tests/check/Makefile.am:
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* tests/check/elements/skeldec.c: Try to fix Annodex plugin.
|
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|
2006-02-24 23:06:27 +00:00
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|
2006-02-25 Julien MOUTTE <julien@moutte.net>
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|
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* tests/check/Makefile.am: Disable those checks as well.
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|
2006-02-24 22:49:29 +00:00
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|
2006-02-24 Julien MOUTTE <julien@moutte.net>
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* ext/Makefile.am: Disable annodex for now until we figure out
|
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|
|
how to make it build.
|
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* ext/gdk_pixbuf/Makefile.am: Note for Thomas :
|
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|
Add a rule to your checklist : "please try to at least build
|
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|
|
what you are going to commit into -good, or if you are too lazy
|
|
|
|
to do that, please check that the buildbots are not crying because
|
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|
|
of your commit."
|
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|
2006-02-24 19:51:29 +00:00
|
|
|
2006-02-24 Edgard Lima <edgard.lima@indt.org.br>
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* configure.ac:
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|
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* ext/Makefile.am:
|
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* ext/gdk_pixbuf/Makefile.am:
|
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* ext/gdk_pixbuf/gstgdkpixbuf.c:
|
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|
|
* ext/gdk_pixbuf/gstgdkpixbuf.h:
|
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|
|
* ext/gdk_pixbuf/pixbufscale.c:
|
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|
|
* ext/gdk_pixbuf/pixbufscale.h:
|
|
|
|
Gdkpixbuf ported from 0.8 to 0.10 by
|
|
|
|
Renato Filho <renato.filho@indt.org.br>.
|
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|
|
gst_loader and gdkpixbufanimation still need port.
|
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|
|
2006-02-24 19:07:10 +00:00
|
|
|
2006-02-24 Michael Smith <msmith@fluendo.com>
|
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|
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|
|
|
|
* configure.ac:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* ext/Makefile.am:
|
|
|
|
* ext/annodex/Makefile.am:
|
|
|
|
* ext/annodex/gstannodex.c:
|
|
|
|
* ext/annodex/gstannodex.h:
|
|
|
|
* ext/annodex/gstcmmldec.c:
|
|
|
|
* ext/annodex/gstcmmldec.h:
|
|
|
|
* ext/annodex/gstcmmlenc.c:
|
|
|
|
* ext/annodex/gstcmmlenc.h:
|
|
|
|
* ext/annodex/gstcmmlparser.c:
|
|
|
|
* ext/annodex/gstcmmlparser.h:
|
|
|
|
* ext/annodex/gstcmmltag.c:
|
|
|
|
* ext/annodex/gstcmmltag.h:
|
|
|
|
* ext/annodex/gstcmmlutils.c:
|
|
|
|
* ext/annodex/gstcmmlutils.h:
|
|
|
|
* ext/annodex/gstskeldec.c:
|
|
|
|
* ext/annodex/gstskeldec.h:
|
|
|
|
* ext/annodex/gstskeltag.c:
|
|
|
|
* ext/annodex/gstskeltag.h:
|
|
|
|
* tests/check/Makefile.am:
|
|
|
|
* tests/check/elements/cmmldec.c:
|
|
|
|
* tests/check/elements/cmmlenc.c:
|
|
|
|
* tests/check/elements/skeldec.c:
|
|
|
|
Add Annodex elements from Alessendro Decina: skeleton and CMML.
|
|
|
|
Includes tests & docs, oh my! Passes Thomas's -good checklist
|
|
|
|
entirely. Wow.
|
|
|
|
|
2006-02-24 17:09:56 +00:00
|
|
|
2006-02-24 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* autogen.sh:
|
|
|
|
Check for automake 1.9 as well.
|
|
|
|
|
2006-02-24 14:49:48 +00:00
|
|
|
2006-02-24 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
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|
* ext/flac/gstflacenc.c:
|
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|
|
Change min. sample rate to 8kHz to match flacdec's.
|
|
|
|
|
2006-02-23 20:08:58 +00:00
|
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|
2006-02-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
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|
|
* ext/cdio/Makefile.am:
|
|
|
|
Add GST_BASE_CFLAGS and GST_BASE_LIBS (seems to be
|
|
|
|
required for Cygwin, see #317048)
|
|
|
|
|
|
|
|
* gst/rtp/gstasteriskh263.c:
|
|
|
|
Cygwin has includes for both the unix network socket API
|
|
|
|
and the windows API, but only one can be included, so fix
|
|
|
|
includes to only use one or the other, prefering the unxi
|
|
|
|
one (#317048).
|
|
|
|
|
2006-02-23 12:21:25 +00:00
|
|
|
2006-02-23 Philippe Kalaf <philippe.kalaf at collabora.co.uk>
|
|
|
|
|
|
|
|
* rtp/gst/gstrtppcmadepay.c:
|
|
|
|
* rtp/gst/gstrtppcmadepay.h:
|
|
|
|
* rtp/gst/gstgstrtppcmapay.c:
|
|
|
|
* rtp/gst/gstgstrtppcmapay.h:
|
|
|
|
* rtp/gst/gstrtppcmudepay.c:
|
|
|
|
* rtp/gst/gstrtppcmudepay.h:
|
|
|
|
* rtp/gst/gstrtppcmupay.c:
|
|
|
|
* rtp/gst/gstrtppcmupay.h:
|
|
|
|
* rtp/gst/Makefile.am:
|
|
|
|
* rtp/gst/gstrtp.c:
|
|
|
|
* rtp/gst/README:
|
|
|
|
Separated the G711 payloaders/depayloaders into separate elements for
|
|
|
|
mulaw/alaw. Also removed the old g711 payloaders/depayloaders.
|
|
|
|
|
ext/dv/: Ueber spiffify some more, added debug category.
Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_init),
(gst_dvdec_change_state):
* ext/dv/gstdvdec.h:
* ext/dv/gstdvdemux.c: (gst_dvdemux_base_init), (gst_dvdemux_init),
(gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
(gst_dvdemux_src_query), (gst_dvdemux_sink_query),
(gst_dvdemux_handle_sink_event), (gst_dvdemux_demux_frame),
(gst_dvdemux_flush), (gst_dvdemux_chain),
(gst_dvdemux_change_state):
* ext/dv/gstdvdemux.h:
Ueber spiffify some more, added debug category.
Use _scale.
Use segments, respect playback rate from newsegment.
Fix refcount issue.
2006-02-22 20:22:25 +00:00
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|
2006-02-22 Wim Taymans <wim@fluendo.com>
|
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|
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|
* ext/dv/gstdvdec.c: (gst_dvdec_base_init), (gst_dvdec_init),
|
|
|
|
(gst_dvdec_change_state):
|
|
|
|
* ext/dv/gstdvdec.h:
|
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|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_base_init), (gst_dvdemux_init),
|
|
|
|
(gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
|
|
|
|
(gst_dvdemux_src_query), (gst_dvdemux_sink_query),
|
|
|
|
(gst_dvdemux_handle_sink_event), (gst_dvdemux_demux_frame),
|
|
|
|
(gst_dvdemux_flush), (gst_dvdemux_chain),
|
|
|
|
(gst_dvdemux_change_state):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
Ueber spiffify some more, added debug category.
|
|
|
|
Use _scale.
|
|
|
|
Use segments, respect playback rate from newsegment.
|
|
|
|
Fix refcount issue.
|
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|
|
|
2006-02-20 13:08:50 +00:00
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|
2006-02-20 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
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* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event),
|
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|
|
(gst_signal_processor_process):
|
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|
|
Fix compilation of LADPSA. It doesn't seem to work, and isn't
|
|
|
|
enabled for the build, but it helps me win the feature-count
|
|
|
|
competitions ooh yeah.
|
|
|
|
|
gst/avi/gstavidemux.c: Use scaling code for added precission and more correct stop position in case scale==0.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_src_convert),
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
(gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init),
(gst_avi_demux_parse_avih), (gst_avi_demux_parse_superindex),
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
(gst_avi_demux_stream_header), (gst_avi_demux_change_state):
Use scaling code for added precission and more correct stop
position in case scale==0.
2006-02-19 16:02:25 +00:00
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2006-02-19 Wim Taymans <wim@fluendo.com>
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|
|
|
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|
* gst/avi/gstavidemux.c: (gst_avi_demux_src_convert),
|
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|
|
(gst_avi_demux_handle_src_query), (gst_avi_demux_handle_src_event),
|
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|
|
(gst_avi_demux_parse_file_header), (gst_avi_demux_stream_init),
|
|
|
|
(gst_avi_demux_parse_avih), (gst_avi_demux_parse_superindex),
|
|
|
|
(gst_avi_demux_parse_subindex), (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_stream_header), (gst_avi_demux_change_state):
|
|
|
|
Use scaling code for added precission and more correct stop
|
|
|
|
position in case scale==0.
|
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|
|
|
2006-02-19 12:09:19 +00:00
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2006-02-19 Wim Taymans <wim@fluendo.com>
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|
|
|
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|
* gst/flx/gstflxdec.c: (gst_flxdec_src_query_handler),
|
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|
|
(gst_flxdec_chain):
|
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|
|
* gst/flx/gstflxdec.h:
|
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|
Implement DURATION query.
|
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|
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|
2006-02-19 11:57:58 +00:00
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|
2006-02-19 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/flx/flx_color.h:
|
|
|
|
* gst/flx/flx_fmt.h:
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_init),
|
|
|
|
(gst_flxdec_src_query_handler), (flx_decode_color),
|
|
|
|
(gst_flxdec_chain):
|
|
|
|
* gst/flx/gstflxdec.h:
|
|
|
|
Set MALLOCDATA for the temp buffers so we don't leak.
|
|
|
|
Some debug cleanups.
|
|
|
|
Consume all data in the adapter before leaving the chain
|
|
|
|
function. Fixes #330678.
|
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|
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|
2006-02-18 20:48:09 +00:00
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|
2006-02-18 Jan Schmidt <thaytan@mad.scientist.com>
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|
|
|
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|
* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3v2_genre_fields_to_taglist):
|
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|
|
Handle 0 data size in otherwise valid frames.
|
|
|
|
Handle numeric strings in 2.4.0 even when not in parentheses
|
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|
|
|
2006-02-18 17:20:48 +00:00
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|
2006-02-18 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
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|
* gst/matroska/matroska-demux.c:
|
|
|
|
(gst_matroska_demux_subtitle_caps),
|
|
|
|
(gst_matroska_demux_plugin_init):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Recognise SSA/ASS and USF subtitle formats and
|
|
|
|
set proper caps when they are found.
|
|
|
|
|
2006-02-17 17:54:05 +00:00
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|
2006-02-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
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|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_decode_direct),
|
|
|
|
(gst_jpeg_dec_chain):
|
|
|
|
Fix invalid memory access for some odd-sized images
|
|
|
|
(see image contained in quicktime stream in #327083);
|
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|
|
use g_malloc() instead of g_alloca().
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|
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|
2006-02-17 15:37:38 +00:00
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|
|
2006-02-17 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
|
|
|
|
Patch from Sebastien Cote, fixes #319884
|
|
|
|
|
2006-02-17 11:19:34 +00:00
|
|
|
2006-02-17 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/cdio/gstcdio.c: (plugin_init):
|
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|
|
Init debug category (#331253).
|
|
|
|
|
2006-02-16 20:30:13 +00:00
|
|
|
2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_stream_headers):
|
|
|
|
Pass extra_data to gst_riff_create_audio_caps(), so that
|
|
|
|
WAVEFORMATEX stuff works. Post audio codec name and post
|
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|
it as taglist on the bus. Allow up to 8 channesl for raw
|
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|
PCM in the source pad template caps.
|
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|
gst/multipart/multipartdemux.c: Applied #318663. Gives quite a few false positives in autoscan mode, but it's better ...
Original commit message from CVS:
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_demux_finalize), (gst_multipart_find_pad_by_mime),
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
(gst_multipart_set_property), (gst_multipart_get_property):
Applied #318663. Gives quite a few false positives in
autoscan mode, but it's better than nothing. Not closing yet.
2006-02-16 16:53:52 +00:00
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2006-02-16 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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|
* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
|
|
|
|
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
|
|
|
|
(gst_multipart_demux_finalize), (gst_multipart_find_pad_by_mime),
|
|
|
|
(gst_multipart_demux_chain), (gst_multipart_demux_change_state),
|
|
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(gst_multipart_set_property), (gst_multipart_get_property):
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Applied #318663. Gives quite a few false positives in
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autoscan mode, but it's better than nothing. Not closing yet.
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2006-02-16 14:13:48 +00:00
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2006-02-16 Wim Taymans <wim@fluendo.com>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/inspect/plugin-udp.xml:
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
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(gst_udpsrc_start):
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Update documentation.
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Fix args.
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2006-02-16 10:58:18 +00:00
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2006-02-16 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3tags.c: (id3demux_id3v2_frames_to_tag_list):
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ID3 2.3.0 used synch-safe integers for the tag size, but not for the
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frame size. (Fixes #331368)
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2006-02-16 10:42:25 +00:00
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2006-02-16 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/README:
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Updated README.
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
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(gst_rtspsrc_class_init), (gst_rtspsrc_set_property),
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(gst_rtspsrc_get_property), (gst_rtspsrc_stream_setup_rtp):
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* gst/rtsp/gstrtspsrc.h:
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Make sure the RTP port is an even port an try to allocate
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another if not.
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Added retry property to control max retries for port allocation.
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Make sure RTCP port is RTP port+1.
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Cleanup when port allocation fails.
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Fixes #319183.
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2006-02-16 09:17:58 +00:00
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2006-02-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/alpha/gstalpha.c: (gst_alpha_change_state):
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Don't ignore return value of the parent class's state
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change function (#331385, patch by: Wouter Paesen).
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Add HAL sound device wrapper plugins. Closes #329106
Original commit message from CVS:
* configure.ac:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* docs/plugins/gst-plugins-good-plugins.hierarchy:
* ext/Makefile.am:
* ext/hal/Makefile.am:
* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init),
(gst_hal_audio_sink_class_init), (gst_hal_audio_sink_reset),
(gst_hal_audio_sink_init), (gst_hal_audio_sink_dispose),
(do_toggle_element), (gst_hal_audio_sink_set_property),
(gst_hal_audio_sink_get_property),
(gst_hal_audio_sink_change_state):
* ext/hal/gsthalaudiosink.h:
* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init),
(gst_hal_audio_src_class_init), (gst_hal_audio_src_reset),
(gst_hal_audio_src_init), (gst_hal_audio_src_dispose),
(do_toggle_element), (gst_hal_audio_src_set_property),
(gst_hal_audio_src_get_property), (gst_hal_audio_src_change_state):
* ext/hal/gsthalaudiosrc.h:
* ext/hal/gsthalelements.c: (plugin_init):
* ext/hal/gsthalelements.h:
* ext/hal/hal.c: (gst_hal_get_string),
(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
(gst_hal_get_audio_src):
* ext/hal/hal.h:
Add HAL sound device wrapper plugins. Closes #329106
2006-02-15 12:17:28 +00:00
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2006-02-15 Wim Taymans <wim@fluendo.com>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* ext/Makefile.am:
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* ext/hal/Makefile.am:
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* ext/hal/gsthalaudiosink.c: (gst_hal_audio_sink_base_init),
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(gst_hal_audio_sink_class_init), (gst_hal_audio_sink_reset),
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(gst_hal_audio_sink_init), (gst_hal_audio_sink_dispose),
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(do_toggle_element), (gst_hal_audio_sink_set_property),
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(gst_hal_audio_sink_get_property),
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(gst_hal_audio_sink_change_state):
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* ext/hal/gsthalaudiosink.h:
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* ext/hal/gsthalaudiosrc.c: (gst_hal_audio_src_base_init),
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(gst_hal_audio_src_class_init), (gst_hal_audio_src_reset),
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(gst_hal_audio_src_init), (gst_hal_audio_src_dispose),
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(do_toggle_element), (gst_hal_audio_src_set_property),
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(gst_hal_audio_src_get_property), (gst_hal_audio_src_change_state):
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* ext/hal/gsthalaudiosrc.h:
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* ext/hal/gsthalelements.c: (plugin_init):
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* ext/hal/gsthalelements.h:
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* ext/hal/hal.c: (gst_hal_get_string),
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(gst_hal_render_bin_from_udi), (gst_hal_get_audio_sink),
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(gst_hal_get_audio_src):
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* ext/hal/hal.h:
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Add HAL sound device wrapper plugins. Closes #329106
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2006-02-15 10:15:47 +00:00
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2006-02-15 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event):
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Add comment in a fultile attempt to stop the copy-and-paste
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paradigm leading to duplication of bad code.
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* gst/rtsp/rtsptransport.c: (rtsp_transport_parse):
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Mime parameters have to be checked case insensitive
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2006-02-14 23:23:08 +00:00
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2006-02-14 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
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Advance stream time for lagging subtitle streams by sending
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newsegment events with the update flag set.
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2006-02-14 16:58:30 +00:00
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2006-02-14 Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header):
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There can be bogus data before the hdrl LIST tag in the RIFF header.
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It's hard to say if it's not respecting the AVI specifications or not,
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but since Google Video is producing AVIs like that and the other player
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don't seem to complain, I guess we should do the same.
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2006-02-13 12:00:51 +00:00
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2006-02-13 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3v2frames.c: (parse_insert_string_field),
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(parse_split_strings):
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Add more validation to ensure that a char encoding conversion
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produced a valid UTF-8 string.
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2006-02-13 10:43:15 +00:00
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2006-02-13 Mark Nauwelaerts <manauw@skynet.be>
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Reviewed by: Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
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Properly handle end of segment. Closes #330885.
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2006-02-13 10:36:23 +00:00
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2006-02-13 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gpay.h:
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For got to commit this one.
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2006-02-12 18:59:36 +00:00
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2006-02-12 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_class_init),
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(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_parse_audio_config),
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(gst_rtp_mp4g_pay_parse_video_config), (gst_rtp_mp4g_pay_new_caps),
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(gst_rtp_mp4g_pay_setcaps), (gst_rtp_mp4g_pay_flush):
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* gst/rtp/gstrtpmp4gpay.h:
|
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Make more things work.
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Handle ACC config strings.
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2006-02-12 13:10:20 +00:00
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2006-02-12 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
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set timestamps if no incoming timestamps set
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2006-02-11 13:54:26 +00:00
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2006-02-11 Tim-Philipp Müller <tim at centricular dot net>
|
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* gst/apetag/gsttagdemux.c: (gst_tag_demux_get_upstream_size),
|
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(gst_tag_demux_do_typefind):
|
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|
|
... and fix the very same leaks in GstTagDemux.
|
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|
2006-02-11 13:35:13 +00:00
|
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2006-02-11 Jan Schmidt <thaytan@mad.scientist.com>
|
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|
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* gst/id3demux/gstid3demux.c: (id3demux_get_upstream_size),
|
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(gst_id3demux_do_typefind):
|
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|
|
Fix a couple of mem leaks. (Patch by Jonathan Matthew
|
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|
|
<jonathan at kaolin dot wh9 dot net>)
|
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2006-02-10 17:37:39 +00:00
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2006-02-10 Wim Taymans <wim@fluendo.com>
|
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* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_setcaps):
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First set options, then set caps or else the baseclass
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|
will not know about the options, duh.
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2006-02-10 17:16:55 +00:00
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2006-02-10 Wim Taymans <wim@fluendo.com>
|
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* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_class_init),
|
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|
(gst_rtp_mp4v_pay_setcaps):
|
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|
Don't waste time looking for a config string if we have codec_info
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on the incomming caps.
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gst/rtp/README: Say something about case-sensitivity of caps vs mime-attributes.
Original commit message from CVS:
* gst/rtp/README:
Say something about case-sensitivity of caps vs mime-attributes.
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
(gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_get_type),
(gst_rtp_mp4g_pay_base_init), (gst_rtp_mp4g_pay_class_init),
(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_finalize),
(gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps),
(gst_rtp_mp4g_pay_flush), (gst_rtp_mp4g_pay_handle_buffer),
(gst_rtp_mp4g_pay_set_property), (gst_rtp_mp4g_pay_get_property),
(gst_rtp_mp4g_pay_plugin_init):
* gst/rtp/gstrtpmp4gpay.h:
Added beginnings of mpeg4-generic payloader (RFC 3640)
2006-02-10 16:40:58 +00:00
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2006-02-10 Wim Taymans <wim@fluendo.com>
|
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|
|
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* gst/rtp/README:
|
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|
Say something about case-sensitivity of caps vs mime-attributes.
|
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|
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* gst/rtp/Makefile.am:
|
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|
* gst/rtp/gstrtp.c: (plugin_init):
|
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|
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
|
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(gst_rtp_amr_pay_handle_buffer):
|
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* gst/rtp/gstrtpmp4gpay.c: (gst_rtp_mp4g_pay_get_type),
|
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(gst_rtp_mp4g_pay_base_init), (gst_rtp_mp4g_pay_class_init),
|
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|
|
(gst_rtp_mp4g_pay_init), (gst_rtp_mp4g_pay_finalize),
|
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|
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(gst_rtp_mp4g_pay_new_caps), (gst_rtp_mp4g_pay_setcaps),
|
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(gst_rtp_mp4g_pay_flush), (gst_rtp_mp4g_pay_handle_buffer),
|
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(gst_rtp_mp4g_pay_set_property), (gst_rtp_mp4g_pay_get_property),
|
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(gst_rtp_mp4g_pay_plugin_init):
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* gst/rtp/gstrtpmp4gpay.h:
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|
Added beginnings of mpeg4-generic payloader (RFC 3640)
|
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|
2006-02-09 14:20:14 +00:00
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|
2006-02-09 Wim Taymans <wim@fluendo.com>
|
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|
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* gst/rtsp/Makefile.am:
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* gst/rtsp/gstrtpdec.c: (gst_rtpdec_get_type),
|
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|
(gst_rtpdec_class_init), (gst_rtpdec_init), (gst_rtpdec_getcaps),
|
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|
(gst_rtpdec_chain_rtp), (gst_rtpdec_chain_rtcp),
|
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(gst_rtpdec_set_property), (gst_rtpdec_get_property),
|
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(gst_rtpdec_change_state):
|
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* gst/rtsp/gstrtpdec.h:
|
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|
* gst/rtsp/gstrtsp.c: (plugin_init):
|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_transport):
|
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|
* gst/rtsp/rtspconnection.c: (read_body),
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(rtsp_connection_receive):
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* gst/rtsp/rtspmessage.c: (rtsp_message_dump):
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Resurected rtpdec to make rtspsrc happy again.
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Skip attributes from the session id.
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Don't crash when dumping a message with an empty body.
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2006-02-09 14:14:07 +00:00
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2006-02-09 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpamrdepay.c: (gst_rtp_amr_depay_chain):
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Added more meaningfull warnings when something goes wrong.
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Clear F bit on outgoing AMR packets.
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* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_class_init),
|
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(gst_rtp_amr_pay_handle_buffer):
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|
Added debugging category
|
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|
Support payloading of multiple AMR frames.
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* gst/rtp/gstrtpmp4vpay.c: (gst_rtp_mp4v_pay_depay_data):
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|
Added some debugging.
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2006-02-09 11:25:42 +00:00
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2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
|
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|
Back to CVS
|
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2006-02-09 11:22:38 +00:00
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=== release 0.10.2 ===
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|
2006-02-09 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
|
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releasing 0.10.2, "Papa was a rolling stone"
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2006-02-08 17:16:46 +00:00
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2006-02-08 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
|
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Bump core and plugins-base requirement to 0.10.2.2
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for API additions (and 1 migration of gst_bin_find_unconnected_pad)
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2006-02-08 17:12:40 +00:00
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2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
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|
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* ext/flac/gstflac.c: (plugin_init):
|
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* ext/speex/gstspeex.c: (plugin_init):
|
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Register musicbrainz tags.
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2006-02-07 16:23:33 +00:00
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2006-02-07 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gconf/gconf.h:
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Remove declaration of function that no longer exists.
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2006-02-07 13:39:08 +00:00
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2006-02-07 Zaheer Abbas Merali <zaheerabbas at merali dot org>
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* ext/shout2/gstshout2.c: (gst_shout2send_render),
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(gst_shout2send_setcaps), (gst_shout2send_change_state):
|
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Make shout2 work for non ogg streams
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|
gst/udp/gstmultiudpsink.*: Updated docs.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_render), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add),
(gst_multiudpsink_remove), (gst_multiudpsink_clear),
(gst_multiudpsink_get_stats), (gst_multiudpsink_change_state):
* gst/udp/gstmultiudpsink.h:
Updated docs.
Added properties bytes-served, bytes_to_serve.
Post proper error messages,
Emit client added signal too.
2006-02-06 17:26:43 +00:00
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2006-02-06 Wim Taymans <wim@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
|
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(gst_multiudpsink_render), (gst_multiudpsink_get_property),
|
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|
(gst_multiudpsink_init_send), (gst_multiudpsink_add),
|
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|
|
(gst_multiudpsink_remove), (gst_multiudpsink_clear),
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|
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(gst_multiudpsink_get_stats), (gst_multiudpsink_change_state):
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* gst/udp/gstmultiudpsink.h:
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Updated docs.
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Added properties bytes-served, bytes_to_serve.
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Post proper error messages,
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Emit client added signal too.
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2006-02-06 15:31:16 +00:00
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2006-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
|
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(gst_multiudpsink_get_stats):
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adding docs for multiudpsink
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2006-02-06 15:28:56 +00:00
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2006-02-06 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/level/gstlevel.c: (gst_level_transform_ip):
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peak below decay is not necessarily an error, so don't ERROR log
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2006-02-06 14:25:34 +00:00
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2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/ebml-write.c: (gst_ebml_write_reset),
|
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(gst_ebml_write_flush_cache), (gst_ebml_write_element_push),
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(gst_ebml_write_seek):
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* gst/matroska/ebml-write.h:
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|
Make sure we send a newsegment event in BYTES format
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before sending buffers (#328531).
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2006-02-06 12:18:45 +00:00
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2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_src_query),
|
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(gst_dvdemux_sink_query):
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* ext/flac/gstflacdec.c: (gst_flac_dec_src_query):
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* ext/speex/gstspeexdec.c: (speex_get_query_types),
|
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(speex_dec_src_query):
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* ext/speex/gstspeexenc.c: (gst_speexenc_src_query),
|
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(gst_speexenc_sink_query):
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
|
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* gst/matroska/matroska-demux.c:
|
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(gst_matroska_demux_get_src_query_types),
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(gst_matroska_demux_handle_src_query):
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_get_query_types),
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(gst_wavparse_pad_query):
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|
Pass unhandled queries upstream instead of just dropping
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them (#326446). Update query type arrays here and there.
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2006-02-06 11:57:52 +00:00
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2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
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* tests/check/elements/matroskamux.c: (setup_src_pad):
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Collectpads in core got changed and now also holds a
|
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|
reference to any pad that is part of it. Fix refcount
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|
checks in test case accordingly.
|
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|
2006-02-06 11:41:43 +00:00
|
|
|
2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
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* gst/apetag/gstapedemux.h:
|
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|
Fix include, for now GstTagDemux is in the apetag dir.
|
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|
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|
2006-02-06 11:34:23 +00:00
|
|
|
2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-cdio.xml:
|
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|
|
Add cdio plugin to docs.
|
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|
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* ext/cdio/gstcdiocddasrc.c:
|
|
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|
Add gtk-doc blurb.
|
|
|
|
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|
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|
* ext/cdio/gstcdio.c:
|
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|
|
The plugin is called 'cdio' not 'cddio'.
|
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|
2006-02-06 10:56:07 +00:00
|
|
|
2006-02-06 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
|
|
|
* docs/plugins/inspect/plugin-apetag.xml:
|
|
|
|
* gst/apetag/Makefile.am:
|
|
|
|
* gst/apetag/gstapedemux.c:
|
|
|
|
* gst/apetag/gstapedemux.h:
|
|
|
|
* gst/apetag/gsttagdemux.c:
|
|
|
|
* gst/apetag/gsttagdemux.h:
|
|
|
|
Add APE tag demuxer (#325649).
|
|
|
|
|
ext/gconf/: Ignore changing the GConf key to "". Ignore GConf key updates that don't actually change the string.
Original commit message from CVS:
* ext/gconf/gconf.c: (gst_gconf_get_default_audio_sink),
(gst_gconf_get_default_video_sink),
(gst_gconf_get_default_audio_src),
(gst_gconf_get_default_video_src):
* ext/gconf/gconf.h:
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
(gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose),
(do_toggle_element):
* ext/gconf/gstgconfaudiosink.h:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
(gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
(do_toggle_element):
* ext/gconf/gstgconfaudiosrc.h:
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
(gst_gconf_video_sink_init), (gst_gconf_video_sink_dispose),
(do_toggle_element):
* ext/gconf/gstgconfvideosink.h:
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
(gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
(do_toggle_element):
* ext/gconf/gstgconfvideosrc.h:
Ignore changing the GConf key to "". Ignore GConf key updates
that don't actually change the string.
For now, ignore the GConf key when the state is > READY, as
it breaks streaming. Sometime it will be nice to bring the
new sink online even mid-stream, by sending NEWSEGMENT info
and possibly prerolling.
(Fixes #326736)
2006-02-05 22:22:56 +00:00
|
|
|
2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/gconf/gconf.c: (gst_gconf_get_default_audio_sink),
|
|
|
|
(gst_gconf_get_default_video_sink),
|
|
|
|
(gst_gconf_get_default_audio_src),
|
|
|
|
(gst_gconf_get_default_video_src):
|
|
|
|
* ext/gconf/gconf.h:
|
|
|
|
* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset),
|
|
|
|
(gst_gconf_audio_sink_init), (gst_gconf_audio_sink_dispose),
|
|
|
|
(do_toggle_element):
|
|
|
|
* ext/gconf/gstgconfaudiosink.h:
|
|
|
|
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_reset),
|
|
|
|
(gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
|
|
|
|
(do_toggle_element):
|
|
|
|
* ext/gconf/gstgconfaudiosrc.h:
|
|
|
|
* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset),
|
|
|
|
(gst_gconf_video_sink_init), (gst_gconf_video_sink_dispose),
|
|
|
|
(do_toggle_element):
|
|
|
|
* ext/gconf/gstgconfvideosink.h:
|
|
|
|
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_reset),
|
|
|
|
(gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
|
|
|
|
(do_toggle_element):
|
|
|
|
* ext/gconf/gstgconfvideosrc.h:
|
|
|
|
Ignore changing the GConf key to "". Ignore GConf key updates
|
|
|
|
that don't actually change the string.
|
|
|
|
For now, ignore the GConf key when the state is > READY, as
|
|
|
|
it breaks streaming. Sometime it will be nice to bring the
|
|
|
|
new sink online even mid-stream, by sending NEWSEGMENT info
|
|
|
|
and possibly prerolling.
|
|
|
|
(Fixes #326736)
|
|
|
|
|
gst/goom/: Make goom reentrant by moving all important static variables into instance structures.
Original commit message from CVS:
* gst/goom/filters.c: (zoomFilterNew), (calculatePXandPY),
(setPixelRGB), (setPixelRGB_), (getPixelRGB), (getPixelRGB_),
(zoomFilterSetResolution), (zoomFilterDestroy),
(zoomFilterFastRGB), (pointFilter):
* gst/goom/filters.h:
* gst/goom/goom_core.c: (goom_init), (goom_set_resolution),
(goom_update), (goom_close):
* gst/goom/goom_core.h:
* gst/goom/goom_tools.h:
* gst/goom/graphic.c:
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
(gst_goom_dispose), (gst_goom_src_setcaps), (gst_goom_chain):
* gst/goom/gstgoom.h:
* gst/goom/lines.c: (goom_lines):
* gst/goom/lines.h:
Make goom reentrant by moving all important static variables
into instance structures.
(Fixes #329181)
2006-02-05 20:43:49 +00:00
|
|
|
2006-02-05 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/goom/filters.c: (zoomFilterNew), (calculatePXandPY),
|
|
|
|
(setPixelRGB), (setPixelRGB_), (getPixelRGB), (getPixelRGB_),
|
|
|
|
(zoomFilterSetResolution), (zoomFilterDestroy),
|
|
|
|
(zoomFilterFastRGB), (pointFilter):
|
|
|
|
* gst/goom/filters.h:
|
|
|
|
* gst/goom/goom_core.c: (goom_init), (goom_set_resolution),
|
|
|
|
(goom_update), (goom_close):
|
|
|
|
* gst/goom/goom_core.h:
|
|
|
|
* gst/goom/goom_tools.h:
|
|
|
|
* gst/goom/graphic.c:
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_class_init), (gst_goom_init),
|
|
|
|
(gst_goom_dispose), (gst_goom_src_setcaps), (gst_goom_chain):
|
|
|
|
* gst/goom/gstgoom.h:
|
|
|
|
* gst/goom/lines.c: (goom_lines):
|
|
|
|
* gst/goom/lines.h:
|
|
|
|
Make goom reentrant by moving all important static variables
|
|
|
|
into instance structures.
|
|
|
|
(Fixes #329181)
|
|
|
|
|
2006-02-04 15:41:43 +00:00
|
|
|
2006-02-04 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_all_source_pads_unlinked),
|
|
|
|
(gst_avi_demux_process_next_entry):
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
Third attempt, use gst_pad_is_linked() this time.
|
|
|
|
|
2006-02-04 13:30:12 +00:00
|
|
|
2006-02-04 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(parse_split_strings):
|
|
|
|
Adjust for data length indicators when parsing (Fixes #329810)
|
|
|
|
Fix stupid bug parsing UTF-8 tag text.
|
|
|
|
Output tag strings with multiple fields as multiple tags, so the
|
|
|
|
app gets all the data.
|
|
|
|
|
2006-02-03 20:05:20 +00:00
|
|
|
2006-02-03 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* ext/flac/gstflacenc.c:
|
|
|
|
Fixed a bug add in last commit, where no event is send. Thanks Tim to
|
|
|
|
show me.
|
|
|
|
|
2006-02-03 18:07:35 +00:00
|
|
|
2006-02-03 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* ext/flac/gstflacenc.c:
|
|
|
|
* gst/matroska/ebml-read.c:
|
|
|
|
Just make it compile with --disable-gst-debug.
|
|
|
|
|
2006-02-03 13:06:24 +00:00
|
|
|
2006-02-03 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (parse_text_identification_frame),
|
|
|
|
(id3v2_tag_to_taglist), (id3v2_genre_string_to_taglist),
|
|
|
|
(id3v2_genre_fields_to_taglist):
|
|
|
|
Never output a tag with a null contents string.
|
|
|
|
|
2006-02-02 21:00:16 +00:00
|
|
|
2006-02-02 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked):
|
|
|
|
Only pause if all pads are unlinked AND we've tried to send data
|
|
|
|
on all of them at least once.
|
|
|
|
|
2006-02-02 12:29:24 +00:00
|
|
|
2006-02-02 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_all_source_pads_unlinked),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_loop):
|
|
|
|
Make loop function/task pause itself when all source pads are
|
|
|
|
unlinked.
|
|
|
|
|
2006-02-02 10:47:15 +00:00
|
|
|
2006-02-02 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* ext/gconf/gconf.c: (gst_gconf_render_bin_from_key):
|
|
|
|
Use new functions from core to render a bin from a
|
|
|
|
string. Fixes build. Up requirements to core CVS.
|
|
|
|
|
2006-02-01 11:01:04 +00:00
|
|
|
2006-02-01 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/auparse/gstauparse.c: (gst_au_parse_chain):
|
|
|
|
Don't push buffers into the adapter that we are going to
|
|
|
|
push downstream again without framing anyway. Also, the
|
|
|
|
adaptor takes ownership of buffers put into it (fixes
|
|
|
|
auparse pushing invalid buffers for .au files with
|
|
|
|
ADPCM contents). Finally, set caps on all outgoing buffers.
|
|
|
|
|
2006-01-30 23:13:05 +00:00
|
|
|
2006-01-31 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_chain),
|
|
|
|
(gst_id3demux_read_id3v1), (gst_id3demux_sink_activate),
|
|
|
|
(gst_id3demux_send_tag_event):
|
|
|
|
* gst/id3demux/id3tags.c: (id3demux_read_id3v1_tag):
|
|
|
|
Someone should kick my butt. Remove ID3v1 tags from the end of the
|
|
|
|
file.
|
|
|
|
|
|
|
|
Improve error messages. Send the TAG message as soon as we complete
|
|
|
|
typefinding, instead of waiting until we send the first buffer.
|
|
|
|
Downstream tag event is still sent before the first buffer.
|
|
|
|
|
2006-01-27 12:17:56 +00:00
|
|
|
2006-01-27 Jan Gerber <j@bootlab.org>
|
|
|
|
|
|
|
|
Reviewed by: Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c (gst_dvdec_change_state):
|
|
|
|
* ext/dv/gstdvdemux.c (gst_dvdemux_change_state):
|
|
|
|
Call dv_set_error_log (dv_decoder_t *, NULL); after dv_decoder_new
|
|
|
|
to not have warings flooding stderr. this is the suggested way
|
|
|
|
also used in dvgrab and kino. (#328336)
|
|
|
|
|
2006-01-27 01:43:07 +00:00
|
|
|
2006-01-27 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* sys/oss/gstosssink.c: (gst_oss_sink_class_init),
|
|
|
|
(gst_oss_sink_init), (gst_oss_sink_finalise):
|
|
|
|
Free the device name string when finalised.
|
|
|
|
|
2006-01-25 22:05:28 +00:00
|
|
|
2006-01-25 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_remove_srcpad):
|
|
|
|
Don't put function calls in g_return_if_fail() statements,
|
|
|
|
or they'll be replaced with NOOPs if someone compiles with
|
|
|
|
G_DISABLE_CHECKS defined.
|
|
|
|
|
2006-01-25 18:23:05 +00:00
|
|
|
2006-01-25 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
|
|
|
|
Never trust ANY information encoded in a media file, especially
|
|
|
|
when it's giving you sizes. (Fixes #328452)
|
|
|
|
|
2006-01-24 18:03:46 +00:00
|
|
|
2006-01-24 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpg711pay.c:
|
|
|
|
Patch written by Kai Vehmanen <kai.vehmanen@nokia.com> applied. See
|
|
|
|
bug #325148.
|
|
|
|
|
2006-01-23 18:39:31 +00:00
|
|
|
2006-01-23 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps),
|
|
|
|
(gst_matroska_demux_audio_caps), (gst_matroska_demux_plugin_init):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
Added recognition of Real Audio and Video streams in matroska demuxer.
|
|
|
|
|
2006-01-23 14:32:47 +00:00
|
|
|
2006-01-23 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3v2_tag_to_taglist):
|
|
|
|
Remove errant break statement, and fix compilation with
|
|
|
|
older GCC.
|
|
|
|
|
2006-01-23 11:06:34 +00:00
|
|
|
2006-01-23 Brian Cameron <brian dot cameron at sun dot com>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_init):
|
|
|
|
Export functions that are needed in other parts of the code,
|
|
|
|
makes the mixer actually work; adjust magic minimum buffer-time
|
|
|
|
value from 3ms to 5ms to work around stuttering during mp3
|
|
|
|
playback (#327765).
|
|
|
|
|
2006-01-23 10:44:03 +00:00
|
|
|
2006-01-23 Michal Benes <michal dot benes at xeris dot cz>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_best_pad),
|
|
|
|
(gst_matroska_mux_write_data), (gst_matroska_mux_collected):
|
|
|
|
Fix possible deadlock in matroska muxer (#327825).
|
|
|
|
|
2006-01-23 09:59:03 +00:00
|
|
|
2006-01-23 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_chain):
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_invert):
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c: (gst_rtp_mp4v_depay_setcaps):
|
|
|
|
* gst/rtsp/sdpmessage.h:
|
|
|
|
* gst/udp/gstdynudpsink.c: (gst_dynudpsink_render):
|
|
|
|
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_get_stats):
|
|
|
|
C89 fixes: declare variables at the beginning of a block and
|
|
|
|
make gcc-2.9x happy (#328264; patch by: Jens Granseuer
|
|
|
|
<jensgr at gmx dot net>).
|
|
|
|
|
2006-01-23 09:22:17 +00:00
|
|
|
2006-01-23 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
|
|
|
|
* gst/id3demux/id3tags.h:
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
|
|
|
(parse_comment_frame), (parse_text_identification_frame),
|
|
|
|
(id3v2_tag_to_taglist), (id3v2_are_digits),
|
|
|
|
(id3v2_genre_string_to_taglist), (id3v2_genre_fields_to_taglist),
|
|
|
|
(parse_split_strings), (free_tag_strings):
|
|
|
|
Rewrite parsing of text tags to handle multiple NULL terminated
|
|
|
|
strings. Parse numeric genre strings and ID3v2 type
|
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|
|
"(3)(6)Alternative" style genre strings.
|
|
|
|
Parse dates that are only YYYY or YYYY-mm format.
|
2006-01-25 20:33:05 +00:00
|
|
|
(Fixes #328241 and #322154)
|
2006-01-23 09:22:17 +00:00
|
|
|
|
2006-01-20 15:06:28 +00:00
|
|
|
2006-01-20 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/videobox/gstvideobox.c: (gst_video_box_class_init):
|
|
|
|
Don't forget to initialize liboil, otherwise our oil functions
|
|
|
|
will crash (fixes #327871; patch by: Christoph Burghardt
|
|
|
|
<hawkes at web dot de>).
|
|
|
|
|
2006-01-19 21:00:50 +00:00
|
|
|
2006-01-19 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
Use plain AS_LIBTOOL_TAGS instead of AS_LIBTOOL_TAGS([CXX]), just
|
2006-01-19 21:46:32 +00:00
|
|
|
like in the core and gst-plugins-base. Fixes #324367 and #326683;
|
|
|
|
patch by: Brian Cameron <brian dot cameron at sun dot com>
|
2006-01-19 21:00:50 +00:00
|
|
|
|
2006-01-19 00:10:51 +00:00
|
|
|
2006-01-19 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/cdio/gstcdio.c: (gst_cdio_add_cdtext_field), (plugin_init):
|
|
|
|
* ext/cdio/gstcdio.h:
|
|
|
|
* ext/cdio/gstcdiocddasrc.c: (gst_cdio_cdda_src_get_cdtext):
|
|
|
|
Fix build for libcdio versions >= 76; give slightly lower rank
|
|
|
|
than cdparanoia.
|
|
|
|
|
2006-01-18 19:30:36 +00:00
|
|
|
2006-01-18 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* ext/Makefile.am:
|
|
|
|
* ext/cdio/Makefile.am:
|
|
|
|
* ext/cdio/gstcdio.c:
|
|
|
|
* ext/cdio/gstcdio.h:
|
|
|
|
* ext/cdio/gstcdiocddasrc.c:
|
|
|
|
* ext/cdio/gstcdiocddasrc.h:
|
|
|
|
Port libcdio cdda source, formerly known as cddasrc, now known as
|
|
|
|
cdiocddasrc (fixes #323327). Should also read CD-TEXT if available,
|
|
|
|
but that's not tested (fixes #317658).
|
|
|
|
|
2006-01-18 19:08:08 +00:00
|
|
|
2006-01-18 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_pad_convert):
|
|
|
|
Fix conversion from TIME to BYTES format (fixes #326864;
|
|
|
|
patch by: Tommi Myöhänen <ext-tommi dot myohanen at nokia dot com>)
|
|
|
|
|
2006-01-15 20:21:48 +00:00
|
|
|
2006-01-15 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame):
|
|
|
|
Fix compilation of id3demux when zlib is not present.
|
|
|
|
(Fixes #326602; patch by: Sergey Scobich)
|
|
|
|
|
2006-01-15 14:12:12 +00:00
|
|
|
2006-01-15 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/esd/Makefile.am:
|
|
|
|
Add $(ESD_CFLAGS), otherwise build will fail for folks
|
|
|
|
with libesd in a non-standard prefix (#327009).
|
|
|
|
|
2006-01-13 19:29:27 +00:00
|
|
|
2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
back to HEAD
|
|
|
|
|
2006-01-13 19:25:40 +00:00
|
|
|
=== release 0.10.1 ===
|
|
|
|
|
|
|
|
2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.10.1, "Li"
|
|
|
|
|
2006-01-13 18:37:13 +00:00
|
|
|
2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
patch by: Wim Taymans
|
|
|
|
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
|
|
|
|
fix memleak. Fixes #326618
|
|
|
|
|
2006-01-13 18:35:00 +00:00
|
|
|
2006-01-13 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
patch by: Mike Smith
|
|
|
|
|
|
|
|
* gst/level/gstlevel.c: (gst_level_message_new),
|
|
|
|
(gst_level_message_append_channel):
|
|
|
|
Fix memleak. Fixes #326612
|
|
|
|
|
2006-01-11 11:39:10 +00:00
|
|
|
2006-01-11 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
prereleasing
|
|
|
|
* po/af.po:
|
|
|
|
* po/az.po:
|
|
|
|
* po/cs.po:
|
|
|
|
* po/en_GB.po:
|
|
|
|
* po/hu.po:
|
|
|
|
* po/it.po:
|
|
|
|
* po/nb.po:
|
|
|
|
* po/nl.po:
|
|
|
|
* po/or.po:
|
|
|
|
* po/sq.po:
|
|
|
|
* po/sr.po:
|
|
|
|
* po/sv.po:
|
|
|
|
* po/uk.po:
|
|
|
|
* po/vi.po:
|
|
|
|
update translations
|
|
|
|
|
2006-01-10 12:38:59 +00:00
|
|
|
2006-01-10 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/level/gstlevel.c: (gst_level_class_init),
|
|
|
|
(gst_level_dispose):
|
|
|
|
Don't leak filter arrays.
|
|
|
|
|
2006-01-09 17:04:52 +00:00
|
|
|
2006-01-09 Brian Cameron <brian.cameron@sun.com>
|
|
|
|
|
|
|
|
reviewed by: Christian Schaller <uraeus@gnome.org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* gst-plugins-good.spec.in:
|
|
|
|
* sys/Makefile.am:
|
|
|
|
* sys/sunaudio/Makefile.am:
|
|
|
|
* sys/sunaudio/gstsunaudio.c: (plugin_init):
|
|
|
|
* sys/sunaudio/gstsunaudiomixer.c: (gst_sunaudiomixer_base_init),
|
2006-01-11 11:39:10 +00:00
|
|
|
(gst_sunaudiomixer_class_init), (gst_sunaudiomixer_init),
|
|
|
|
(gst_sunaudiomixer_change_state):
|
|
|
|
* sys/sunaudio/gstsunaudiomixer.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.c:
|
|
|
|
(gst_sunaudiomixer_ctrl_open), (gst_sunaudiomixer_ctrl_build_list),
|
|
|
|
(gst_sunaudiomixer_ctrl_new), (gst_sunaudiomixer_ctrl_free),
|
|
|
|
(gst_sunaudiomixer_ctrl_list_tracks),
|
|
|
|
(gst_sunaudiomixer_ctrl_get_volume),
|
|
|
|
(gst_sunaudiomixer_ctrl_set_volume),
|
|
|
|
(gst_sunaudiomixer_ctrl_set_mute),
|
|
|
|
(gst_sunaudiomixer_ctrl_set_record):
|
|
|
|
* sys/sunaudio/gstsunaudiomixerctrl.h:
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.c:
|
|
|
|
(gst_sunaudiomixer_track_class_init),
|
|
|
|
(gst_sunaudiomixer_track_init), (fill_labels),
|
|
|
|
(gst_sunaudiomixer_track_new):
|
|
|
|
* sys/sunaudio/gstsunaudiomixertrack.h:
|
|
|
|
* sys/sunaudio/gstsunaudiosink.c: (gst_sunaudiosink_get_type),
|
|
|
|
(gst_sunaudiosink_dispose), (gst_sunaudiosink_base_init),
|
|
|
|
(gst_sunaudiosink_set_property), (gst_sunaudiosink_get_property),
|
|
|
|
(gst_sunaudiosink_getcaps), (gst_sunaudiosink_open),
|
|
|
|
(gst_sunaudiosink_close), (gst_sunaudiosink_prepare),
|
|
|
|
(gst_sunaudiosink_unprepare), (gst_sunaudiosink_write),
|
|
|
|
(gst_sunaudiosink_delay), (gst_sunaudiosink_reset):
|
|
|
|
* sys/sunaudio/gstsunaudiosink.h:
|
|
|
|
|
|
|
|
Add SunAudio plugin - tested to make sure it doesn't break
|
|
|
|
the build under GNU/Linux.
|
2006-01-09 17:04:52 +00:00
|
|
|
|
2006-01-07 20:01:09 +00:00
|
|
|
2005-01-07 Philippe Khalaf <philippe.kalaf@collabora.co.uk>
|
|
|
|
|
|
|
|
* gst-plugins-good/gst/udp/gstdynudpsink.c:
|
|
|
|
* gst-plugins-good/gst/udp/gstudpsrc.c:
|
2006-01-11 11:39:10 +00:00
|
|
|
Allow udpsrc and dynudpsink to take a sockfd as a parameter. For udpsrc,
|
|
|
|
overrides the port or multicast parameters. Fixes bugs #323021.
|
2006-01-13 19:25:40 +00:00
|
|
|
API addition: adds GstUDPSrc::sockfd property
|
2006-01-07 20:01:09 +00:00
|
|
|
|
ext/gconf/: Add new gconfaudiosrc and gconfvideosrc elements (needed for gnome-sound-recorder).
Original commit message from CVS:
* ext/gconf/Makefile.am:
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_reset),
(gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
(do_toggle_element), (cb_toggle_element),
(gst_gconf_audio_src_change_state):
* ext/gconf/gstgconfaudiosrc.h:
* ext/gconf/gstgconfelements.c: (plugin_init):
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
(gst_gconf_video_src_class_init), (gst_gconf_video_src_reset),
(gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
(do_toggle_element), (cb_toggle_element),
(gst_gconf_video_src_change_state):
* ext/gconf/gstgconfvideosrc.h:
Add new gconfaudiosrc and gconfvideosrc elements
(needed for gnome-sound-recorder).
2006-01-06 16:28:30 +00:00
|
|
|
2006-01-06 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/gconf/Makefile.am:
|
|
|
|
* ext/gconf/gstgconfaudiosrc.c: (gst_gconf_audio_src_base_init),
|
|
|
|
(gst_gconf_audio_src_class_init), (gst_gconf_audio_src_reset),
|
|
|
|
(gst_gconf_audio_src_init), (gst_gconf_audio_src_dispose),
|
|
|
|
(do_toggle_element), (cb_toggle_element),
|
|
|
|
(gst_gconf_audio_src_change_state):
|
|
|
|
* ext/gconf/gstgconfaudiosrc.h:
|
|
|
|
* ext/gconf/gstgconfelements.c: (plugin_init):
|
|
|
|
* ext/gconf/gstgconfvideosrc.c: (gst_gconf_video_src_base_init),
|
|
|
|
(gst_gconf_video_src_class_init), (gst_gconf_video_src_reset),
|
|
|
|
(gst_gconf_video_src_init), (gst_gconf_video_src_dispose),
|
|
|
|
(do_toggle_element), (cb_toggle_element),
|
|
|
|
(gst_gconf_video_src_change_state):
|
|
|
|
* ext/gconf/gstgconfvideosrc.h:
|
|
|
|
Add new gconfaudiosrc and gconfvideosrc elements
|
|
|
|
(needed for gnome-sound-recorder).
|
|
|
|
|
2006-01-06 11:46:53 +00:00
|
|
|
2006-01-06 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/id3demux/gstid3demux.c: (gst_id3demux_add_srcpad):
|
|
|
|
Add gst_element_no_more_pads() for proper decodebin behaviour.
|
|
|
|
* gst/id3demux/id3v2frames.c: (parse_comment_frame),
|
|
|
|
(parse_text_identification_frame), (parse_split_strings):
|
|
|
|
Failure to decode some tags is not a GST_ERROR() but a
|
|
|
|
GST_WARNING()
|
|
|
|
When iterating over a chunk of text, check that we haven't gone too
|
|
|
|
far.
|
|
|
|
|
2006-01-05 17:03:45 +00:00
|
|
|
2006-01-05 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
|
|
|
|
(plugin_init):
|
|
|
|
call oil_init() when using liboil
|
|
|
|
|
2006-01-04 17:28:49 +00:00
|
|
|
2006-01-04 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/jpeg/gstsmokedec.c: (gst_smokedec_chain):
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
|
|
|
|
Fix leaks.
|
|
|
|
|
2006-01-02 19:38:32 +00:00
|
|
|
2006-01-02 Alessandro Decina <alessandro at nnva dot org>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flac_dec_write),
|
|
|
|
(gst_flac_dec_convert_src), (gst_flac_dec_src_query),
|
|
|
|
(gst_flac_dec_change_state):
|
|
|
|
Don't g_assert() where we should just return FALSE; remove
|
|
|
|
unnecessary g_assert(); initialize some fields properly in
|
|
|
|
state change function (fixes #325504). Also, use
|
|
|
|
GST_DEBUG_OBJECT in two more places.
|
|
|
|
|
2005-12-30 15:51:05 +00:00
|
|
|
2005-12-30 Stefan Kost <ensonic@users.sf.net>
|
|
|
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|
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* configure.ac:
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2006-01-11 11:39:10 +00:00
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also remove smoothwave's Makefile.am
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2005-12-30 15:51:05 +00:00
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* docs/plugins/Makefile.am:
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2006-01-11 11:39:10 +00:00
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fix plugin docs
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2005-12-30 15:51:05 +00:00
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2005-12-30 15:34:18 +00:00
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2005-12-30 Stefan Kost <ensonic@users.sf.net>
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* tests/examples/Makefile.am:
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2006-01-11 11:39:10 +00:00
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added missing Makefile.am
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2005-12-30 15:34:18 +00:00
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2005-12-30 15:28:44 +00:00
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2005-12-30 Stefan Kost <ensonic@users.sf.net>
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* configure.ac:
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* gst/level/Makefile.am:
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* gst/level/level-example.c:
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* tests/Makefile.am:
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* tests/examples/level/Makefile.am:
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* tests/examples/level/level-example.c: (message_handler), (main):
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2006-01-11 11:39:10 +00:00
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moved level-example to tests/examples/level-example
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2005-12-30 15:28:44 +00:00
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* tests/old/examples/level/demo.c: (main):
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* tests/old/examples/level/plot.c: (main):
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2006-01-11 11:39:10 +00:00
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some initial fixes
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2005-12-30 15:28:44 +00:00
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2005-12-29 16:36:19 +00:00
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2005-12-29 Michael Smith <msmith@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
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(gst_multiudpsink_remove), (gst_multiudpsink_get_stats):
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* gst/udp/gstmultiudpsink.h:
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Track packets sent per client in addition to bytes sent; provide
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this info through get-stats signal
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2005-12-29 11:26:12 +00:00
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2005-12-29 Tim-Philipp Müller <tim at centricular dot net>
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* gst/auparse/gstauparse.c: (gst_au_parse_dispose):
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Can't use gst_object_unref() on a GstAdapter (#325191).
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2005-12-28 18:55:32 +00:00
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2005-12-28 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/id3demux/id3tags.c: (id3demux_read_id3v2_tag):
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If a broken tag has 0 bytes payload, at least still skip
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the 10 byte header
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2005-12-22 15:00:41 +00:00
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2005-12-22 Philippe Khalaf <burger@speedy.org>
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* gst-plugins-good/gst/rtp/gstrtph263pdepay.h:
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* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
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* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.h:
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* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
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2006-01-11 11:39:10 +00:00
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Making these depayloaders (H263+ and mpeg4 video) inherit from
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RtpBaseDepayloaderClass. Fixes bugs #323922 and #323908.
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2005-12-22 15:00:41 +00:00
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2005-12-21 17:15:09 +00:00
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2005-12-21 Jan Schmidt <thaytan@mad.scientist.com>
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* docs/plugins/gst-plugins-good-plugins.args:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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Regenerate the plugin hiearchy.
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2005-12-21 15:24:59 +00:00
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2005-12-21 Jan Schmidt <thaytan@mad.scientist.com>
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
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* docs/plugins/gst-plugins-good-plugins.args:
|
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* gst/id3demux/gstid3demux.c: (gst_id3demux_get_type),
|
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(gst_id3demux_base_init), (gst_id3demux_class_init),
|
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(gst_id3demux_chain):
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* gst/id3demux/gstid3demux.h:
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Add documentation for id3demux.
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Don't fail if the first buffer is not at offset 0, just
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attempt to typefind and do pass through
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Rename the gst_type function from gst_gst_id3demux..
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2005-12-20 12:44:25 +00:00
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2005-12-20 Michael Smith <msmith@fluendo.com>
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_render),
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(gst_multiudpsink_add), (gst_multiudpsink_remove),
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(gst_multiudpsink_get_stats):
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* gst/udp/gstmultiudpsink.h:
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Collect statistics; return them from get_stats.
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2005-12-19 15:43:30 +00:00
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2005-12-19 Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_stream_scan):
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Stupid signedness issue...
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gst/avi/gstavidemux.c: Construct index for indexless files.
Original commit message from CVS:
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
(gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag),
(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
(gst_avi_demux_stream_header), (gst_avi_demux_loop):
Construct index for indexless files.
Make sure pad/buffers are correctly reset to NULL once we don't need
them anymore, else we get lovely segfaults/assertions.
* gst/wavparse/gstwavparse.c:
Yes, you can have 96KHz audio and wma in wav :(
2005-12-19 15:06:27 +00:00
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2005-12-19 Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
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(gst_avi_demux_handle_src_event), (gst_avi_demux_parse_stream),
|
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|
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(gst_avi_demux_parse_odml), (gst_avi_demux_peek_tag),
|
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(gst_avi_demux_next_data_buffer), (gst_avi_demux_stream_scan),
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(gst_avi_demux_stream_header), (gst_avi_demux_loop):
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Construct index for indexless files.
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Make sure pad/buffers are correctly reset to NULL once we don't need
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them anymore, else we get lovely segfaults/assertions.
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* gst/wavparse/gstwavparse.c:
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Yes, you can have 96KHz audio and wma in wav :(
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configure.ac: Check for optional dependency on zlib for id3demux
Original commit message from CVS:
* configure.ac:
Check for optional dependency on zlib for id3demux
* gst/id3demux/Makefile.am:
* gst/id3demux/gstid3demux.c: (gst_gst_id3demux_get_type),
(gst_id3demux_base_init), (gst_id3demux_class_init),
(gst_id3demux_reset), (gst_id3demux_init), (gst_id3demux_dispose),
(gst_id3demux_add_srcpad), (gst_id3demux_remove_srcpad),
(gst_id3demux_trim_buffer), (gst_id3demux_chain),
(gst_id3demux_set_property), (gst_id3demux_get_property),
(id3demux_get_upstream_size), (gst_id3demux_srcpad_event),
(gst_id3demux_read_id3v1), (gst_id3demux_read_id3v2),
(gst_id3demux_sink_activate), (gst_id3demux_src_activate_pull),
(gst_id3demux_src_checkgetrange), (gst_id3demux_read_range),
(gst_id3demux_src_getrange), (gst_id3demux_change_state),
(gst_id3demux_pad_query), (gst_id3demux_get_query_types),
(simple_find_peek), (simple_find_suggest),
(gst_id3demux_do_typefind), (gst_id3demux_send_tag_event),
(plugin_init):
* gst/id3demux/gstid3demux.h:
* gst/id3demux/id3tags.c: (read_synch_uint),
(id3demux_read_id3v1_tag), (id3demux_read_id3v2_tag),
(id3demux_id3v2_frame_hdr_size), (convert_fid_to_v240),
(id3demux_id3v2_frames_to_tag_list):
* gst/id3demux/id3tags.h:
* gst/id3demux/id3v2.4.0-frames.txt:
* gst/id3demux/id3v2.4.0-structure.txt:
* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
(parse_comment_frame), (parse_text_identification_frame),
(id3v2_tag_to_taglist), (parse_split_strings):
All new LGPL id3 demuxer. Can use zlib for compressed frames,
otherwise it discards them. Works on my test files.
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop):
Don't send EOS to a non-existing srcpad
The debug category can be static
2005-12-18 15:14:44 +00:00
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2005-12-18 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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Check for optional dependency on zlib for id3demux
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* gst/id3demux/Makefile.am:
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* gst/id3demux/gstid3demux.c: (gst_gst_id3demux_get_type),
|
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(gst_id3demux_base_init), (gst_id3demux_class_init),
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(gst_id3demux_reset), (gst_id3demux_init), (gst_id3demux_dispose),
|
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(gst_id3demux_add_srcpad), (gst_id3demux_remove_srcpad),
|
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(gst_id3demux_trim_buffer), (gst_id3demux_chain),
|
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(gst_id3demux_set_property), (gst_id3demux_get_property),
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(id3demux_get_upstream_size), (gst_id3demux_srcpad_event),
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(gst_id3demux_read_id3v1), (gst_id3demux_read_id3v2),
|
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(gst_id3demux_sink_activate), (gst_id3demux_src_activate_pull),
|
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(gst_id3demux_src_checkgetrange), (gst_id3demux_read_range),
|
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(gst_id3demux_src_getrange), (gst_id3demux_change_state),
|
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(gst_id3demux_pad_query), (gst_id3demux_get_query_types),
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(simple_find_peek), (simple_find_suggest),
|
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(gst_id3demux_do_typefind), (gst_id3demux_send_tag_event),
|
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(plugin_init):
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* gst/id3demux/gstid3demux.h:
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* gst/id3demux/id3tags.c: (read_synch_uint),
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(id3demux_read_id3v1_tag), (id3demux_read_id3v2_tag),
|
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(id3demux_id3v2_frame_hdr_size), (convert_fid_to_v240),
|
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(id3demux_id3v2_frames_to_tag_list):
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* gst/id3demux/id3tags.h:
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* gst/id3demux/id3v2.4.0-frames.txt:
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* gst/id3demux/id3v2.4.0-structure.txt:
|
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* gst/id3demux/id3v2frames.c: (id3demux_id3v2_parse_frame),
|
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(parse_comment_frame), (parse_text_identification_frame),
|
|
|
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(id3v2_tag_to_taglist), (parse_split_strings):
|
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|
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All new LGPL id3 demuxer. Can use zlib for compressed frames,
|
|
|
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otherwise it discards them. Works on my test files.
|
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|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_loop):
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Don't send EOS to a non-existing srcpad
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The debug category can be static
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|
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2005-12-16 21:57:51 +00:00
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2005-12-16 Stefan Kost <ensonic@users.sf.net>
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* ext/esd/esdmon.c: (gst_esdmon_open_audio):
|
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* ext/esd/esdsink.c: (gst_esdsink_prepare):
|
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* gst/multipart/multipartdemux.c:
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2006-01-11 11:39:10 +00:00
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change some char* into char[]
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2005-12-16 21:57:51 +00:00
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2005-12-16 19:32:53 +00:00
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2005-12-16 Wim Taymans <wim@fluendo.com>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
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(gst_wavparse_other), (gst_wavparse_perform_seek),
|
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
|
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|
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(gst_wavparse_loop), (gst_wavparse_pad_convert),
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(gst_wavparse_srcpad_event), (gst_wavparse_sink_activate_pull):
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* gst/wavparse/gstwavparse.h:
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Use GstSegment to implement more seeking features.
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2005-12-16 10:12:49 +00:00
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2005-12-16 Tim-Philipp Müller <tim at centricular dot net>
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* gst/rtsp/rtspconnection.c:
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Add <netinet/in.h> include and move <arpa/inet.h> include
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to make things work on OpenBSD as well (fixes #323717;
|
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|
patch by: Benjamin Pineau)
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2005-12-14 20:05:45 +00:00
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2005-12-14 Edgard Lima <edgard.lima@indt.org.br>
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* gst/rtp/gstrtpspeexdepay.c:
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* gst/rtp/gstrtpspeexpay.c:
|
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Set clock rate to be fixed in 8000. It fixes bug #324012.
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2005-12-14 18:07:16 +00:00
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2005-12-14 Philippe Khalaf <burger@speedy.org>
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* gst-plugins-good/gst/rtp/gstasteriskh263.c:
|
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|
* gst-plugins-good/gst/rtp/gstrtpamrdepay.c:
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* gst-plugins-good/gst/rtp/gstrtpamrpay.c:
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* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
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* gst-plugins-good/gst/rtp/gstrtpg711depay.c:
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* gst-plugins-good/gst/rtp/gstrtpgsmdepay.c:
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* gst-plugins-good/gst/rtp/gstrtph263pay.c:
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* gst-plugins-good/gst/rtp/gstrtph263pdepay.c:
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* gst-plugins-good/gst/rtp/gstrtph263ppay.c:
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* gst-plugins-good/gst/rtp/gstrtpmp4vdepay.c:
|
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* gst-plugins-good/gst/rtp/gstrtpmp4vpay.c:
|
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* gst-plugins-good/gst/rtp/gstrtpmpadepay.c:
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* gst-plugins-good/gst/rtp/gstrtpmpapay.c:
|
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* gst-plugins-good/gst/rtp/README:
|
2006-01-11 11:39:10 +00:00
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Fixed payload range in payloder caps. Removed payload range completely
|
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from depayloaders as they don't require payload type in their caps.
|
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In effect, there isn't any specific payload type for any given codec,
|
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only suggestions.
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2005-12-14 18:07:16 +00:00
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Fixes bug #324011.
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2005-12-13 21:58:42 +00:00
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2005-12-13 Julien MOUTTE <julien@moutte.net>
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* gst/videomixer/videomixer.c: (gst_videomixer_init),
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(gst_videomixer_fill_queues), (gst_videomixer_blend_buffers),
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(gst_videomixer_collected): Code cleanup and re-enabling
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queued time validity check for correct EOS handling.
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2005-12-13 17:18:32 +00:00
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2005-12-13 Tim-Philipp Müller <tim at centricular dot net>
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* sys/oss/gstossmixerelement.c: (gst_oss_mixer_element_class_init),
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(gst_oss_mixer_element_get_property),
|
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(gst_oss_mixer_element_change_state):
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Add 'device-name' property and fix state change function.
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2005-12-13 10:45:04 +00:00
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2005-12-13 Edward Hervey <edward@fluendo.com>
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* gst/flx/gstflxdec.c: (gst_flxdec_chain):
|
2006-01-11 11:39:10 +00:00
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If the speed of the file is null in the header, set the frame_time to
|
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the default setting of GST_SECOND / 70. Which is the default
|
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|
frame_delay for .fli files as stated in this document :
|
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|
http://www.compuphase.com/flic.htm
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|
Would be nice to have the time conversion done properly too (duration =
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flxh->frames * flxdec->frame_time)
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2005-12-13 10:45:04 +00:00
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2005-12-12 22:29:34 +00:00
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|
2005-12-12 Julien MOUTTE <julien@moutte.net>
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* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps),
|
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(gst_videomixer_getcaps), (gst_videomixer_fill_queues),
|
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|
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(gst_videomixer_update_queues), (gst_videomixer_collected): Adding
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|
documentation for videomixer on my way with a funny sample
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pipeline.
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2005-12-12 21:43:00 +00:00
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2005-12-12 Julien MOUTTE <julien@moutte.net>
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* gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps),
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(gst_videomixer_getcaps), (gst_videomixer_fill_queues),
|
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|
(gst_videomixer_update_queues), (gst_videomixer_collected):
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|
Fix caps negotiation. (#323896)
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|
2005-12-12 18:14:58 +00:00
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2005-12-12 Arwed v. Merkatz <v.merkatz@gmx.net>
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* gst/matroska/matroska-demux.c:
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(gst_matroska_demux_parse_blockgroup_or_simpleblock):
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Set correct timestamps on audio laces, fixes playback of mp3 from
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matroska.
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2005-12-12 10:30:20 +00:00
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2005-12-12 Tim-Philipp Müller <tim at centricular dot net>
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* gst/auparse/gstauparse.c: (gst_au_parse_base_init),
|
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(gst_au_parse_class_init), (gst_au_parse_init),
|
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(gst_au_parse_dispose), (gst_au_parse_chain),
|
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(gst_au_parse_change_state), (plugin_init):
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* gst/auparse/gstauparse.h:
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|
Use gst_object_unref() for GstObjects instead of
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g_object_unref() and fix a mem leak in a debug
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statement; while we're at it, also borgify, use
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boilerplate macros and clean up a little bit.
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2005-12-11 20:27:06 +00:00
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2005-12-11 Edward Hervey <edward@fluendo.com>
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* gst/debug/efence.c: (gst_efence_init), (gst_efence_getrange),
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(gst_efence_checkgetrange), (gst_efence_activate_src_pull):
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Added pull mode.
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2005-12-11 19:25:41 +00:00
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2005-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/goom/gstgoom.c:
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* gst/level/level-example.c: (main):
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* gst/smoothwave/demo-osssrc.c: (main):
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Use audiotestsrc instead of sinesrc (#323798).
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2005-12-11 17:50:50 +00:00
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2005-12-11 Stefan Kost <ensonic@users.sf.net>
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* sys/oss/gstosssink.c: (gst_oss_sink_class_init):
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2006-01-11 11:39:10 +00:00
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more debug-func-ptr usage
|
2005-12-11 17:50:50 +00:00
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2005-12-11 16:43:42 +00:00
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2005-12-11 Zeeshan Ali <zeenix at gmail dot com>
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* gst/flx/flx_color.c: (flx_colorspace_convert):
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* gst/flx/flx_color.h:
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* gst/flx/flx_fmt.h:
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* gst/flx/gstflxdec.c: (flx_decode_chunks), (gst_flxdec_chain):
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* gst/flx/gstflxdec.h:
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Now flxdec works on big-endian machines as well.
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2005-12-11 16:14:22 +00:00
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2005-12-11 Tim-Philipp Müller <tim at centricular dot net>
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* gst/debug/efence.c: (gst_efence_init), (gst_efence_chain),
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(gst_fenced_buffer_copy):
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Make sure GST_BUFFER_DATA is set on fenced copied buffers; fix
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GST_DEBUG crasher where GST_TIME_FORMAT was not used in
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conjunction with GST_TIME_ARGS. Also, don't leak pad templates
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and use GST_DEBUG_FUNCPTR for pad functions.
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ext/flac/gstflacdec.*: Rewrite flacdec a bit, so that even seeking might work now. Most importantly, don't act upon a...
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flac_dec_base_init),
(gst_flac_dec_class_init), (gst_flac_dec_init),
(gst_flac_dec_metadata_callback), (gst_flac_dec_error_callback),
(gst_flac_dec_eof), (gst_flac_dec_write), (gst_flac_dec_loop),
(gst_flac_dec_convert_src), (gst_flac_dec_get_src_query_types),
(gst_flac_dec_src_query), (gst_flac_dec_send_newsegment),
(gst_flac_dec_handle_seek_event), (gst_flac_dec_src_event),
(gst_flac_dec_change_state):
* ext/flac/gstflacdec.h:
Rewrite flacdec a bit, so that even seeking might work now. Most
importantly, don't act upon any flow return values we get, just tell
the decoder everything's dandy and act on the flow return values
later on in the loop function. We don't want to mess up the internal
decoder state for non-fatal things like flushing pads etc. Other
than that, use GstSegment (segment seeks don't work yet though, but
should be easy to add), use boilerplate macros, drop the superfluous
'flacdec:' from debug messages, use gst_util_uint64_scale_int, and
lots of other things.
2005-12-10 20:26:33 +00:00
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2005-12-10 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (gst_flac_dec_base_init),
|
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|
(gst_flac_dec_class_init), (gst_flac_dec_init),
|
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|
|
(gst_flac_dec_metadata_callback), (gst_flac_dec_error_callback),
|
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(gst_flac_dec_eof), (gst_flac_dec_write), (gst_flac_dec_loop),
|
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|
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(gst_flac_dec_convert_src), (gst_flac_dec_get_src_query_types),
|
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(gst_flac_dec_src_query), (gst_flac_dec_send_newsegment),
|
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(gst_flac_dec_handle_seek_event), (gst_flac_dec_src_event),
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(gst_flac_dec_change_state):
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* ext/flac/gstflacdec.h:
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|
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|
Rewrite flacdec a bit, so that even seeking might work now. Most
|
|
|
|
importantly, don't act upon any flow return values we get, just tell
|
|
|
|
the decoder everything's dandy and act on the flow return values
|
|
|
|
later on in the loop function. We don't want to mess up the internal
|
|
|
|
decoder state for non-fatal things like flushing pads etc. Other
|
|
|
|
than that, use GstSegment (segment seeks don't work yet though, but
|
|
|
|
should be easy to add), use boilerplate macros, drop the superfluous
|
|
|
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'flacdec:' from debug messages, use gst_util_uint64_scale_int, and
|
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lots of other things.
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|
2005-12-10 14:57:48 +00:00
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2005-12-10 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Update comment in OSS includes check.
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* sys/oss/gstossdmabuffer.c:
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* sys/oss/gstosshelper.c:
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* sys/oss/gstossmixer.c:
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* sys/oss/gstossmixertrack.c:
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* sys/oss/gstosssink.c:
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* sys/oss/gstosssrc.c:
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* sys/oss/oss_probe.c:
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Don't assume the OSS soundcard.h include is always in
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the sys/ directory. Instead, use the existing defines
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from config.h to include the right file. Fixes
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compilation on OpenBSD 3.8 (#323718).
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2005-12-09 19:51:03 +00:00
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2005-12-09 Thomas Vander Stichele <thomas at apestaart dot org>
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.hierarchy:
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* docs/plugins/inspect/plugin-1394.xml:
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* docs/plugins/inspect/plugin-aasink.xml:
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* docs/plugins/inspect/plugin-alaw.xml:
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* docs/plugins/inspect/plugin-alpha.xml:
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* docs/plugins/inspect/plugin-alphacolor.xml:
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* docs/plugins/inspect/plugin-auparse.xml:
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* docs/plugins/inspect/plugin-autodetect.xml:
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* docs/plugins/inspect/plugin-avi.xml:
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* docs/plugins/inspect/plugin-cacasink.xml:
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* docs/plugins/inspect/plugin-cairo.xml:
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* docs/plugins/inspect/plugin-cutter.xml:
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* docs/plugins/inspect/plugin-debug.xml:
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* docs/plugins/inspect/plugin-dv.xml:
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* docs/plugins/inspect/plugin-efence.xml:
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* docs/plugins/inspect/plugin-effectv.xml:
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* docs/plugins/inspect/plugin-esdsink.xml:
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|
* docs/plugins/inspect/plugin-flac.xml:
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* docs/plugins/inspect/plugin-flxdec.xml:
|
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* docs/plugins/inspect/plugin-gconfelements.xml:
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* docs/plugins/inspect/plugin-goom.xml:
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* docs/plugins/inspect/plugin-jpeg.xml:
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* docs/plugins/inspect/plugin-level.xml:
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* docs/plugins/inspect/plugin-matroska.xml:
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* docs/plugins/inspect/plugin-mulaw.xml:
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* docs/plugins/inspect/plugin-multipart.xml:
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* docs/plugins/inspect/plugin-navigationtest.xml:
|
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|
* docs/plugins/inspect/plugin-ossaudio.xml:
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* docs/plugins/inspect/plugin-png.xml:
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* docs/plugins/inspect/plugin-rtp.xml:
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* docs/plugins/inspect/plugin-rtsp.xml:
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* docs/plugins/inspect/plugin-shout2send.xml:
|
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* docs/plugins/inspect/plugin-smpte.xml:
|
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|
* docs/plugins/inspect/plugin-speex.xml:
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* docs/plugins/inspect/plugin-udp.xml:
|
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|
|
* docs/plugins/inspect/plugin-videobox.xml:
|
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* docs/plugins/inspect/plugin-videoflip.xml:
|
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|
* docs/plugins/inspect/plugin-videomixer.xml:
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|
* docs/plugins/inspect/plugin-wavenc.xml:
|
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|
* docs/plugins/inspect/plugin-wavparse.xml:
|
|
|
|
* ext/flac/gstflac.c: (plugin_init):
|
|
|
|
* ext/flac/gstflacdec.c: (flacdec_get_type),
|
|
|
|
(gst_flac_dec_base_init), (gst_flac_dec_class_init),
|
|
|
|
(gst_flac_dec_init), (gst_flac_dec_finalize),
|
|
|
|
(gst_flac_dec_update_metadata), (gst_flac_dec_metadata_callback),
|
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|
|
(gst_flac_dec_error_callback), (gst_flac_dec_seek),
|
|
|
|
(gst_flac_dec_tell), (gst_flac_dec_length), (gst_flac_dec_eof),
|
|
|
|
(gst_flac_dec_read), (gst_flac_dec_write), (gst_flac_dec_loop),
|
|
|
|
(gst_flac_dec_get_src_formats), (gst_flac_dec_convert_src),
|
|
|
|
(gst_flac_dec_get_src_query_types), (gst_flac_dec_src_query),
|
|
|
|
(gst_flac_dec_src_event), (gst_flac_dec_sink_activate),
|
|
|
|
(gst_flac_dec_sink_activate_pull), (gst_flac_dec_change_state):
|
|
|
|
* ext/flac/gstflacdec.h:
|
|
|
|
* ext/flac/gstflacenc.c: (gst_flac_enc_quality_get_type),
|
|
|
|
(gst_flac_enc_base_init), (gst_flac_enc_class_init),
|
|
|
|
(gst_flac_enc_init), (gst_flac_enc_finalize), (add_one_tag),
|
|
|
|
(gst_flac_enc_set_metadata), (gst_flac_enc_sink_setcaps),
|
|
|
|
(gst_flac_enc_update_quality), (gst_flac_enc_seek_callback),
|
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|
|
(gst_flac_enc_write_callback), (gst_flac_enc_tell_callback),
|
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|
|
(gst_flac_enc_sink_event), (gst_flac_enc_chain),
|
|
|
|
(gst_flac_enc_set_property), (gst_flac_enc_get_property),
|
|
|
|
(gst_flac_enc_change_state):
|
|
|
|
* ext/flac/gstflacenc.h:
|
|
|
|
borgify and fix up documentation
|
|
|
|
|
2005-12-09 11:12:48 +00:00
|
|
|
2005-12-09 Michael Smith <msmith@fluendo.com>
|
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|
|
|
|
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* ext/flac/gstflacdec.c: (raw_caps_factory), (gst_flacdec_write):
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|
Accept a wider range of flac files, more closely matching flac spec.
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|
2005-12-08 16:27:12 +00:00
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|
2005-12-08 Julien MOUTTE <julien@moutte.net>
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|
|
|
|
|
|
* docs/plugins/Makefile.am: Add multipart elements.
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt: Fix flac.
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.hierarchy:
|
|
|
|
* gst/multipart/multipartdemux.c:
|
|
|
|
* gst/multipart/multipartmux.c: Add docs.
|
|
|
|
|
2005-12-07 11:40:46 +00:00
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|
|
2005-12-07 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_event),
|
|
|
|
(gst_avi_demux_parse_stream), (gst_avi_demux_stream_header),
|
|
|
|
(gst_avi_demux_invert):
|
|
|
|
Memleak and crasher fixes.
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_base_init),
|
|
|
|
(gst_wavparse_create_sourcepad), (gst_wavparse_stream_headers):
|
|
|
|
Memleak fixes
|
|
|
|
|
2005-12-06 19:44:58 +00:00
|
|
|
2005-12-06 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.h:
|
|
|
|
* ext/cairo/gsttextoverlay.h:
|
|
|
|
* ext/dv/gstdvdec.h:
|
|
|
|
* ext/dv/gstdvdemux.c:
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
* ext/esd/esdsink.h:
|
|
|
|
* ext/flac/flac_compat.h:
|
|
|
|
* ext/flac/gstflacdec.h:
|
|
|
|
* ext/flac/gstflacenc.h:
|
|
|
|
* ext/gconf/gconf.h:
|
|
|
|
* ext/gconf/gstgconfaudiosink.h:
|
|
|
|
* ext/gconf/gstgconfvideosink.h:
|
|
|
|
* ext/gdk_pixbuf/gstgdkanimation.h:
|
|
|
|
* ext/jpeg/gstjpegdec.h:
|
|
|
|
* ext/jpeg/smokecodec.h:
|
|
|
|
* ext/jpeg/smokeformat.h:
|
|
|
|
* ext/ladspa/gstsignalprocessor.h:
|
|
|
|
* ext/ladspa/search.c: (LADSPAPluginSearch):
|
|
|
|
* ext/ladspa/utils.h:
|
|
|
|
* ext/libmng/gstmng.c:
|
|
|
|
* ext/libmng/gstmngdec.h:
|
|
|
|
* ext/libmng/gstmngenc.c:
|
|
|
|
* ext/libmng/gstmngenc.h:
|
|
|
|
* ext/libpng/gstpng.c:
|
|
|
|
* ext/libpng/gstpngenc.c:
|
|
|
|
* ext/libpng/gstpngenc.h:
|
|
|
|
* ext/shout2/gstshout2.h:
|
|
|
|
* ext/speex/gstspeexdec.h:
|
|
|
|
* ext/speex/gstspeexenc.c:
|
|
|
|
* ext/speex/gstspeexenc.h:
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_chain):
|
|
|
|
* gst/autodetect/gstautoaudiosink.h:
|
|
|
|
* gst/autodetect/gstautovideosink.h:
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
* gst/cutter/gstcutter.c: (gst_cutter_chain),
|
|
|
|
(gst_cutter_set_property), (gst_cutter_get_caps):
|
|
|
|
* gst/cutter/gstcutter.h:
|
|
|
|
* gst/debug/tests.c: (md5_process_block):
|
|
|
|
* gst/debug/tests.h:
|
|
|
|
* gst/effectv/gstwarp.c:
|
|
|
|
* gst/flx/flx_fmt.h:
|
|
|
|
* gst/flx/gstflxdec.h:
|
|
|
|
* gst/goom/filters.c: (setPixelRGB), (getPixelRGB), (getPixelRGB_),
|
|
|
|
(zoomFilterFastRGB):
|
|
|
|
* gst/goom/filters.h:
|
|
|
|
* gst/goom/goom_tools.h:
|
|
|
|
* gst/law/alaw-encode.c:
|
|
|
|
* gst/level/gstlevel.c:
|
|
|
|
* gst/level/gstlevel.h:
|
|
|
|
* gst/matroska/ebml-write.h:
|
|
|
|
* gst/matroska/matroska-demux.h:
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
|
|
* gst/matroska/matroska-mux.h:
|
|
|
|
* gst/monoscope/convolve.c: (convolve_match):
|
|
|
|
* gst/monoscope/convolve.h:
|
|
|
|
* gst/multipart/multipartmux.c:
|
|
|
|
* gst/oldcore/gstaggregator.c:
|
|
|
|
* gst/oldcore/gstaggregator.h:
|
|
|
|
* gst/oldcore/gstmd5sink.c: (md5_process_block):
|
|
|
|
* gst/oldcore/gstmd5sink.h:
|
|
|
|
* gst/oldcore/gstmultifilesrc.c:
|
|
|
|
* gst/oldcore/gstmultifilesrc.h:
|
|
|
|
* gst/oldcore/gstpipefilter.h:
|
|
|
|
* gst/oldcore/gstshaper.h:
|
|
|
|
* gst/rtp/gstrtpL16depay.h:
|
|
|
|
* gst/rtp/gstrtpL16pay.h:
|
|
|
|
* gst/rtp/gstrtpdepay.h:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.h:
|
|
|
|
* gst/rtsp/gstrtspsrc.c:
|
|
|
|
* gst/rtsp/gstrtspsrc.h:
|
|
|
|
* gst/rtsp/rtspconnection.h:
|
|
|
|
* gst/rtsp/rtspdefs.h:
|
|
|
|
* gst/rtsp/rtspmessage.h:
|
|
|
|
* gst/rtsp/rtsptransport.h:
|
|
|
|
* gst/rtsp/rtspurl.c:
|
|
|
|
* gst/rtsp/rtspurl.h:
|
|
|
|
* gst/rtsp/sdpmessage.c:
|
|
|
|
* gst/rtsp/sdpmessage.h:
|
|
|
|
* gst/smpte/barboxwipes.c:
|
|
|
|
* gst/smpte/gstmask.h:
|
|
|
|
* gst/smpte/gstsmpte.h:
|
|
|
|
* gst/smpte/paint.c:
|
|
|
|
* gst/smpte/paint.h:
|
|
|
|
* gst/udp/gstdynudpsink.h:
|
|
|
|
* gst/udp/gstmultiudpsink.h:
|
|
|
|
* gst/udp/gstudpsink.c:
|
|
|
|
* gst/udp/gstudpsink.h:
|
|
|
|
* gst/udp/gstudpsrc.c:
|
|
|
|
* gst/videomixer/videomixer.c:
|
|
|
|
* gst/wavenc/riff.h:
|
|
|
|
* gst/wavparse/gstwavparse.h:
|
|
|
|
* sys/oss/gstossdmabuffer.h:
|
|
|
|
* sys/oss/gstossmixer.h:
|
|
|
|
* sys/oss/gstossmixerelement.h:
|
|
|
|
* sys/oss/gstossmixertrack.h:
|
|
|
|
* sys/oss/gstosssink.c:
|
|
|
|
* sys/oss/gstosssink.h:
|
|
|
|
* sys/oss/gstosssrc.c:
|
|
|
|
* sys/oss/gstosssrc.h:
|
|
|
|
* sys/osxaudio/gstosxaudioelement.h:
|
|
|
|
* sys/osxaudio/gstosxaudiosink.h:
|
|
|
|
* sys/osxaudio/gstosxaudiosrc.h:
|
|
|
|
expand tabs
|
|
|
|
|
2005-12-05 18:03:23 +00:00
|
|
|
=== release 0.10.0 ===
|
|
|
|
|
|
|
|
2005-12-05 <thomas (at) apestaart (dot) org>
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|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.10.0, "Abondance"
|
|
|
|
|
2005-12-05 13:03:00 +00:00
|
|
|
2005-12-05 Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_chain):
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flacdec_write):
|
|
|
|
* ext/flac/gstflacenc.c: (gst_flacenc_write_callback):
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_chain):
|
|
|
|
* ext/gdk_pixbuf/pixbufscale.c: (gst_pixbufscale_chain):
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_chain):
|
|
|
|
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_process):
|
|
|
|
* ext/libpng/gstpngdec.c: (user_info_callback), (gst_pngdec_task):
|
|
|
|
* ext/speex/gstspeexdec.c: (speex_dec_chain):
|
|
|
|
* ext/speex/gstspeexenc.c: (gst_speexenc_chain):
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_chain):
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_chain):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_chain):
|
|
|
|
* gst/matroska/matroska-demux.c:
|
|
|
|
(gst_matroska_demux_push_vorbis_codec_priv_data),
|
|
|
|
(gst_matroska_demux_add_wvpk_header):
|
|
|
|
* gst/multipart/multipartdemux.c: (gst_multipart_demux_chain):
|
|
|
|
* gst/multipart/multipartmux.c: (gst_multipart_mux_collected):
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_collected):
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Update for
|
|
|
|
alloc_buffer changes.
|
|
|
|
|
2005-12-05 12:23:22 +00:00
|
|
|
2005-12-05 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
Remove args for plugins that aren't in -good.
|
|
|
|
|
2005-12-01 19:14:26 +00:00
|
|
|
=== release 0.9.7 ===
|
|
|
|
|
|
|
|
2005-12-01 <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.9.7, "Mijn Hond Heeft Geen Neus"
|
|
|
|
|
2005-12-01 15:34:13 +00:00
|
|
|
2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* docs/plugins/.cvsignore:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/inspect/plugin-multipart.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtp.xml:
|
|
|
|
add multipart plugin to docs
|
|
|
|
|
2005-12-01 15:22:25 +00:00
|
|
|
2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* ext/Makefile.am:
|
|
|
|
* ext/pango/Makefile.am:
|
|
|
|
* ext/pango/gstclockoverlay.c:
|
|
|
|
* ext/pango/gstclockoverlay.h:
|
|
|
|
* ext/pango/gsttextoverlay.c:
|
|
|
|
* ext/pango/gsttextoverlay.h:
|
|
|
|
* ext/pango/gsttextrender.c:
|
|
|
|
* ext/pango/gsttextrender.h:
|
|
|
|
* ext/pango/gsttimeoverlay.c:
|
|
|
|
* ext/pango/gsttimeoverlay.h:
|
|
|
|
move pango to base
|
|
|
|
|
2005-12-01 14:39:30 +00:00
|
|
|
2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstrtpL16depay.c:
|
|
|
|
* gst/rtp/gstrtpL16depay.h:
|
|
|
|
* gst/rtp/gstrtpL16parse.c:
|
|
|
|
* gst/rtp/gstrtpL16parse.h:
|
|
|
|
* gst/rtp/gstrtpgsmdepay.c:
|
|
|
|
* gst/rtp/gstrtpgsmdepay.h:
|
|
|
|
* gst/rtp/gstrtpgsmparse.c:
|
|
|
|
* gst/rtp/gstrtpgsmparse.h:
|
|
|
|
parsers are depayers
|
|
|
|
|
2005-12-01 14:30:01 +00:00
|
|
|
2005-12-01 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstasteriskh263.c:
|
|
|
|
* gst/rtp/gstrtp.c:
|
|
|
|
* gst/rtp/gstrtpL16enc.c:
|
|
|
|
* gst/rtp/gstrtpL16enc.h:
|
|
|
|
* gst/rtp/gstrtpL16parse.c:
|
|
|
|
* gst/rtp/gstrtpL16parse.h:
|
|
|
|
* gst/rtp/gstrtpL16pay.c:
|
|
|
|
* gst/rtp/gstrtpL16pay.h:
|
|
|
|
* gst/rtp/gstrtpamrdec.c:
|
|
|
|
* gst/rtp/gstrtpamrdec.h:
|
|
|
|
* gst/rtp/gstrtpamrdepay.c:
|
|
|
|
* gst/rtp/gstrtpamrdepay.h:
|
|
|
|
* gst/rtp/gstrtpamrenc.c:
|
|
|
|
* gst/rtp/gstrtpamrenc.h:
|
|
|
|
* gst/rtp/gstrtpamrpay.c:
|
|
|
|
* gst/rtp/gstrtpamrpay.h:
|
|
|
|
* gst/rtp/gstrtpdec.c:
|
|
|
|
* gst/rtp/gstrtpdec.h:
|
|
|
|
* gst/rtp/gstrtpdepay.c:
|
|
|
|
* gst/rtp/gstrtpdepay.h:
|
|
|
|
* gst/rtp/gstrtpg711dec.c:
|
|
|
|
* gst/rtp/gstrtpg711dec.h:
|
|
|
|
* gst/rtp/gstrtpg711depay.c:
|
|
|
|
* gst/rtp/gstrtpg711depay.h:
|
|
|
|
* gst/rtp/gstrtpg711enc.c:
|
|
|
|
* gst/rtp/gstrtpg711enc.h:
|
|
|
|
* gst/rtp/gstrtpg711pay.c:
|
|
|
|
* gst/rtp/gstrtpg711pay.h:
|
|
|
|
* gst/rtp/gstrtpgsmenc.c:
|
|
|
|
* gst/rtp/gstrtpgsmenc.h:
|
|
|
|
* gst/rtp/gstrtpgsmparse.c:
|
|
|
|
* gst/rtp/gstrtpgsmparse.h:
|
|
|
|
* gst/rtp/gstrtpgsmpay.c:
|
|
|
|
* gst/rtp/gstrtpgsmpay.h:
|
|
|
|
* gst/rtp/gstrtph263enc.c:
|
|
|
|
* gst/rtp/gstrtph263enc.h:
|
|
|
|
* gst/rtp/gstrtph263pay.c:
|
|
|
|
* gst/rtp/gstrtph263pay.h:
|
|
|
|
* gst/rtp/gstrtph263pdec.c:
|
|
|
|
* gst/rtp/gstrtph263pdec.h:
|
|
|
|
* gst/rtp/gstrtph263pdepay.c:
|
|
|
|
* gst/rtp/gstrtph263pdepay.h:
|
|
|
|
* gst/rtp/gstrtph263penc.c:
|
|
|
|
* gst/rtp/gstrtph263penc.h:
|
|
|
|
* gst/rtp/gstrtph263ppay.c:
|
|
|
|
* gst/rtp/gstrtph263ppay.h:
|
|
|
|
* gst/rtp/gstrtpmp4vdec.c:
|
|
|
|
* gst/rtp/gstrtpmp4vdec.h:
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vdepay.h:
|
|
|
|
* gst/rtp/gstrtpmp4venc.c:
|
|
|
|
* gst/rtp/gstrtpmp4venc.h:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.c:
|
|
|
|
* gst/rtp/gstrtpmp4vpay.h:
|
|
|
|
* gst/rtp/gstrtpmpadec.c:
|
|
|
|
* gst/rtp/gstrtpmpadec.h:
|
|
|
|
* gst/rtp/gstrtpmpadepay.c:
|
|
|
|
* gst/rtp/gstrtpmpadepay.h:
|
|
|
|
* gst/rtp/gstrtpmpaenc.c:
|
|
|
|
* gst/rtp/gstrtpmpaenc.h:
|
|
|
|
* gst/rtp/gstrtpmpapay.c:
|
|
|
|
* gst/rtp/gstrtpmpapay.h:
|
|
|
|
* gst/rtp/gstrtpspeexdec.c:
|
|
|
|
* gst/rtp/gstrtpspeexdec.h:
|
|
|
|
* gst/rtp/gstrtpspeexdepay.c:
|
|
|
|
* gst/rtp/gstrtpspeexdepay.h:
|
|
|
|
* gst/rtp/gstrtpspeexenc.c:
|
|
|
|
* gst/rtp/gstrtpspeexenc.h:
|
|
|
|
* gst/rtp/gstrtpspeexpay.c:
|
|
|
|
* gst/rtp/gstrtpspeexpay.h:
|
|
|
|
Do burger's rename for rtp payloaders and depayloaders
|
|
|
|
|
2005-11-30 19:02:35 +00:00
|
|
|
2005-11-30 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_chain):
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
Fix seeking in dvdemux again, add some more debug info.
|
|
|
|
|
2005-11-30 18:48:56 +00:00
|
|
|
2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
fix tests
|
|
|
|
|
2005-11-30 18:28:53 +00:00
|
|
|
2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* PORTED_09:
|
|
|
|
* docs/random/PORTED_09:
|
|
|
|
move
|
|
|
|
* tests/Makefile.am:
|
|
|
|
add
|
|
|
|
* win32/gst.sln:
|
|
|
|
remove
|
|
|
|
|
2005-11-30 18:24:08 +00:00
|
|
|
2005-11-30 Thomas Vander Stichele <thomas (at) apestaart (dot) org>
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
* check/.cvsignore:
|
|
|
|
* check/Makefile.am:
|
|
|
|
* check/elements/.cvsignore:
|
|
|
|
* check/elements/level.c:
|
|
|
|
* check/elements/matroskamux.c:
|
|
|
|
* configure.ac:
|
|
|
|
* examples/Makefile.am:
|
|
|
|
* examples/capsfilter/Makefile.am:
|
|
|
|
* examples/capsfilter/capsfilter1.c:
|
|
|
|
* examples/gob/Makefile.am:
|
|
|
|
* examples/gob/gst-identity2.gob:
|
|
|
|
* examples/gstplay/.cvsignore:
|
|
|
|
* examples/gstplay/Makefile.am:
|
|
|
|
* examples/gstplay/player.c:
|
|
|
|
* examples/indexing/.cvsignore:
|
|
|
|
* examples/indexing/Makefile.am:
|
|
|
|
* examples/indexing/indexmpeg.c:
|
|
|
|
* examples/level/Makefile.am:
|
|
|
|
* examples/level/README:
|
|
|
|
* examples/level/demo.c:
|
|
|
|
* examples/level/plot.c:
|
|
|
|
* examples/stats/Makefile.am:
|
|
|
|
* examples/stats/mp2ogg.c:
|
|
|
|
* examples/switch/.cvsignore:
|
|
|
|
* examples/switch/Makefile.am:
|
|
|
|
* examples/switch/switcher.c:
|
|
|
|
move under tests/
|
|
|
|
|
ext/pango/gsttextrender.*: Add missing files.
Original commit message from CVS:
* ext/pango/gsttextrender.c: (gst_text_render_base_init),
(gst_text_render_class_init), (resize_bitmap),
(gst_text_render_render_text), (gst_text_render_setcaps),
(gst_text_render_fixate_caps), (gst_text_renderer_bitmap_to_ayuv),
(gst_text_render_chain), (gst_text_render_finalize),
(gst_text_render_init), (gst_text_render_set_property):
* ext/pango/gsttextrender.h:
Add missing files.
2005-11-30 14:53:29 +00:00
|
|
|
2005-11-30 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/pango/gsttextrender.c: (gst_text_render_base_init),
|
|
|
|
(gst_text_render_class_init), (resize_bitmap),
|
|
|
|
(gst_text_render_render_text), (gst_text_render_setcaps),
|
|
|
|
(gst_text_render_fixate_caps), (gst_text_renderer_bitmap_to_ayuv),
|
|
|
|
(gst_text_render_chain), (gst_text_render_finalize),
|
|
|
|
(gst_text_render_init), (gst_text_render_set_property):
|
|
|
|
* ext/pango/gsttextrender.h:
|
|
|
|
Add missing files.
|
|
|
|
|
Port pango-based textoverlay, timeoverlay and textrender to 0.9 and add background shading and text wrapping modes. M...
Original commit message from CVS:
* configure.ac:
* ext/Makefile.am:
* ext/pango/Makefile.am:
* ext/pango/gstclockoverlay.c: (gst_clock_overlay_base_init),
(gst_clock_overlay_render_time), (gst_clock_overlay_get_text),
(gst_clock_overlay_class_init), (gst_clock_overlay_init):
* ext/pango/gstclockoverlay.h:
* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_get_text), (gst_text_overlay_class_init),
(gst_text_overlay_finalize), (gst_text_overlay_init),
(gst_text_overlay_update_wrap_mode), (gst_text_overlay_setcaps),
(gst_text_overlay_text_pad_linked),
(gst_text_overlay_text_pad_unlinked),
(gst_text_overlay_set_property), (gst_text_overlay_getcaps),
(gst_text_overlay_shade_y), (gst_text_overlay_blit_yuv420),
(gst_text_overlay_resize_bitmap), (gst_text_overlay_render_text),
(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
(gst_text_overlay_change_state), (plugin_init):
* ext/pango/gsttextoverlay.h:
* ext/pango/gsttimeoverlay.c: (gst_time_overlay_base_init),
(gst_time_overlay_render_time), (gst_time_overlay_get_text),
(gst_time_overlay_class_init), (gst_time_overlay_init):
* ext/pango/gsttimeoverlay.h:
Port pango-based textoverlay, timeoverlay and textrender to 0.9
and add background shading and text wrapping modes. Make
timoverlay derive from textoverlay. Also add new clockoverlay
element.
2005-11-30 13:20:57 +00:00
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|
2005-11-30 Tim-Philipp Müller <tim at centricular dot net>
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|
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|
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|
* configure.ac:
|
|
|
|
* ext/Makefile.am:
|
|
|
|
* ext/pango/Makefile.am:
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* ext/pango/gstclockoverlay.c: (gst_clock_overlay_base_init),
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(gst_clock_overlay_render_time), (gst_clock_overlay_get_text),
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(gst_clock_overlay_class_init), (gst_clock_overlay_init):
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* ext/pango/gstclockoverlay.h:
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* ext/pango/gsttextoverlay.c: (gst_text_overlay_base_init),
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(gst_text_overlay_get_text), (gst_text_overlay_class_init),
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(gst_text_overlay_finalize), (gst_text_overlay_init),
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(gst_text_overlay_update_wrap_mode), (gst_text_overlay_setcaps),
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(gst_text_overlay_text_pad_linked),
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(gst_text_overlay_text_pad_unlinked),
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(gst_text_overlay_set_property), (gst_text_overlay_getcaps),
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(gst_text_overlay_shade_y), (gst_text_overlay_blit_yuv420),
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(gst_text_overlay_resize_bitmap), (gst_text_overlay_render_text),
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(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
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(gst_text_overlay_pop_text), (gst_text_overlay_collected),
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(gst_text_overlay_change_state), (plugin_init):
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* ext/pango/gsttextoverlay.h:
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* ext/pango/gsttimeoverlay.c: (gst_time_overlay_base_init),
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(gst_time_overlay_render_time), (gst_time_overlay_get_text),
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(gst_time_overlay_class_init), (gst_time_overlay_init):
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* ext/pango/gsttimeoverlay.h:
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Port pango-based textoverlay, timeoverlay and textrender to 0.9
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and add background shading and text wrapping modes. Make
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timoverlay derive from textoverlay. Also add new clockoverlay
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element.
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2005-11-30 11:10:01 +00:00
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2005-11-30 Julien MOUTTE <julien@moutte.net>
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* gst/udp/Makefile.am: Moved to netbuffer.
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Ported multipart mux/demux to 0.9.
Original commit message from CVS:
2005-11-30 Julien MOUTTE <julien@moutte.net>
* configure.ac:
* PORTED_O9:
* gst/multipart/Makefile.am:
* gst/multipart/multipartdemux.c:
(gst_multipart_demux_base_init),
(gst_multipart_demux_class_init), (gst_multipart_demux_init),
(gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain),
(gst_multipart_demux_change_state),
(gst_multipart_demux_plugin_init):
* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
(gst_multipart_mux_init), (gst_multipart_mux_finalize),
(gst_multipart_mux_sinkconnect),
(gst_multipart_mux_request_new_pad),
(gst_multipart_mux_handle_src_event),
(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
(gst_multipart_mux_change_state): Ported multipart mux/demux to
0.9.
2005-11-30 10:18:42 +00:00
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2005-11-30 Julien MOUTTE <julien@moutte.net>
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* configure.ac:
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* PORTED_O9:
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* gst/multipart/Makefile.am:
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* gst/multipart/multipartdemux.c: (gst_multipart_demux_base_init),
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(gst_multipart_demux_class_init), (gst_multipart_demux_init),
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(gst_multipart_find_pad_by_mime), (gst_multipart_demux_chain),
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(gst_multipart_demux_change_state),
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(gst_multipart_demux_plugin_init):
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* gst/multipart/multipartmux.c: (gst_multipart_mux_class_init),
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(gst_multipart_mux_init), (gst_multipart_mux_finalize),
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(gst_multipart_mux_sinkconnect),
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(gst_multipart_mux_request_new_pad),
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(gst_multipart_mux_handle_src_event),
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(gst_multipart_mux_queue_pads), (gst_multipart_mux_collected),
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(gst_multipart_mux_change_state): Ported multipart mux/demux to
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0.9.
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2005-11-30 08:26:47 +00:00
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2005-11-30 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/debug/gstnavigationtest.c: (gst_navigationtest_get_type):
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* gst/debug/gstnavigationtest.h:
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* gst/effectv/gstaging.c: (gst_agingtv_get_type):
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* gst/effectv/gstdice.c: (gst_dicetv_get_type):
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* gst/effectv/gstedge.c: (gst_edgetv_get_type):
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* gst/effectv/gstquark.c: (gst_quarktv_get_type):
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* gst/effectv/gstrev.c: (gst_revtv_get_type):
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* gst/effectv/gstshagadelic.c: (gst_shagadelictv_get_type):
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* gst/effectv/gstvertigo.c: (gst_vertigotv_get_type):
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* gst/effectv/gstwarp.c: (gst_warptv_get_type):
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* gst/videofilter/gstvideoflip.c: (gst_video_flip_set_property),
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(gst_video_flip_get_type):
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* gst/videofilter/gstvideoflip.h:
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update for symbols change
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2005-11-29 17:46:04 +00:00
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2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/udp/gstdynudpsink.c:
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* gst/udp/gstudpsrc.c:
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the old gstnet lib was renamed gstnetbuffer (#322257)
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2005-11-29 15:42:01 +00:00
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2005-11-29 Tim-Philipp Müller <tim at centricular dot net>
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* ext/cairo/gsttextoverlay.c: (gst_text_overlay_render_text),
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(gst_text_overlay_collected):
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Actually render the text from the text pad.
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2005-11-29 14:49:00 +00:00
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2005-11-29 Edward Hervey <edward@fluendo.com>
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* gst/debug/gstnavseek.c: (gst_navseek_event):
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* gst/debug/progressreport.c: (gst_progress_report_event):
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Update for GstBaseTransform event virtual method
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2005-11-29 10:55:09 +00:00
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2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/cairo/Makefile.am:
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no need to link to videofilter
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2005-11-29 10:46:00 +00:00
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2005-11-29 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/debug/Makefile.am:
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* gst/debug/gstnavigationtest.h:
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* gst/effectv/Makefile.am:
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* gst/effectv/gstaging.c:
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* gst/effectv/gstdice.c:
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* gst/effectv/gstedge.c:
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* gst/effectv/gstquark.c:
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* gst/effectv/gstrev.c:
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* gst/effectv/gstshagadelic.c:
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* gst/effectv/gstvertigo.c:
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* gst/effectv/gstwarp.c:
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* gst/videofilter/Makefile.am:
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* gst/videofilter/gstvideofilter.c:
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* gst/videofilter/gstvideofilter.h:
|
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* gst/videofilter/gstvideoflip.h:
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remove the videofilter library and link to the one in base
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2005-11-28 17:31:44 +00:00
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2005-11-28 Edward Hervey <edward@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_process_next_entry):
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Useless check now we're setting the current entry correctly.
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2005-11-28 16:54:03 +00:00
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2005-11-28 Tim-Philipp Müller <tim at centricular dot net>
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* ext/jpeg/gstjpegenc.c: (gst_jpegenc_resync), (gst_jpegenc_chain),
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(gst_jpegenc_set_property), (gst_jpegenc_get_property),
|
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(gst_jpegenc_change_state):
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Don't leak input buffer in chain function (fixes #322667); make
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|
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state change function thread-safe; don't repeat the current function
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name in GST_DEBUG statements; use GST_ROUND_UP_* macros; use
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gst_pad_alloc_buffer(); misc. minor cleanups.
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2005-11-28 15:13:22 +00:00
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|
2005-11-28 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_video_pad_setcaps):
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|
Look for pixel-aspect-ratio in caps, not pixel_width and
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pixel_height (Fixes: #322645)
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2005-11-28 12:59:05 +00:00
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|
2005-11-28 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_video_pad_setcaps):
|
|
|
|
From Michal Benes:
|
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|
|
frame duration should be GST_SECOND / framerate, not
|
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|
|
GST_SECOND * framerate. (Fixes: #322643)
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2005-11-27 17:02:53 +00:00
|
|
|
2005-11-27 Thomas Vander Stichele <thomas at apestaart dot org>
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|
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* configure.ac:
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|
|
|
fix up GST_PLUGIN_LDFLAGS
|
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|
|
* gst/rtsp/rtspconnection.c:
|
|
|
|
fix includes (see #317043)
|
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|
|
* gst/videofilter/Makefile.am:
|
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|
|
stop installing this library
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|
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|
2005-11-26 12:54:47 +00:00
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|
2005-11-26 Edward Hervey <edward@fluendo.com>
|
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|
|
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|
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* ext/dv/gstdvdec.c: (gst_dvdec_sink_setcaps):
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|
|
|
Handle the case where the incoming Video dv stream doesn't have
|
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|
|
a pixel aspect ratio set.
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|
2005-11-25 22:14:47 +00:00
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|
|
2005-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
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* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* ext/flac/gstflacdec.c:
|
|
|
|
document flacdec
|
|
|
|
|
2005-11-25 21:36:18 +00:00
|
|
|
2005-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* docs/plugins/inspect/plugin-autodetect.xml:
|
|
|
|
* ext/cairo/gstcairo.c: (plugin_init):
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init),
|
|
|
|
(gst_text_overlay_class_init), (gst_text_overlay_finalize),
|
|
|
|
(gst_text_overlay_init), (gst_text_overlay_font_init),
|
|
|
|
(gst_text_overlay_set_property), (gst_text_overlay_render_text),
|
|
|
|
(gst_text_overlay_getcaps), (gst_text_overlay_setcaps),
|
|
|
|
(gst_text_overlay_text_pad_linked),
|
|
|
|
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y),
|
|
|
|
(gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
|
|
|
|
(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
|
|
|
|
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
|
|
|
|
(gst_text_overlay_change_state):
|
|
|
|
* ext/cairo/gsttextoverlay.h:
|
|
|
|
* ext/cairo/gsttimeoverlay.c:
|
|
|
|
(gst_cairo_time_overlay_update_font_height),
|
|
|
|
(gst_cairo_time_overlay_set_caps),
|
|
|
|
(gst_cairo_time_overlay_get_unit_size),
|
|
|
|
(gst_cairo_time_overlay_print_smpte_time),
|
|
|
|
(gst_cairo_time_overlay_transform),
|
|
|
|
(gst_cairo_time_overlay_base_init),
|
|
|
|
(gst_cairo_time_overlay_class_init), (gst_cairo_time_overlay_init),
|
|
|
|
(gst_cairo_time_overlay_get_type):
|
|
|
|
* ext/cairo/gsttimeoverlay.h:
|
|
|
|
do some name borgifying
|
|
|
|
document
|
|
|
|
|
2005-11-25 21:02:16 +00:00
|
|
|
2005-11-25 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
|
|
|
|
* docs/plugins/gst-plugins-good-plugins-sections.txt:
|
|
|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
(gst_auto_audio_sink_base_init):
|
|
|
|
* gst/autodetect/gstautovideosink.c:
|
|
|
|
(gst_auto_video_sink_base_init),
|
|
|
|
(gst_auto_video_sink_factory_filter):
|
|
|
|
documenting auto*sink
|
|
|
|
using strstr for the video sink lookup, class field is not ordered
|
|
|
|
* docs/plugins/inspect/plugin-1394.xml:
|
|
|
|
* docs/plugins/inspect/plugin-aasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alpha.xml:
|
|
|
|
* docs/plugins/inspect/plugin-alphacolor.xml:
|
|
|
|
* docs/plugins/inspect/plugin-auparse.xml:
|
|
|
|
* docs/plugins/inspect/plugin-autodetect.xml:
|
|
|
|
* docs/plugins/inspect/plugin-avi.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cacasink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cairo.xml:
|
|
|
|
* docs/plugins/inspect/plugin-cutter.xml:
|
|
|
|
* docs/plugins/inspect/plugin-debug.xml:
|
|
|
|
* docs/plugins/inspect/plugin-dv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-efence.xml:
|
|
|
|
* docs/plugins/inspect/plugin-effectv.xml:
|
|
|
|
* docs/plugins/inspect/plugin-esdsink.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flac.xml:
|
|
|
|
* docs/plugins/inspect/plugin-flxdec.xml:
|
|
|
|
* docs/plugins/inspect/plugin-gconfelements.xml:
|
|
|
|
* docs/plugins/inspect/plugin-goom.xml:
|
|
|
|
* docs/plugins/inspect/plugin-jpeg.xml:
|
|
|
|
* docs/plugins/inspect/plugin-level.xml:
|
|
|
|
* docs/plugins/inspect/plugin-matroska.xml:
|
|
|
|
* docs/plugins/inspect/plugin-mulaw.xml:
|
|
|
|
* docs/plugins/inspect/plugin-navigationtest.xml:
|
|
|
|
* docs/plugins/inspect/plugin-ossaudio.xml:
|
|
|
|
* docs/plugins/inspect/plugin-png.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtsp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-shout2send.xml:
|
|
|
|
* docs/plugins/inspect/plugin-smpte.xml:
|
|
|
|
* docs/plugins/inspect/plugin-speex.xml:
|
|
|
|
* docs/plugins/inspect/plugin-udp.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videobox.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videoflip.xml:
|
|
|
|
* docs/plugins/inspect/plugin-videomixer.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavenc.xml:
|
|
|
|
* docs/plugins/inspect/plugin-wavparse.xml:
|
|
|
|
update for HEAD version
|
|
|
|
|
2005-11-25 18:03:24 +00:00
|
|
|
2005-11-25 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_stop):
|
|
|
|
Patch from Sebastien Cote to close control sockets in udpsrc.
|
|
|
|
|
2005-11-24 15:07:06 +00:00
|
|
|
2005-11-24 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* gst/effectv/gstquark.c: (gst_quarktv_set_caps),
|
|
|
|
(gst_quarktv_get_unit_size), (gst_quarktv_transform),
|
|
|
|
(gst_quarktv_planetable_clear), (gst_quarktv_change_state),
|
|
|
|
(gst_quarktv_base_init), (gst_quarktv_class_init),
|
|
|
|
(gst_quarktv_init): Flush the planes list on reverse caps
|
|
|
|
negotiation. This was crashing because of differently sized
|
|
|
|
buffers.
|
|
|
|
|
gst/: Handle strides correctly, fix identity flipping, convert navigation event correctly again.
Original commit message from CVS:
2005-11-24 Julien MOUTTE <julien@moutte.net>
* gst/debug/gstnavigationtest.c: (draw_box_planar411):
* gst/videofilter/gstvideoflip.c:
(gst_videoflip_method_get_type),
(gst_videoflip_set_caps), (gst_videoflip_transform_caps),
(gst_videoflip_get_unit_size), (gst_videoflip_flip),
(gst_videoflip_transform), (gst_videoflip_handle_src_event),
(gst_videoflip_set_property), (gst_videoflip_base_init),
(gst_videoflip_class_init), (gst_videoflip_init): Handle strides
correctly, fix identity flipping, convert navigation event
correctly again.
2005-11-24 12:50:28 +00:00
|
|
|
2005-11-24 Julien MOUTTE <julien@moutte.net>
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* gst/debug/gstnavigationtest.c: (draw_box_planar411):
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* gst/videofilter/gstvideoflip.c: (gst_videoflip_method_get_type),
|
|
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|
(gst_videoflip_set_caps), (gst_videoflip_transform_caps),
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|
(gst_videoflip_get_unit_size), (gst_videoflip_flip),
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(gst_videoflip_transform), (gst_videoflip_handle_src_event),
|
|
|
|
(gst_videoflip_set_property), (gst_videoflip_base_init),
|
|
|
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(gst_videoflip_class_init), (gst_videoflip_init): Handle strides
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|
correctly, fix identity flipping, convert navigation event
|
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|
correctly again.
|
|
|
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|
2005-11-23 20:05:26 +00:00
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2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac: back to HEAD
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2005-11-23 19:56:31 +00:00
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=== release 0.9.6 ===
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2005-11-23 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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releasing 0.9.6, "Everything's Not Lost"
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2005-11-23 16:49:16 +00:00
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2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
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* gst/debug/gstnavigationtest.c: (gst_navigationtest_init):
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Oops, initialise the framerate GValue
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|
VideoFilter inherits from
Original commit message from CVS:
2005-11-23 Julien MOUTTE <julien@moutte.net>
* ext/cairo/gsttimeoverlay.c:
(gst_timeoverlay_update_font_height),
(gst_timeoverlay_set_caps), (gst_timeoverlay_get_unit_size),
(gst_timeoverlay_transform), (gst_timeoverlay_base_init),
(gst_timeoverlay_class_init), (gst_timeoverlay_init),
(gst_timeoverlay_get_type):
* ext/cairo/gsttimeoverlay.h:
* gst/debug/Makefile.am:
* gst/debug/gstnavigationtest.c:
(gst_navigationtest_handle_src_event),
(gst_navigationtest_get_unit_size),
(gst_navigationtest_set_caps),
(gst_navigationtest_transform),
(gst_navigationtest_change_state),
(gst_navigationtest_base_init), (gst_navigationtest_class_init),
(gst_navigationtest_init), (gst_navigationtest_get_type),
(plugin_init):
* gst/debug/gstnavigationtest.h:
* gst/effectv/Makefile.am:
* gst/effectv/gstaging.c: (gst_agingtv_set_caps),
(gst_agingtv_get_unit_size), (gst_agingtv_transform),
(gst_agingtv_base_init), (gst_agingtv_class_init),
(gst_agingtv_init), (gst_agingtv_get_type):
* gst/effectv/gstdice.c: (gst_dicetv_set_caps),
(gst_dicetv_get_unit_size), (gst_dicetv_transform),
(gst_dicetv_base_init), (gst_dicetv_class_init),
(gst_dicetv_init),
(gst_dicetv_get_type):
* gst/effectv/gstedge.c: (gst_edgetv_set_caps),
(gst_edgetv_get_unit_size), (gst_edgetv_transform),
(gst_edgetv_base_init), (gst_edgetv_class_init),
(gst_edgetv_init),
(gst_edgetv_get_type):
* gst/effectv/gsteffectv.c:
* gst/effectv/gsteffectv.h:
* gst/effectv/gstquark.c: (gst_quarktv_set_caps),
(gst_quarktv_get_unit_size), (fastrand),
(gst_quarktv_transform),
(gst_quarktv_change_state), (gst_quarktv_base_init),
(gst_quarktv_class_init), (gst_quarktv_init),
(gst_quarktv_get_type):
* gst/effectv/gstrev.c: (gst_revtv_set_caps),
(gst_revtv_get_unit_size), (gst_revtv_transform),
(gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init),
(gst_revtv_get_type):
* gst/effectv/gstshagadelic.c: (gst_shagadelictv_set_caps),
(gst_shagadelictv_get_unit_size), (gst_shagadelictv_transform),
(gst_shagadelictv_base_init), (gst_shagadelictv_class_init),
(gst_shagadelictv_init), (gst_shagadelictv_get_type):
* gst/effectv/gstvertigo.c: (gst_vertigotv_set_caps),
(gst_vertigotv_get_unit_size), (gst_vertigotv_transform),
(gst_vertigotv_base_init), (gst_vertigotv_class_init),
(gst_vertigotv_init), (gst_vertigotv_get_type):
* gst/effectv/gstwarp.c: (gst_warptv_set_caps),
(gst_warptv_get_unit_size), (gst_warptv_transform),
(gst_warptv_base_init), (gst_warptv_class_init),
(gst_warptv_init),
(gst_warptv_get_type):
* gst/videofilter/Makefile.am:
* gst/videofilter/gstvideobalance.c:
* gst/videofilter/gstvideobalance.h:
* gst/videofilter/gstvideofilter.c: (gst_videofilter_get_type),
(gst_videofilter_class_init), (gst_videofilter_init):
* gst/videofilter/gstvideofilter.h:
* gst/videofilter/gstvideoflip.c: (gst_videoflip_set_caps),
(gst_videoflip_transform_caps), (gst_videoflip_get_unit_size),
(gst_videoflip_flip), (gst_videoflip_transform),
(gst_videoflip_handle_src_event), (gst_videoflip_set_property),
(gst_videoflip_base_init), (gst_videoflip_class_init),
(gst_videoflip_init), (plugin_init), (gst_videoflip_get_type):
* gst/videofilter/gstvideoflip.h: VideoFilter inherits from
BaseTransform, it's just a place holder for now and every video
effect plugin has been ported to use BaseTransform features
directly. QuarkTV was fixed too (was broken), navigationtest
works
and best for the end, videoflip converts navigation events
depending
on flip method ! Fixes #320953
2005-11-23 15:50:51 +00:00
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|
|
2005-11-23 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height),
|
|
|
|
(gst_timeoverlay_set_caps), (gst_timeoverlay_get_unit_size),
|
|
|
|
(gst_timeoverlay_transform), (gst_timeoverlay_base_init),
|
|
|
|
(gst_timeoverlay_class_init), (gst_timeoverlay_init),
|
|
|
|
(gst_timeoverlay_get_type):
|
|
|
|
* ext/cairo/gsttimeoverlay.h:
|
|
|
|
* gst/debug/Makefile.am:
|
|
|
|
* gst/debug/gstnavigationtest.c:
|
|
|
|
(gst_navigationtest_handle_src_event),
|
|
|
|
(gst_navigationtest_get_unit_size), (gst_navigationtest_set_caps),
|
|
|
|
(gst_navigationtest_transform), (gst_navigationtest_change_state),
|
|
|
|
(gst_navigationtest_base_init), (gst_navigationtest_class_init),
|
|
|
|
(gst_navigationtest_init), (gst_navigationtest_get_type),
|
|
|
|
(plugin_init):
|
|
|
|
* gst/debug/gstnavigationtest.h:
|
|
|
|
* gst/effectv/Makefile.am:
|
|
|
|
* gst/effectv/gstaging.c: (gst_agingtv_set_caps),
|
|
|
|
(gst_agingtv_get_unit_size), (gst_agingtv_transform),
|
|
|
|
(gst_agingtv_base_init), (gst_agingtv_class_init),
|
|
|
|
(gst_agingtv_init), (gst_agingtv_get_type):
|
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|
|
* gst/effectv/gstdice.c: (gst_dicetv_set_caps),
|
|
|
|
(gst_dicetv_get_unit_size), (gst_dicetv_transform),
|
|
|
|
(gst_dicetv_base_init), (gst_dicetv_class_init), (gst_dicetv_init),
|
|
|
|
(gst_dicetv_get_type):
|
|
|
|
* gst/effectv/gstedge.c: (gst_edgetv_set_caps),
|
|
|
|
(gst_edgetv_get_unit_size), (gst_edgetv_transform),
|
|
|
|
(gst_edgetv_base_init), (gst_edgetv_class_init), (gst_edgetv_init),
|
|
|
|
(gst_edgetv_get_type):
|
|
|
|
* gst/effectv/gsteffectv.c:
|
|
|
|
* gst/effectv/gsteffectv.h:
|
|
|
|
* gst/effectv/gstquark.c: (gst_quarktv_set_caps),
|
|
|
|
(gst_quarktv_get_unit_size), (fastrand), (gst_quarktv_transform),
|
|
|
|
(gst_quarktv_change_state), (gst_quarktv_base_init),
|
|
|
|
(gst_quarktv_class_init), (gst_quarktv_init),
|
|
|
|
(gst_quarktv_get_type):
|
|
|
|
* gst/effectv/gstrev.c: (gst_revtv_set_caps),
|
|
|
|
(gst_revtv_get_unit_size), (gst_revtv_transform),
|
|
|
|
(gst_revtv_base_init), (gst_revtv_class_init), (gst_revtv_init),
|
|
|
|
(gst_revtv_get_type):
|
|
|
|
* gst/effectv/gstshagadelic.c: (gst_shagadelictv_set_caps),
|
|
|
|
(gst_shagadelictv_get_unit_size), (gst_shagadelictv_transform),
|
|
|
|
(gst_shagadelictv_base_init), (gst_shagadelictv_class_init),
|
|
|
|
(gst_shagadelictv_init), (gst_shagadelictv_get_type):
|
|
|
|
* gst/effectv/gstvertigo.c: (gst_vertigotv_set_caps),
|
|
|
|
(gst_vertigotv_get_unit_size), (gst_vertigotv_transform),
|
|
|
|
(gst_vertigotv_base_init), (gst_vertigotv_class_init),
|
|
|
|
(gst_vertigotv_init), (gst_vertigotv_get_type):
|
|
|
|
* gst/effectv/gstwarp.c: (gst_warptv_set_caps),
|
|
|
|
(gst_warptv_get_unit_size), (gst_warptv_transform),
|
|
|
|
(gst_warptv_base_init), (gst_warptv_class_init), (gst_warptv_init),
|
|
|
|
(gst_warptv_get_type):
|
|
|
|
* gst/videofilter/Makefile.am:
|
|
|
|
* gst/videofilter/gstvideobalance.c:
|
|
|
|
* gst/videofilter/gstvideobalance.h:
|
|
|
|
* gst/videofilter/gstvideofilter.c: (gst_videofilter_get_type),
|
|
|
|
(gst_videofilter_class_init), (gst_videofilter_init):
|
|
|
|
* gst/videofilter/gstvideofilter.h:
|
|
|
|
* gst/videofilter/gstvideoflip.c: (gst_videoflip_set_caps),
|
|
|
|
(gst_videoflip_transform_caps), (gst_videoflip_get_unit_size),
|
|
|
|
(gst_videoflip_flip), (gst_videoflip_transform),
|
|
|
|
(gst_videoflip_handle_src_event), (gst_videoflip_set_property),
|
|
|
|
(gst_videoflip_base_init), (gst_videoflip_class_init),
|
|
|
|
(gst_videoflip_init), (plugin_init), (gst_videoflip_get_type):
|
|
|
|
* gst/videofilter/gstvideoflip.h: VideoFilter inherits from
|
|
|
|
BaseTransform, it's just a place holder for now and every video
|
|
|
|
effect plugin has been ported to use BaseTransform features
|
|
|
|
directly. QuarkTV was fixed too (was broken), navigationtest works
|
|
|
|
and best for the end, videoflip converts navigation events depending
|
|
|
|
on flip method ! Fixes #320953
|
|
|
|
|
2005-11-23 14:22:18 +00:00
|
|
|
2005-11-23 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.c: (gst_aasink_fixate):
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_collected):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps),
|
|
|
|
(gst_goom_src_negotiate), (gst_goom_chain):
|
|
|
|
* gst/matroska/matroska-mux.c:
|
|
|
|
(gst_matroska_mux_video_pad_setcaps):
|
|
|
|
* sys/osxvideo/osxvideosink.m:
|
|
|
|
Fixes for API changes
|
|
|
|
|
2005-11-22 23:58:14 +00:00
|
|
|
2005-11-23 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_setcaps),
|
|
|
|
(gst_jpeg_dec_chain), (gst_jpeg_dec_change_state):
|
|
|
|
* ext/jpeg/gstjpegdec.h:
|
|
|
|
* ext/jpeg/gstjpegenc.c: (gst_jpegenc_setcaps):
|
|
|
|
* ext/jpeg/gstjpegenc.h:
|
|
|
|
* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_setcaps),
|
|
|
|
(gst_smokeenc_resync):
|
|
|
|
* ext/jpeg/gstsmokeenc.h:
|
|
|
|
JPEG fractiony goodness.
|
|
|
|
|
2005-11-22 22:35:57 +00:00
|
|
|
2005-11-22 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/goom/filters.c:
|
|
|
|
* gst/goom/graphic.h:
|
|
|
|
Fix compilation by making some functions static inline instead of
|
|
|
|
extern inline, matching the way they're used.
|
|
|
|
|
2005-11-22 22:21:37 +00:00
|
|
|
2005-11-22 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_init),
|
|
|
|
(gst_text_overlay_setcaps), (gst_text_overlay_collected):
|
|
|
|
* ext/cairo/gsttextoverlay.h:
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_sink_link):
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.h:
|
|
|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_init),
|
|
|
|
(gst_pngdec_caps_create_and_set):
|
|
|
|
* ext/libpng/gstpngdec.h:
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_setcaps):
|
|
|
|
* gst/alpha/gstalphacolor.c: (gst_alpha_color_set_caps):
|
|
|
|
* gst/avi/gstavimux.c: (gst_avimux_init),
|
|
|
|
(gst_avimux_vidsinkconnect):
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_chain):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_init), (gst_goom_src_setcaps),
|
|
|
|
(gst_goom_src_negotiate), (gst_goom_chain):
|
|
|
|
* gst/goom/gstgoom.h:
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
|
|
|
|
* gst/matroska/matroska-mux.c:
|
|
|
|
(gst_matroska_mux_video_pad_setcaps):
|
|
|
|
* sys/osxvideo/osxvideosink.h:
|
|
|
|
* sys/osxvideo/osxvideosink.m:
|
|
|
|
More fractional framerate conversions
|
|
|
|
|
2005-11-22 20:07:47 +00:00
|
|
|
2005-11-22 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.c: (gst_aasink_fixate):
|
|
|
|
* gst/debug/gstnavigationtest.c:
|
|
|
|
(gst_navigationtest_handle_src_event):
|
|
|
|
* gst/videofilter/gstvideofilter.c:
|
|
|
|
(gst_videofilter_format_get_structure), (gst_videofilter_setcaps),
|
|
|
|
(gst_videofilter_init):
|
|
|
|
* gst/videofilter/gstvideofilter.h:
|
|
|
|
Convert to fractional framerates.
|
|
|
|
|
2005-11-22 18:11:58 +00:00
|
|
|
2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.c:
|
|
|
|
* ext/dv/gstdvdec.c:
|
|
|
|
* ext/esd/esdmon.c:
|
|
|
|
* ext/flac/gstflacenc.c:
|
|
|
|
* ext/gdk_pixbuf/pixbufscale.c:
|
|
|
|
* ext/libcaca/gstcacasink.c:
|
|
|
|
* ext/shout2/gstshout2.c:
|
|
|
|
* gst/alpha/gstalpha.c:
|
|
|
|
* gst/oldcore/gstaggregator.c:
|
|
|
|
* gst/oldcore/gstshaper.c:
|
|
|
|
* gst/smpte/barboxwipes.c:
|
|
|
|
* gst/smpte/gstsmpte.c:
|
|
|
|
* gst/videobox/gstvideobox.c:
|
|
|
|
* gst/videofilter/gstvideoflip.c:
|
|
|
|
* gst/videomixer/videomixer.c:
|
|
|
|
fix up more enums
|
|
|
|
|
2005-11-22 17:39:11 +00:00
|
|
|
2005-11-22 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_pad_sink_setcaps),
|
|
|
|
(gst_videomixer_getcaps), (gst_videomixer_fill_queues),
|
|
|
|
(gst_videomixer_update_queues):
|
|
|
|
Fractional framerates, videomixer.
|
|
|
|
|
2005-11-22 17:09:36 +00:00
|
|
|
2005-11-22 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* ext/dv/gstdvdec.c: (gst_dvdec_init), (gst_dvdec_sink_setcaps):
|
|
|
|
* ext/dv/gstdvdec.h:
|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_init),
|
|
|
|
(gst_dvdemux_src_convert), (gst_dvdemux_sink_convert),
|
|
|
|
(gst_dvdemux_demux_video), (gst_dvdemux_demux_frame),
|
|
|
|
(gst_dvdemux_flush):
|
|
|
|
* ext/dv/gstdvdemux.h:
|
|
|
|
Fractional framerates for DV.
|
|
|
|
|
2005-11-22 14:44:26 +00:00
|
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2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
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* gst/autodetect/gstautoaudiosink.c:
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(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
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* gst/autodetect/gstautovideosink.c:
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|
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(gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
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Use gst_plugin_feature_list_free() to free feature list and
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in the case of autovideosink free the list at all. Also
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miscellaneous cosmetic fixes.
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2005-11-22 13:13:21 +00:00
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2005-11-22 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/cutter/gstcutter.c: (gst_cutter_chain),
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(gst_cutter_set_property), (gst_cutter_get_caps):
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copy calculation code from level; remove use of some audio
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|
functions
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2005-11-22 12:38:33 +00:00
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2005-11-22 Andy Wingo <wingo@pobox.com>
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|
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* Update for gst_tag_setter API changes.
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2005-11-22 11:53:34 +00:00
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2005-11-22 Andy Wingo <wingo@pobox.com>
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* ext/dv/gstdvdemux.c (gst_dvdemux_handle_sink_event)
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|
(gst_dvdemux_demux_frame)
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* ext/flac/gstflacdec.c (gst_flacdec_write)
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* ext/flac/gstflacenc.c (gst_flacenc_seek_callback)
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(gst_flacenc_sink_event)
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* ext/gconf/gstgconfaudiosink.c (gst_gconf_audio_sink_init)
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* ext/gconf/gstgconfvideosink.c (gst_gconf_video_sink_init)
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* ext/libpng/gstpngdec.c (gst_pngdec_caps_create_and_set)
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* ext/speex/gstspeexdec.c (speex_dec_event, speex_dec_chain)
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* gst/auparse/gstauparse.c (gst_auparse_chain)
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* gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_init)
|
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* gst/autodetect/gstautovideosink.c (gst_auto_video_sink_init)
|
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* gst/avi/gstavidemux.c (gst_avi_demux_stream_header)
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(gst_avi_demux_handle_seek)
|
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* gst/goom/gstgoom.c (gst_goom_event)
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* gst/matroska/ebml-write.c (gst_ebml_write_seek)
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* gst/matroska/matroska-demux.c
|
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|
(gst_matroska_demux_handle_seek_event)
|
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(gst_matroska_demux_loop_stream_parse_id)
|
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|
* gst/wavenc/gstwavenc.c (gst_wavenc_stop_file)
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* gst/wavparse/gstwavparse.c (gst_wavparse_handle_seek)
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(gst_wavparse_stream_headers): Run update-funcnames.
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|
2005-11-22 11:49:30 +00:00
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|
2005-11-22 Edward Hervey <edward@fluendo.com>
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* ext/raw1394/gstdv1394src.c: (gst_dv1394src_class_init),
|
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|
(gst_dv1394src_init), (gst_dv1394src_dispose),
|
|
|
|
(gst_dv1394src_set_property), (gst_dv1394src_discover_avc_node),
|
|
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|
(gst_dv1394src_uri_set_uri):
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
|
|
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|
(gst_udpsrc_update_uri), (gst_udpsrc_set_uri),
|
|
|
|
(gst_udpsrc_set_property), (gst_udpsrc_uri_get_uri):
|
|
|
|
URIHandler interface and element properties are now properly
|
|
|
|
synchronized for DV1394src and UDPSrc
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|
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|
2005-11-22 11:36:04 +00:00
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|
|
2005-11-22 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
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|
|
* ext/flac/Makefile.am:
|
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|
* ext/speex/Makefile.am:
|
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|
|
libgsttagedit has been renamed to libgsttag.
|
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|
|
2005-11-21 20:11:59 +00:00
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|
2005-11-21 Wim Taymans <wim@fluendo.com>
|
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|
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|
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|
|
* gst/rtsp/rtspconnection.c: (read_body):
|
|
|
|
Apply patch from Sebastien Cote to fix #319184.
|
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|
|
|
2005-11-21 19:50:25 +00:00
|
|
|
2005-11-21 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
* gst/cutter/Makefile.am:
|
|
|
|
* gst/cutter/gstcutter.c: (gst_cutter_class_init),
|
|
|
|
(gst_cutter_init), (gst_cutter_message_new), (gst_cutter_chain),
|
|
|
|
(gst_cutter_set_property), (gst_cutter_get_property),
|
|
|
|
(plugin_init), (gst_cutter_get_caps):
|
|
|
|
port cutter
|
|
|
|
* gst/level/gstlevel.c:
|
|
|
|
fix up plugin details
|
|
|
|
|
2005-11-21 18:09:02 +00:00
|
|
|
2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
|
|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event):
|
|
|
|
* ext/flac/gstflacdec.c: (gst_flacdec_loop),
|
|
|
|
(gst_flacdec_src_event):
|
|
|
|
* ext/flac/gstflacenc.c: (gst_flacenc_sink_event):
|
|
|
|
* ext/ladspa/gstsignalprocessor.c: (gst_signal_processor_event),
|
|
|
|
(gst_signal_processor_getrange), (gst_signal_processor_chain):
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_src_event_handler),
|
|
|
|
(gst_flxdec_sink_event_handler):
|
|
|
|
* gst/matroska/matroska-demux.c:
|
|
|
|
(gst_matroska_demux_handle_seek_event):
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek):
|
|
|
|
Update for stream lock API changes: don't take stream log
|
|
|
|
in sink event handlers any longer and change GST_STREAM_LOCK
|
|
|
|
to GST_PAD_STREAM_LOCK. Don't leak references in flxdec event
|
|
|
|
functions.
|
|
|
|
|
2005-11-21 17:49:21 +00:00
|
|
|
2005-11-21 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_dispose):
|
|
|
|
gst_object_unref, not g_object_unref
|
|
|
|
|
2005-11-21 17:23:46 +00:00
|
|
|
2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_init),
|
|
|
|
(gst_wavparse_create_sourcepad), (gst_wavparse_sink_activate):
|
|
|
|
Use GST_DEBUG_FUNCPTR; add debug message in pad activate function.
|
|
|
|
|
2005-11-21 17:18:01 +00:00
|
|
|
2005-11-21 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/auparse/Makefile.am:
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_class_init),
|
|
|
|
(gst_auparse_init), (gst_auparse_dispose), (gst_auparse_chain),
|
|
|
|
(gst_auparse_change_state):
|
|
|
|
* gst/auparse/gstauparse.h:
|
|
|
|
Partially fix #161712. playbin still doesn't work on these files,
|
|
|
|
(on the bug report, Andy says we aren't typefinding it for some
|
|
|
|
reason?) but at least auparse isn't totally busted like it was before.
|
|
|
|
|
2005-11-21 16:36:05 +00:00
|
|
|
2005-11-21 Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* *.h:
|
|
|
|
* *.c: Ran scripts/update-macros. Oh yes.
|
|
|
|
|
2005-11-21 15:06:35 +00:00
|
|
|
2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_sync_streams):
|
|
|
|
Filler events are gone for now, comment out section generating
|
|
|
|
them.
|
|
|
|
|
2005-11-21 14:31:05 +00:00
|
|
|
2005-11-21 Jan Schmidt <thaytan@mad.scientist.com>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.c: (gst_aasink_fixate):
|
|
|
|
* ext/mikmod/gstmikmod.c: (gst_mikmod_srcfixate):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_src_negotiate):
|
|
|
|
* sys/osxvideo/osxvideosink.m:
|
2005-11-23 19:56:31 +00:00
|
|
|
Rename gst_caps_structure_fixate_* to gst_structure_fixate_*
|
|
|
|
(#322027)
|
2005-11-21 14:31:05 +00:00
|
|
|
|
2005-11-21 13:38:24 +00:00
|
|
|
2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/aalib/gstaasink.c: (gst_aasink_setcaps):
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_file_header),
|
|
|
|
(gst_avi_demux_read_subindexes), (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_parse_odml), (gst_avi_demux_stream_index),
|
|
|
|
(gst_avi_demux_sync), (gst_avi_demux_stream_header),
|
|
|
|
(gst_avi_demux_stream_data):
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
|
|
|
|
* gst/wavenc/gstwavenc.c: (write_metadata):
|
|
|
|
* gst/wavparse/gstwavparse.c: (gst_wavparse_parse_adtl),
|
|
|
|
(gst_wavparse_parse_file_header), (gst_wavparse_stream_headers):
|
|
|
|
Fixes for GST_FOURCC_FORMAT API change.
|
|
|
|
|
Fix for collect pads API change. Also fix textoverlay state change function.
Original commit message from CVS:
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_finalize),
(gst_text_overlay_init), (gst_text_overlay_text_pad_linked),
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_pop_video),
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
(gst_text_overlay_change_state):
* gst/matroska/matroska-mux.c: (gst_matroska_mux_init),
(gst_matroska_mux_reset), (gst_matroska_mux_request_new_pad),
(gst_matroska_mux_best_pad), (gst_matroska_mux_change_state):
* gst/smpte/gstsmpte.c: (gst_smpte_init), (gst_smpte_collected):
* gst/videomixer/videomixer.c: (gst_videomixer_init),
(gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
(gst_videomixer_change_state):
Fix for collect pads API change. Also fix textoverlay state
change function.
2005-11-21 12:13:48 +00:00
|
|
|
2005-11-21 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_finalize),
|
|
|
|
(gst_text_overlay_init), (gst_text_overlay_text_pad_linked),
|
|
|
|
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_pop_video),
|
|
|
|
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
|
|
|
|
(gst_text_overlay_change_state):
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_init),
|
|
|
|
(gst_matroska_mux_reset), (gst_matroska_mux_request_new_pad),
|
|
|
|
(gst_matroska_mux_best_pad), (gst_matroska_mux_change_state):
|
|
|
|
* gst/smpte/gstsmpte.c: (gst_smpte_init), (gst_smpte_collected):
|
|
|
|
* gst/videomixer/videomixer.c: (gst_videomixer_init),
|
|
|
|
(gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
|
|
|
|
(gst_videomixer_change_state):
|
|
|
|
Fix for collect pads API change. Also fix textoverlay state
|
|
|
|
change function.
|
|
|
|
|
2005-11-20 17:04:55 +00:00
|
|
|
2005-11-20 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Replace
|
|
|
|
GST_PAD_IS_USABLE by something approaching it.
|
|
|
|
|
2005-11-20 16:43:32 +00:00
|
|
|
2005-11-20 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_start): Fix for
|
|
|
|
API changes.
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_chain): Fix for API changes,
|
|
|
|
but also fix the code that was not checking return values from
|
|
|
|
pad_push neither using pad_alloc_buffer.
|
|
|
|
|
2005-11-18 18:19:21 +00:00
|
|
|
2005-11-18 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngenc.c: (gst_pngenc_class_init),
|
|
|
|
(gst_pngenc_chain):
|
|
|
|
Added debug category
|
|
|
|
Return GST_FLOW_UNEXPECTED when sending an EOS, so the whole pipeline
|
|
|
|
goes to EOS.
|
|
|
|
|
2005-11-17 18:23:23 +00:00
|
|
|
2005-11-17 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am
|
|
|
|
* gst/rtp/gstrtp.c
|
|
|
|
* gst/rtp/gstrtpg711enc.c: (gst_rtpg711enc_src_template),
|
|
|
|
(gst_rtpg711enc_class_init), (gst_rtpg711enc_init),
|
|
|
|
(gst_rtpg711enc_finalize), (gst_rtpg711enc_setcaps),
|
|
|
|
(gst_rtpg711enc_flush), (gst_rtpg711enc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpg711enc.h:
|
|
|
|
* gst/rtp/gstrtpg711dec.c: (gst_rtpg711dec_sink_template):
|
|
|
|
* gst/rtp/gstrtpspeexenc.c:
|
|
|
|
* gst/rtp/gstrtpspeexenc.h:
|
|
|
|
* gst/rtp/gstrtpspeexdec.c:
|
|
|
|
* gst/rtp/gstrtpspeexdec.h:
|
|
|
|
Created Speex payloader and depayloader; Optimize G711 payloader to
|
|
|
|
use adapter and send packets until MTU size.
|
|
|
|
|
2005-11-16 19:08:54 +00:00
|
|
|
2005-11-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad):
|
|
|
|
Fix leak in check.
|
|
|
|
|
2005-11-16 17:00:32 +00:00
|
|
|
2005-11-16 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/flx/gstflxdec.c: (gst_flxdec_change_state):
|
|
|
|
Fix state change.
|
|
|
|
|
2005-11-16 10:43:44 +00:00
|
|
|
2005-11-16 Andy Wingo <wingo@pobox.com>
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: Clean up with the boilerplate macro.
|
2005-11-16 11:02:24 +00:00
|
|
|
(gst_udpsrc_create): Move comment.
|
2005-11-16 10:43:44 +00:00
|
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|
|
2005-11-15 19:41:21 +00:00
|
|
|
2005-11-15 Vincent Torri <torri at iecn dot u-nancy dot fr>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroskademux_do_index_seek):
|
|
|
|
When seeking, seek to closest index entry at or before the requested
|
|
|
|
seek position, not just the closest one (#321001).
|
|
|
|
|
2005-11-15 12:16:00 +00:00
|
|
|
2005-11-15 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/avi/gstavidemux.c: (swap_line), (gst_avi_demux_invert),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data):
|
|
|
|
Invert DIB images again (see #132341).
|
|
|
|
|
2005-11-14 02:13:35 +00:00
|
|
|
2005-11-14 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* configure.ac:
|
|
|
|
rework configure.ac file
|
|
|
|
* ext/aalib/gstaasink.c:
|
|
|
|
* ext/cairo/gstcairo.c:
|
|
|
|
* ext/dv/gstdv.c:
|
|
|
|
* ext/esd/gstesd.c:
|
|
|
|
* ext/flac/gstflac.c:
|
|
|
|
* ext/gconf/gstgconfelements.c:
|
|
|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c:
|
|
|
|
* ext/jpeg/gstjpeg.c:
|
|
|
|
* ext/ladspa/gstladspa.c:
|
|
|
|
* ext/libcaca/gstcacasink.c:
|
|
|
|
* ext/libmng/gstmng.c:
|
|
|
|
* ext/libpng/gstpng.c:
|
|
|
|
* ext/mikmod/gstmikmod.c:
|
|
|
|
* ext/pango/gsttextoverlay.c:
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|
|
|
* ext/pango/gsttimeoverlay.c:
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|
|
|
* ext/raw1394/gst1394.c:
|
|
|
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* ext/speex/gstspeex.c:
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|
|
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* gst/alpha/Makefile.am:
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* gst/alpha/gstalpha.c:
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* gst/alpha/gstalphacolor.c:
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* gst/auparse/gstauparse.c:
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* gst/autodetect/gstautoaudiosink.c:
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(gst_auto_audio_sink_factory_filter),
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(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
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* gst/autodetect/gstautodetect.c:
|
|
|
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* gst/avi/gstavi.c:
|
|
|
|
* gst/cutter/gstcutter.c:
|
|
|
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* gst/debug/efence.c:
|
|
|
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* gst/debug/gstdebug.c:
|
|
|
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* gst/debug/gstnavigationtest.c:
|
|
|
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* gst/effectv/gsteffectv.c:
|
|
|
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* gst/flx/gstflxdec.c:
|
|
|
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* gst/goom/gstgoom.c:
|
|
|
|
* gst/interleave/plugin.c:
|
|
|
|
* gst/law/alaw.c:
|
|
|
|
* gst/law/mulaw.c:
|
|
|
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* gst/level/gstlevel.c:
|
|
|
|
* gst/matroska/ebml-write.c: (gst_ebml_write_seek):
|
|
|
|
* gst/matroska/matroska.c:
|
|
|
|
* gst/median/gstmedian.c:
|
|
|
|
* gst/monoscope/gstmonoscope.c:
|
|
|
|
* gst/multipart/multipart.c:
|
|
|
|
* gst/oldcore/gstelements.c:
|
|
|
|
* gst/rtp/gstrtp.c:
|
|
|
|
* gst/rtsp/gstrtsp.c:
|
|
|
|
* gst/smoothwave/gstsmoothwave.c:
|
|
|
|
* gst/smpte/gstsmpte.c:
|
|
|
|
* gst/udp/gstudp.c:
|
|
|
|
* gst/videobox/gstvideobox.c:
|
|
|
|
* gst/videofilter/gstgamma.c:
|
|
|
|
* gst/videofilter/gstvideobalance.c:
|
|
|
|
* gst/videofilter/gstvideoflip.c:
|
|
|
|
* gst/videofilter/gstvideotemplate.c:
|
|
|
|
* gst/videomixer/videomixer.c:
|
|
|
|
* gst/wavenc/gstwavenc.c:
|
|
|
|
* gst/wavparse/gstwavparse.c:
|
|
|
|
* sys/oss/gstossaudio.c:
|
|
|
|
* sys/osxaudio/gstosxaudio.c:
|
|
|
|
update GST_PLUGIN_DEFINE
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/gstasteriskh263.c:
|
|
|
|
check for htons/htonl headers and possibly link to winsock2 lib
|
|
|
|
|
2005-11-12 13:31:56 +00:00
|
|
|
2005-11-12 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_chain):
|
|
|
|
Only GST_DEBUG() information on the valid components.
|
|
|
|
|
2005-11-11 19:34:50 +00:00
|
|
|
2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
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|
|
|
|
|
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* configure.ac: back to HEAD
|
|
|
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|
2005-11-11 19:33:23 +00:00
|
|
|
=== release 0.9.5 ===
|
|
|
|
|
|
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|
2005-11-11 Thomas Vander Stichele <thomas at apestaart dot org>
|
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|
|
|
|
|
|
* configure.ac:
|
|
|
|
releasing 0.9.5, "Phone Alarm"
|
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|
|
|
2005-11-11 16:48:58 +00:00
|
|
|
2005-11-11 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
|
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* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
|
|
|
|
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_event),
|
|
|
|
(gst_avi_demux_stream_header), (gst_avi_demux_handle_seek),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
|
|
|
|
(gst_avi_demux_loop):
|
|
|
|
* gst/avi/gstavidemux.h:
|
|
|
|
Yeah, implement proper seeking. Exact seeking and segment seeking.
|
|
|
|
Still need to do some checks for segment_stop.
|
|
|
|
|
2005-11-10 12:22:30 +00:00
|
|
|
2005-11-10 Josef Zlomek <josef dot zlomek at xeris dot cz>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
|
|
|
|
Don't try to ready buffer duration from buffer that we don't
|
|
|
|
own any longer and that might already have been unreffed.
|
|
|
|
(#321136)
|
|
|
|
|
2005-11-09 21:35:29 +00:00
|
|
|
2005-11-09 Zeeshan Ali <zeenix@gmail.com>
|
|
|
|
|
|
|
|
* gst/flx/gstflxdec.c: (flx_decode_delta_fli),
|
|
|
|
(flx_decode_delta_flc), (gst_flxdec_chain):
|
|
|
|
Attempting to optimize the code for embedded systems.
|
|
|
|
|
2005-11-08 08:54:30 +00:00
|
|
|
2005-11-08 Alessandro Decina <alessandro at nnva dot org>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/oss/gstosssink.c: (gst_oss_sink_close):
|
|
|
|
Don't re-use already closed file descriptor. (#320920)
|
|
|
|
|
2005-11-07 17:35:20 +00:00
|
|
|
2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* sys/oss/gstosssink.c: (gst_oss_sink_dispose),
|
|
|
|
(gst_oss_sink_set_property), (gst_oss_sink_getcaps),
|
|
|
|
(gst_oss_sink_prepare):
|
|
|
|
* sys/oss/gstosssink.h:
|
|
|
|
Cache probed caps; fix debug output for SET_PARAM macros.
|
|
|
|
|
ext/cairo/: Port cairo textoverlay plugin to 0.9. Add 'shaded-background' property and redo position. Doesn't handle ...
Original commit message from CVS:
* ext/cairo/Makefile.am:
* ext/cairo/gstcairo.c: (plugin_init):
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_class_init), (gst_text_overlay_finalize),
(gst_text_overlay_init), (gst_text_overlay_font_init),
(gst_text_overlay_set_property), (gst_text_overlay_render_text),
(gst_text_overlay_getcaps), (gst_text_overlay_setcaps),
(gst_text_overlay_text_pad_linked),
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y),
(gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
(gst_text_overlay_change_state):
* ext/cairo/gsttextoverlay.h:
Port cairo textoverlay plugin to 0.9. Add 'shaded-background'
property and redo position. Doesn't handle upstream renegotiation
yet though.
2005-11-07 15:09:54 +00:00
|
|
|
2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* ext/cairo/Makefile.am:
|
|
|
|
* ext/cairo/gstcairo.c: (plugin_init):
|
|
|
|
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init),
|
|
|
|
(gst_text_overlay_class_init), (gst_text_overlay_finalize),
|
|
|
|
(gst_text_overlay_init), (gst_text_overlay_font_init),
|
|
|
|
(gst_text_overlay_set_property), (gst_text_overlay_render_text),
|
|
|
|
(gst_text_overlay_getcaps), (gst_text_overlay_setcaps),
|
|
|
|
(gst_text_overlay_text_pad_linked),
|
|
|
|
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y),
|
|
|
|
(gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
|
|
|
|
(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
|
|
|
|
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
|
|
|
|
(gst_text_overlay_change_state):
|
|
|
|
* ext/cairo/gsttextoverlay.h:
|
|
|
|
Port cairo textoverlay plugin to 0.9. Add 'shaded-background'
|
2005-11-07 17:35:20 +00:00
|
|
|
property and redo positioning. Doesn't handle upstream renegotiation
|
ext/cairo/: Port cairo textoverlay plugin to 0.9. Add 'shaded-background' property and redo position. Doesn't handle ...
Original commit message from CVS:
* ext/cairo/Makefile.am:
* ext/cairo/gstcairo.c: (plugin_init):
* ext/cairo/gsttextoverlay.c: (gst_text_overlay_base_init),
(gst_text_overlay_class_init), (gst_text_overlay_finalize),
(gst_text_overlay_init), (gst_text_overlay_font_init),
(gst_text_overlay_set_property), (gst_text_overlay_render_text),
(gst_text_overlay_getcaps), (gst_text_overlay_setcaps),
(gst_text_overlay_text_pad_linked),
(gst_text_overlay_text_pad_unlinked), (gst_text_overlay_shade_y),
(gst_text_overlay_blit_1), (gst_text_overlay_blit_sub2x2),
(gst_text_overlay_push_frame), (gst_text_overlay_pop_video),
(gst_text_overlay_pop_text), (gst_text_overlay_collected),
(gst_text_overlay_change_state):
* ext/cairo/gsttextoverlay.h:
Port cairo textoverlay plugin to 0.9. Add 'shaded-background'
property and redo position. Doesn't handle upstream renegotiation
yet though.
2005-11-07 15:09:54 +00:00
|
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|
yet though.
|
|
|
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|
2005-11-07 10:31:32 +00:00
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|
2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
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|
* gst/avi/gstavidemux.c: (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_process_next_entry), (gst_avi_demux_stream_data),
|
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|
|
(gst_avi_demux_loop):
|
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|
|
No need to take the STREAM_LOCK in the loop function. Improve
|
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|
|
some debug messages. Don't leak pad names in debug messages.
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|
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2005-11-07 10:27:00 +00:00
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|
2005-11-07 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c:
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|
|
|
(gst_matroska_demux_push_vorbis_codec_priv_data),
|
|
|
|
(gst_matroska_demux_add_wvpk_header):
|
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|
Don't error out when the source pad isn't linked.
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|
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|
2005-11-02 19:42:38 +00:00
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2005-11-02 Tim-Philipp Müller <tim at centricular dot net>
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* ext/gconf/gstgconfaudiosink.c: (do_toggle_element),
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(gst_gconf_audio_sink_change_state):
|
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|
* ext/gconf/gstgconfvideosink.c: (do_toggle_element),
|
|
|
|
(gst_gconf_video_sink_change_state):
|
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|
|
Fix state change functions here as well and set kid
|
|
|
|
to NULL state before removing it.
|
|
|
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|
2005-11-01 15:15:44 +00:00
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|
2005-11-01 Edward Hervey <edward@fluendo.com>
|
|
|
|
|
|
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* gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file),
|
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|
(gst_wavenc_init), (gst_wavenc_event), (gst_wavenc_chain):
|
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|
|
Added proper event handlind,
|
|
|
|
made downstream newsegment event use GST_FORMAT_BYTES (otherwise it's
|
|
|
|
ignored),
|
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|
|
and don't set a duration of 0 for buffers otherwise they are discarded
|
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|
|
by GstBaseSink.
|
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|
|
|
|
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|
GstWavEnc needs some serious loving, after going through the code I'm
|
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|
|
really wondering how this can stay in -good ...
|
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2005-11-01 15:11:16 +00:00
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|
2005-11-01 Thomas Vander Stichele <thomas at apestaart dot org>
|
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|
|
|
|
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|
* check/elements/matroskamux.c: (setup_src_pad), (setup_sink_pad),
|
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|
|
(setup_matroskamux), (check_buffer_data), (GST_START_TEST):
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_finalize),
|
|
|
|
(gst_matroska_mux_reset), (gst_matroska_mux_audio_pad_setcaps),
|
|
|
|
(gst_matroska_mux_start), (gst_matroska_mux_write_data),
|
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|
|
(gst_matroska_mux_collected):
|
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|
Fix leaks and invalid memory access as reported by valgrind
|
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|
|
|
2005-11-01 14:36:02 +00:00
|
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|
2005-11-01 Thomas Vander Stichele <thomas at apestaart dot org>
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Patch by: Michal Benes <michal.benes@xeris.cz>
|
|
|
|
|
|
|
|
* check/Makefile.am:
|
|
|
|
* gst/matroska/ebml-write.c: (gst_ebml_write_seek):
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_handle_src_event),
|
|
|
|
(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
|
|
|
|
add a unit test for matroskamux
|
|
|
|
fix the bugs that the unit test exposed
|
|
|
|
|
2005-11-01 12:39:16 +00:00
|
|
|
2005-11-01 Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/autodetect/gstautoaudiosink.c:
|
|
|
|
(gst_auto_audio_sink_class_init),
|
|
|
|
(gst_auto_audio_sink_change_state):
|
|
|
|
* gst/autodetect/gstautovideosink.c:
|
|
|
|
(gst_auto_video_sink_class_init),
|
|
|
|
(gst_auto_video_sink_change_state):
|
|
|
|
Fix state change function and use GST_DEBUG_FUNCPTR in
|
|
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|
class_init.
|
|
|
|
|
2005-11-01 12:35:39 +00:00
|
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2005-11-01 Josef Zlomek <josef dot zlomek at xeris dot cz>
|
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|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
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* gst/matroska/ebml-write.c: (gst_ebml_write_new),
|
|
|
|
(gst_ebml_write_reset), (gst_ebml_write_element_new):
|
|
|
|
* gst/matroska/ebml-write.h:
|
|
|
|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_write_data):
|
|
|
|
Set timestamps on outgoing ebml headers as well, so that the
|
|
|
|
element after matroskamux can get the timestamp already when
|
|
|
|
reading the first ebml element and doesn't have to wait for
|
|
|
|
the actual data buffer for that (#320308).
|
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|
2005-10-31 22:08:52 +00:00
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|
2005-10-31 Andy Wingo <wingo@pobox.com>
|
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|
|
|
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* gst/videomixer/videomixer.c (gst_videomixer_pad_unlink)
|
|
|
|
(gst_videomixer_pad_link): Kill some memleaks.
|
|
|
|
(gst_videomixer_pad_get_property): Style fix.
|
|
|
|
(gst_videomixer_pad_set_property): Style fix.
|
|
|
|
(gst_videomixer_pad_init): Style fix.
|
|
|
|
(gst_videomixer_update_queues): Kill memleak.
|
|
|
|
(gst_videomixer_loop): Kill memleak.
|
|
|
|
(gst_videomixer_collected): Kill memleak.
|
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|
|
|
2005-10-31 19:08:27 +00:00
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|
2005-10-31 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* gst/auparse/gstauparse.c: gst_auparse_init, gst_auparse_chain,
|
|
|
|
gst_auparse_change_state:
|
|
|
|
Just some cleanup.
|
|
|
|
|
2005-10-31 14:41:31 +00:00
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|
|
2005-10-31 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
|
|
|
* ext/speex/gstspeexenc.c: (gst_speexenc_chain)
|
|
|
|
Add checks to GST_FLOW_NOT_LINKED for values returned
|
|
|
|
from gst_pad_push.
|
|
|
|
|
2005-10-31 12:00:10 +00:00
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|
2005-10-31 Zeeshan Ali <zeenix@gmail.com>
|
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|
|
|
|
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* gst/rtp/gstrtpg711dec.c: (gst_rtpg711dec_setcaps),
|
|
|
|
(gst_rtpg711dec_process):
|
|
|
|
* gst/rtp/gstrtpgsmenc.c:
|
|
|
|
Payloader now sets some default caps on the srcpad if caps on the sinkpad
|
|
|
|
are never set. This is important for the g711 to work with burger's rtpbin
|
|
|
|
element.
|
|
|
|
|
2005-10-28 19:19:40 +00:00
|
|
|
2005-10-28 Edgard Lima <edgard.lima@cin.ufpe.br>
|
|
|
|
|
|
|
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* ext/speex/gstspeexenc.c: (gst_speexenc_chain),
|
|
|
|
(gst_speexenc_push_buffer):
|
|
|
|
Add checks for return values from gst_pad_push and
|
|
|
|
gst_pad_alloc_buffer.
|
|
|
|
|
2005-10-28 15:32:48 +00:00
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|
2005-10-28 Michal Benes <michal dot benes at xeris dot cz>
|
|
|
|
|
|
|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
|
|
|
|
|
|
|
* gst/matroska/matroska-demux.c: (gst_matroska_demux_init_stream),
|
|
|
|
(gst_matroska_demux_parse_info),
|
|
|
|
(gst_matroska_demux_parse_blockgroup_or_simpleblock),
|
|
|
|
(gst_matroska_demux_parse_cluster):
|
|
|
|
* gst/matroska/matroska-ids.h:
|
|
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|
* gst/matroska/matroska-mux.c: (gst_matroska_mux_class_init),
|
|
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|
(gst_matroska_mux_init), (gst_matroska_mux_start),
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(gst_matroska_mux_create_buffer_header),
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(gst_matroska_mux_write_data), (gst_matroska_mux_set_property),
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(gst_matroska_mux_get_property):
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* gst/matroska/matroska-mux.h:
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|
Add SimpleBlock support to matroska demuxer and muxer (part of
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Matroska v2). (#319731)
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2005-10-28 13:24:40 +00:00
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2005-10-28 Wim Taymans <wim@fluendo.com>
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* ext/jpeg/gstjpegdec.c: (gst_jpeg_dec_init), (gst_jpeg_dec_chain),
|
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(gst_jpeg_dec_change_state):
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* ext/jpeg/gstjpegdec.h:
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Cleanups. Don't create caps for every chain.
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|
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|
2005-10-27 18:46:32 +00:00
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2005-10-27 Edgard Lima Edgard Lima <edgard.lima@indt.org.br>
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* gst/law/alaw-encode.c: (gst_alawenc_init),
|
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(alawenc_setcaps), (gst_alawenc_chain)
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* gst/law/alaw-encode.h:
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* gst/law/mulaw-encode.c: (gst_mulawenc_init),
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(mulawenc_setcaps), (gst_mulawenc_chain)
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* gst/law/mulaw-encode.h:
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Set timestamp on buffer and it allows RTP G711 elements
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work properly.
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2005-10-27 11:00:40 +00:00
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2005-10-27 Wim Taymans <wim@fluendo.com>
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* sys/oss/gstosssrc.c: (gst_oss_src_prepare):
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Set correct format on oss instead of a silly value.
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2005-10-27 09:52:08 +00:00
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2005-10-27 Julien MOUTTE <julien@moutte.net>
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
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(gst_video_box_transform_caps), (gst_video_box_set_caps),
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(gst_video_box_get_unit_size), (gst_video_box_copy_plane_i420),
|
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(gst_video_box_i420), (gst_video_box_ayuv): Use liboil for
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I420 rendering as well, doesn't bring much for my platform.
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Might help on some other platforms.
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2005-10-26 21:47:36 +00:00
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2005-10-27 Zeeshan Ali <zeenix@gmail.com>
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* gst/rtp/gstrtpgsmenc.c:
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* gst/rtp/gstrtpgsmparse.c:
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Declaring the padtemplate correctly.
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2005-10-26 20:28:32 +00:00
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2005-10-26 Zeeshan Ali <zeenix@gmail.com>
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* gst/rtp/gstrtpg711dec.c:
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* gst/rtp/gstrtpg711enc.c:
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* gst/rtp/gstrtpgsmenc.c:
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* gst/rtp/gstrtpgsmparse.c:
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Setting the proper copyright notice.
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2005-10-26 17:23:06 +00:00
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2005-10-26 Julien MOUTTE <julien@moutte.net>
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* gst/videobox/Makefile.am: Use liboil.
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
|
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(gst_video_box_set_property), (gst_video_box_transform_caps),
|
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(gst_video_box_set_caps), (gst_video_box_get_unit_size),
|
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(gst_video_box_ayuv): Lot of optimization in AYUV rendering
|
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using liboil. Will dot the same to I420 border generation
|
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tomorrow.
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2005-10-26 14:50:59 +00:00
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2005-10-26 Zeeshan Ali <zeenix@gmail.com>
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* gst/rtp/gstrtpg711dec.c: (gst_rtpg711dec_base_init),
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(gst_rtpg711dec_class_init), (gst_rtpg711dec_init),
|
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(gst_rtpg711dec_setcaps), (gst_rtpg711dec_process):
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* gst/rtp/gstrtpg711dec.h:
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* gst/rtp/gstrtpg711enc.c: (gst_rtpg711enc_base_init),
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(gst_rtpg711enc_init), (gst_rtpg711enc_handle_buffer):
|
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* gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_base_init),
|
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(gst_rtpgsmenc_init), (gst_rtpgsmenc_setcaps),
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(gst_rtpgsmenc_handle_buffer):
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* gst/rtp/gstrtpgsmparse.c: (gst_rtpgsmparse_base_init),
|
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(gst_rtpgsmparse_class_init), (gst_rtpgsmparse_init),
|
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(gst_rtpgsmparse_setcaps), (gst_rtpgsmparse_process):
|
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* gst/rtp/gstrtpgsmparse.h:
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|
Hacked the G711 (de)payloader to try to make things right. rtpg711dec now
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|
|
inherits from the basertpdepayloader.
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|
2005-10-26 14:23:45 +00:00
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|
2005-10-26 Julien MOUTTE <julien@moutte.net>
|
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
|
|
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(gst_video_box_transform_caps), (gst_video_box_get_unit_size),
|
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|
|
(gst_video_box_ayuv): Removing this forgotten debug.
|
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|
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|
2005-10-26 14:08:49 +00:00
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|
2005-10-26 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
|
|
|
|
(gst_video_box_transform_caps), (gst_video_box_get_unit_size),
|
|
|
|
(gst_video_box_ayuv): Fix the stride issue when boxing to AYUV.
|
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|
|
|
2005-10-26 11:12:34 +00:00
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|
2005-10-26 Tim-Philipp Müller <tim at centricular dot net>
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|
|
|
|
|
|
* sys/oss/gstossaudio.c:
|
|
|
|
* sys/oss/gstossdmabuffer.c:
|
|
|
|
* sys/oss/gstosshelper.c:
|
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|
|
* sys/oss/gstossmixer.c:
|
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|
|
* sys/oss/gstossmixerelement.c:
|
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|
|
* sys/oss/gstossmixertrack.c:
|
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|
|
* sys/oss/gstosssink.c:
|
|
|
|
* sys/oss/gstosssrc.c:
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|
|
Actually use the 'oss' debug category we register.
|
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|
|
|
2005-10-26 10:38:18 +00:00
|
|
|
2005-10-26 Julien MOUTTE <julien@moutte.net>
|
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|
|
|
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* gst/videomixer/videomixer.c: (gst_videomixer_pad_set_property),
|
|
|
|
(gst_videomixer_pad_sink_setcaps), (gst_videomixer_getcaps):
|
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|
|
Use gst_pad_get_parent and drop the ref that was added through
|
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|
|
that call.
|
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|
2005-10-26 10:03:02 +00:00
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|
2005-10-26 Thomas Vander Stichele <thomas at apestaart dot org>
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|
|
|
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|
|
* gst/rtp/gstrtpgsmenc.c:
|
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|
|
Fix compilation
|
|
|
|
|
2005-10-25 21:09:36 +00:00
|
|
|
2005-10-25 Edgard Lima <edgard.lima@indt.org.br>
|
|
|
|
|
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|
|
* gst/rtp/gstrtpg711dec.c
|
|
|
|
Just removed a couple of lines of weird code used during
|
|
|
|
development/test time.
|
|
|
|
|
2005-10-25 19:19:38 +00:00
|
|
|
2005-10-25 Edgard Lima <edgard.lima@indt.org.br>
|
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|
|
|
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|
|
* gst/rtp/Makefile.am
|
|
|
|
* gst/rtp/gstrtp.c
|
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|
|
* gst/rtp/gstrtpg711enc.c
|
|
|
|
* gst/rtp/gstrtpg711enc.h
|
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|
|
* gst/rtp/gstrtpg711dec.c
|
|
|
|
* gst/rtp/gstrtpg711dec.h
|
|
|
|
Created G711 payloader and depayloader (it supports mulaw and alaw
|
|
|
|
(dec)encoders)
|
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|
|
|
2005-10-25 17:55:19 +00:00
|
|
|
2005-10-25 Julien MOUTTE <julien@moutte.net>
|
2005-10-25 16:02:38 +00:00
|
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|
2005-10-25 17:55:19 +00:00
|
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|
* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
|
|
|
|
(gst_video_box_transform_caps), (gst_video_box_get_unit_size):
|
|
|
|
Doh ! I introduced wingo's bug again ! Sorry...
|
|
|
|
|
|
|
|
2005-10-25 Christian Schaller <christian@fluendo.com>
|
2005-10-25 16:02:38 +00:00
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* gst/rtp/Makefile.am: add missing header files for disting
|
|
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|
|
2005-10-25 15:07:02 +00:00
|
|
|
2005-10-25 Zeeshan Ali <zeenix@gmail.com>
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_get_type),
|
|
|
|
(gst_rtpgsmenc_base_init), (gst_rtpgsmenc_class_init),
|
|
|
|
(gst_rtpgsmenc_init), (gst_rtpgsmenc_setcaps),
|
|
|
|
(gst_rtpgsmenc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpgsmenc.h:
|
|
|
|
* gst/rtp/gstrtpgsmparse.c: (gst_rtpgsmparse_get_type),
|
|
|
|
(gst_rtpgsmparse_base_init), (gst_rtpgsmparse_class_init),
|
|
|
|
(gst_rtpgsmparse_init), (gst_rtpgsmparse_setcaps),
|
|
|
|
(gst_rtpgsmparse_finalize), (gst_rtpgsmparse_process):
|
|
|
|
* gst/rtp/gstrtpgsmparse.h:
|
|
|
|
Getting the GSM (de)payloader working and compatible with our plans for RTP.
|
|
|
|
|
2005-10-25 10:47:09 +00:00
|
|
|
2005-10-25 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngdec.c: (user_info_callback),
|
|
|
|
(gst_pngdec_caps_create_and_set), (gst_pngdec_task): Push
|
|
|
|
a newsegment event, move some redundant code in a single place.
|
|
|
|
|
2005-10-25 10:23:26 +00:00
|
|
|
2005-10-25 Julien MOUTTE <julien@moutte.net>
|
|
|
|
|
|
|
|
* ext/libpng/gstpngdec.c: (user_info_callback),
|
|
|
|
(gst_pngdec_caps_create_and_set), (gst_pngdec_task): Temporary
|
|
|
|
hack to get correct colors order when we have a png image with
|
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|
|
alpha channel.
|
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|
|
|
2005-10-24 17:29:02 +00:00
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|
|
2005-10-24 Edward Hervey <edward@fluendo.com>
|
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|
|
|
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|
|
* ext/dv/gstdvdemux.c: (gst_dvdemux_add_pads):
|
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|
|
Call gst_element_no_more_pads when there will be no more pads.
|
|
|
|
|
gst/rtp/: Added two new payloaders, an RFC 2190 payloader for h263 and a payload convertor for an asterisk server.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_get_type),
(gst_asteriskh263_base_init), (gst_asteriskh263_class_init),
(gst_asteriskh263_init), (gst_asteriskh263_finalize),
(gst_asteriskh263_chain), (gst_asteriskh263_set_property),
(gst_asteriskh263_get_property), (gst_asteriskh263_change_state),
(gst_asteriskh263_plugin_init):
* gst/rtp/gstasteriskh263.h:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtph263enc.c: (gst_rtph263enc_get_type),
(gst_rtph263enc_base_init), (gst_rtph263enc_class_init),
(gst_rtph263enc_init), (gst_rtph263enc_finalize),
(gst_rtph263enc_setcaps), (gst_rtph263enc_gobfiner),
(gst_rtph263enc_flush), (gst_rtph263enc_handle_buffer),
(gst_rtph263enc_plugin_init):
* gst/rtp/gstrtph263enc.h:
Added two new payloaders, an RFC 2190 payloader for h263 and
a payload convertor for an asterisk server.
2005-10-24 16:39:38 +00:00
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|
2005-10-24 Wim Taymans <wim@fluendo.com>
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|
* gst/rtp/Makefile.am:
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|
* gst/rtp/gstasteriskh263.c: (gst_asteriskh263_get_type),
|
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|
(gst_asteriskh263_base_init), (gst_asteriskh263_class_init),
|
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(gst_asteriskh263_init), (gst_asteriskh263_finalize),
|
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(gst_asteriskh263_chain), (gst_asteriskh263_set_property),
|
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(gst_asteriskh263_get_property), (gst_asteriskh263_change_state),
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(gst_asteriskh263_plugin_init):
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* gst/rtp/gstasteriskh263.h:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtph263enc.c: (gst_rtph263enc_get_type),
|
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(gst_rtph263enc_base_init), (gst_rtph263enc_class_init),
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(gst_rtph263enc_init), (gst_rtph263enc_finalize),
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(gst_rtph263enc_setcaps), (gst_rtph263enc_gobfiner),
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(gst_rtph263enc_flush), (gst_rtph263enc_handle_buffer),
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(gst_rtph263enc_plugin_init):
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* gst/rtp/gstrtph263enc.h:
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|
Added two new payloaders, an RFC 2190 payloader for h263 and
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|
|
a payload convertor for an asterisk server.
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2005-10-24 15:57:17 +00:00
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2005-10-24 Tim-Philipp Müller <tim at centricular dot net>
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* sys/oss/gstosssrc.c: (gst_oss_src_prepare):
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Set bytes_per_sample correctly (is not always 4, but
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depends on width and number of channels).
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ext/flac/gstflacenc.*: Fix seeking, so that flacenc can rewrite the header with the correct duration and amount of sa...
Original commit message from CVS:
* ext/flac/gstflacenc.c: (gst_flacenc_base_init),
(gst_flacenc_init), (gst_flacenc_sink_setcaps),
(gst_flacenc_seek_callback), (gst_flacenc_write_callback),
(gst_flacenc_sink_event), (gst_flacenc_chain),
(gst_flacenc_set_property), (gst_flacenc_get_property),
(gst_flacenc_change_state):
* ext/flac/gstflacenc.h:
Fix seeking, so that flacenc can rewrite the header with the
correct duration and amount of samples and all that at EOS;
also set timestamps and granulepos on outgoing buffers; add
debug category; fix state change function.
2005-10-24 15:50:06 +00:00
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2005-10-24 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacenc.c: (gst_flacenc_base_init),
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(gst_flacenc_init), (gst_flacenc_sink_setcaps),
|
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(gst_flacenc_seek_callback), (gst_flacenc_write_callback),
|
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|
(gst_flacenc_sink_event), (gst_flacenc_chain),
|
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(gst_flacenc_set_property), (gst_flacenc_get_property),
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(gst_flacenc_change_state):
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|
* ext/flac/gstflacenc.h:
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|
Fix seeking, so that flacenc can rewrite the header with the
|
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|
|
correct duration and amount of samples and all that at EOS;
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|
also set timestamps and granulepos on outgoing buffers; add
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debug category; fix state change function.
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2005-10-24 13:46:09 +00:00
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2005-10-24 Julien MOUTTE <julien@moutte.net>
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* gst/videomixer/videomixer.c: Don't restrict video geometry
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from 16 to 4096.
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2005-10-24 13:22:14 +00:00
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2005-10-24 Julien MOUTTE <julien@moutte.net>
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* gst/videobox/gstvideobox.c: (gst_video_box_class_init),
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(gst_video_box_transform_caps), (gst_video_box_get_unit_size):
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Fix caps negotiation correctly, add debugging category.
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2005-10-24 13:02:47 +00:00
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2005-10-24 Christian Schaller <christian@fluendo.com>
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* configure.ac: Port over Thomas's change from base listing all plugins
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2005-10-24 08:59:24 +00:00
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2005-10-24 Julien MOUTTE <julien@moutte.net>
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* ext/libpng/gstpngdec.c: (gst_pngdec_init): Don't use fixed caps on
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a sink pad.
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2005-10-23 23:05:59 +00:00
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2005-10-24 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* docs/upload.mak:
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back to HEAD
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2005-10-23 22:43:08 +00:00
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=== release 0.9.4 ===
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2005-10-24 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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Releasing 0.9.4, "Dromiceiomimus"
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2005-10-21 17:00:58 +00:00
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2005-10-21 Tim-Philipp Müller <tim at centricular dot net>
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* ext/speex/gstspeexenc.c: (gst_speexenc_get_query_types),
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(gst_speexenc_src_query):
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Add position and duration query, fix query type function.
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_add_stream),
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(gst_matroska_demux_video_caps), (gst_matroska_demux_audio_caps):
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Let's not set non-fixed caps on source pads.
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2005-10-21 16:15:57 +00:00
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2005-10-21 Wim Taymans <wim@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_demux_frame):
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* gst/avi/gstavidemux.c: (gst_avi_demux_get_src_query_types),
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(gst_avi_demux_handle_seek):
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Set correct stream_time in newsegment event.
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avi can also handle a duration query now.
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2005-10-20 19:14:27 +00:00
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2005-10-20 Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-demux.c:
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(gst_matroska_demux_handle_src_query),
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(gst_matroska_demux_handle_seek_event),
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(gst_matroska_demux_loop_stream_parse_id):
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Fix duration query; fix basetime in newsegment event after
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seek; fix duration in initial newsegment event.
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_audio_pad_setcaps), (gst_matroska_mux_start):
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Extract number of channels and samplerate from vorbis headers;
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add some debug messages when querying the durations of the
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input streams.
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2005-10-20 11:50:53 +00:00
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2005-10-20 Wim Taymans <wim@fluendo.com>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
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(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
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Set stream time correctly in newsegment.
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2005-10-20 11:39:40 +00:00
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2005-10-20 Wim Taymans <wim@fluendo.com>
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_seek):
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Correctly fill in the stream time.
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2005-10-19 20:48:24 +00:00
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2005-10-19 Thomas Vander Stichele <thomas at apestaart dot org>
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* check/elements/level.c: (GST_START_TEST):
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* gst/level/gstlevel.c: (gst_level_message_new):
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* gst/level/level-example.c: (message_handler):
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use ELEMENT messages instead
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2005-10-19 15:57:04 +00:00
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2005-10-19 Wim Taymans <wim@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_get_src_query_types),
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(gst_dvdemux_src_query):
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* ext/flac/gstflacdec.c: (gst_flacdec_length),
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(gst_flacdec_src_query):
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* ext/raw1394/gstdv1394src.c: (gst_dv1394src_query):
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* ext/speex/gstspeexdec.c: (speex_dec_src_query):
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* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query):
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* gst/debug/gstnavseek.c: (gst_navseek_seek):
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* gst/debug/progressreport.c: (gst_progress_report_report):
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* gst/matroska/ebml-read.c: (gst_ebml_read_get_length):
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* gst/matroska/matroska-demux.c:
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(gst_matroska_demux_handle_src_query):
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* gst/matroska/matroska-mux.c: (gst_matroska_mux_start):
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* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
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(gst_wavparse_pad_convert), (gst_wavparse_pad_query),
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(gst_wavparse_srcpad_event):
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API change fix.
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2005-10-19 10:57:46 +00:00
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2005-10-19 Tim-Philipp Müller <tim at centricular dot net>
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* gst/goom/filters.c:
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* gst/goom/graphic.h:
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* gst/goom/lines.c:
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Make inline functions either 'static inline' or 'extern inline',
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otherwise the Forte compiler apparently won't inline them (#317300).
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ext/libpng/gstpngdec.*: Complete rewrite of pngdec. It's now very nice and handle push/pull based model. if you have ...
Original commit message from CVS:
2005-10-19 Julien MOUTTE <julien@moutte.net>
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
(gst_pngdec_init), (user_error_fn), (user_warning_fn),
(user_info_callback), (user_endrow_callback),
(user_end_callback),
(user_read_data), (gst_pngdec_caps_create_and_set),
(gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event),
(gst_pngdec_libpng_clear), (gst_pngdec_libpng_init),
(gst_pngdec_change_state), (gst_pngdec_sink_activate_push),
(gst_pngdec_sink_activate_pull), (gst_pngdec_sink_activate):
* ext/libpng/gstpngdec.h: Complete rewrite of pngdec. It's now
very nice and handle push/pull based model. if you have filesrc
connected to it, it will do random access to load the png file.
If you have a network source that can't do _getrange, it does
progressive loading through the chain function.
* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
(transform_rgb), (transform_bgr): Fix caps negotiation correctly
thanks to Master Wim Taymans ;-)
2005-10-18 22:44:11 +00:00
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2005-10-19 Julien MOUTTE <julien@moutte.net>
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* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
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(gst_pngdec_init), (user_error_fn), (user_warning_fn),
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(user_info_callback), (user_endrow_callback), (user_end_callback),
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(user_read_data), (gst_pngdec_caps_create_and_set),
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(gst_pngdec_task), (gst_pngdec_chain), (gst_pngdec_sink_event),
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(gst_pngdec_libpng_clear), (gst_pngdec_libpng_init),
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(gst_pngdec_change_state), (gst_pngdec_sink_activate_push),
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(gst_pngdec_sink_activate_pull), (gst_pngdec_sink_activate):
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* ext/libpng/gstpngdec.h: Complete rewrite of pngdec. It's now
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very nice and handle push/pull based model. if you have filesrc
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connected to it, it will do random access to load the png file.
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If you have a network source that can't do _getrange, it does
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progressive loading through the chain function.
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
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(transform_rgb), (transform_bgr): Fix caps negotiation correctly
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thanks to Master Wim Taymans ;-)
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2005-10-18 18:12:31 +00:00
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2005-10-18 Tim-Philipp Müller <tim at centricular dot net>
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2005-10-23 22:43:08 +00:00
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2005-10-18 18:12:31 +00:00
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* gst/matroska/Makefile.am:
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* gst/matroska/ebml-read.c:
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* gst/matroska/ebml-read.h:
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* gst/matroska/matroska-demux.c:
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* gst/matroska/matroska-demux.h:
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* gst/matroska/matroska.c: (plugin_init):
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Ported matroska demuxer to 0.9.
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2005-10-18 18:06:14 +00:00
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2005-10-18 Michal Benes <michal dot benes at xeris dot cz>
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Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
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* gst/matroska/matroska-mux.c:
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(gst_matroska_mux_video_pad_setcaps),
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(gst_matroska_mux_audio_pad_setcaps):
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Fix mpeg4 input handling (#318847); also, while we're at it,
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fix media type for Motion-JPEG: should be image/jpeg.
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2005-10-18 13:21:18 +00:00
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2005-10-18 Wim Taymans <wim@fluendo.com>
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* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
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(gst_wavparse_pad_convert), (gst_wavparse_srcpad_event):
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Fix for segment-start/stop API change.
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2005-10-17 17:18:56 +00:00
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2005-10-17 Julien MOUTTE <julien@moutte.net>
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_transform_caps),
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(transform_rgb), (transform_bgr): Handle caps negotiation in a better
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way.
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2005-10-17 16:59:20 +00:00
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2005-10-17 Julien MOUTTE <julien@moutte.net>
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* gst/videobox/gstvideobox.c: (gst_video_box_transform_caps),
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(gst_video_box_get_unit_size): Fix caps nego some more to get AYUV
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output declared in transform_caps.
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2005-10-17 15:23:24 +00:00
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2005-10-17 Julien MOUTTE <julien@moutte.net>
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* ext/libpng/gstpngdec.c: (gst_pngdec_init): We use fixed caps.
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2005-10-17 15:14:29 +00:00
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2005-10-17 Julien MOUTTE <julien@moutte.net>
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* gst/videobox/gstvideobox.c: (gst_video_box_transform_caps),
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(gst_video_box_get_unit_size): Fix wrong size calculations and
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implement get_unit_size correctly.
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2005-10-17 14:56:12 +00:00
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2005-10-17 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Enable flx plugin.
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* gst/flx/gstflxdec.c: (flx_decode_chunks):
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Fix gcc4 signedness issue.
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configure.ac: Adding videomixer.
Original commit message from CVS:
2005-10-17 Julien MOUTTE <julien@moutte.net>
* configure.ac: Adding videomixer.
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
(user_read_data), (gst_pngdec_chain): More debugging.
* gst/alpha/Makefile.am: Adding alphacolor
* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
(gst_alpha_color_class_init), (gst_alpha_color_init),
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps),
(transform_rgb), (transform_bgr),
(gst_alpha_color_transform_ip),
(plugin_init): Ported to 0.9 using in place base tranform.
* gst/videomixer/Makefile.am:
* gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
(gst_videomixer_pad_class_init),
(gst_videomixer_pad_sink_setcaps),
(gst_videomixer_pad_link), (gst_videomixer_pad_unlink),
(gst_videomixer_pad_init), (gst_videomixer_class_init),
(gst_videomixer_init), (gst_videomixer_getcaps),
(gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
(gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
(gst_videomixer_collected), (gst_videomixer_change_state):
Ported
to 0.9 using collectpads.
2005-10-17 08:46:30 +00:00
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2005-10-17 Julien MOUTTE <julien@moutte.net>
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* configure.ac: Adding videomixer.
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* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
|
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(user_read_data), (gst_pngdec_chain): More debugging.
|
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* gst/alpha/Makefile.am: Adding alphacolor
|
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* gst/alpha/gstalphacolor.c: (gst_alpha_color_base_init),
|
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|
(gst_alpha_color_class_init), (gst_alpha_color_init),
|
|
|
|
(gst_alpha_color_transform_caps), (gst_alpha_color_set_caps),
|
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|
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(transform_rgb), (transform_bgr), (gst_alpha_color_transform_ip),
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(plugin_init): Ported to 0.9 using in place base tranform.
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|
|
* gst/videomixer/Makefile.am:
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* gst/videomixer/videomixer.c: (gst_videomixer_pad_get_type),
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|
(gst_videomixer_pad_class_init), (gst_videomixer_pad_sink_setcaps),
|
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|
|
(gst_videomixer_pad_link), (gst_videomixer_pad_unlink),
|
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(gst_videomixer_pad_init), (gst_videomixer_class_init),
|
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(gst_videomixer_init), (gst_videomixer_getcaps),
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(gst_videomixer_request_new_pad), (gst_videomixer_fill_queues),
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(gst_videomixer_blend_buffers), (gst_videomixer_update_queues),
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(gst_videomixer_collected), (gst_videomixer_change_state): Ported
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to 0.9 using collectpads.
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2005-10-16 21:19:44 +00:00
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2005-10-17 Zeeshan Ali <zeenix at gmail dot com>
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* configure.ac:
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* gst/flx/Makefile.am:
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* gst/flx/gstflxdec.c: (gst_flxdec_init),
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(gst_flxdec_src_event_handler), (gst_flxdec_sink_event_handler),
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(gst_flxdec_chain), (gst_flxdec_change_state), (plugin_init):
|
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|
|
* gst/flx/gstflxdec.h:
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|
flx plugin ported to 0.9
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2005-10-16 14:33:05 +00:00
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2005-10-16 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/shout2/gstshout2.c: (gst_shout2send_change_state):
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use new gst_version_string()
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2005-10-16 13:17:11 +00:00
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2005-10-16 Andy Wingo <wingo@pobox.com>
|
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* configure.ac: GLIB_CHECK.
|
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|
2005-10-15 16:48:55 +00:00
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|
2005-10-15 Julien MOUTTE <julien@moutte.net>
|
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|
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|
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* ext/libpng/Makefile.am:
|
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|
|
* ext/libpng/gstpng.c: (plugin_init):
|
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|
|
* ext/libpng/gstpngdec.c: (gst_pngdec_class_init),
|
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|
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(gst_pngdec_init), (user_read_data), (gst_pngdec_chain):
|
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|
|
* ext/libpng/gstpngdec.h: Ported pngdec to 0.9
|
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|
|
|
2005-10-14 12:43:30 +00:00
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|
2005-10-14 Michal Benes <michal dot benes at xeris dot cz>
|
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|
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|
|
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
|
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|
* configure.ac:
|
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|
|
* gst/matroska/Makefile.am:
|
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|
|
* gst/matroska/ebml-ids.h:
|
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|
|
* gst/matroska/ebml-write.c:
|
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|
|
* gst/matroska/ebml-write.h:
|
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|
|
* gst/matroska/matroska-ids.h:
|
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|
|
* gst/matroska/matroska-mux.c:
|
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|
|
* gst/matroska/matroska-mux.h:
|
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|
|
* gst/matroska/matroska.c: (plugin_init):
|
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|
|
Port matroska muxer to 0.9 (#318847).
|
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2005-10-13 18:59:35 +00:00
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|
2005-10-13 Tim-Philipp Müller <tim at centricular dot net>
|
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* ext/speex/gstspeexenc.c: (gst_speexenc_get_tag_value),
|
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|
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(comment_init), (comment_add):
|
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|
|
Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE;
|
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|
|
use GST_READ_UINT32_LE() and friends rather than the private
|
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|
implementation of those same macros.
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2005-10-13 15:28:01 +00:00
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|
2005-10-13 Stefan Kost <ensonic@users.sf.net>
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* examples/stats/mp2ogg.c:
|
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|
more typo fixes
|
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|
2005-10-12 14:29:43 +00:00
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|
2005-10-12 Stefan Kost <ensonic@users.sf.net>
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* examples/indexing/indexmpeg.c: (main):
|
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|
* ext/esd/esdmon.c: (gst_esdmon_open_audio),
|
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|
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(gst_esdmon_close_audio), (gst_esdmon_change_state):
|
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|
* ext/esd/esdmon.h:
|
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|
|
* ext/gdk_pixbuf/gstgdkpixbuf.c: (gst_gdk_pixbuf_init):
|
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|
* ext/pango/gsttextoverlay.c: (gst_textoverlay_init):
|
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|
* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
|
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* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
|
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* gst/avi/gstavimux.c: (gst_avimux_init):
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* gst/matroska/matroska-demux.c: (gst_matroska_demux_init):
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* gst/multipart/multipartdemux.c: (gst_multipart_demux_init):
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* gst/multipart/multipartmux.c: (gst_multipart_mux_init):
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* gst/oldcore/gstmultifilesrc.c: (gst_multifilesrc_init),
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(gst_multifilesrc_get), (gst_multifilesrc_open_file),
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(gst_multifilesrc_close_file), (gst_multifilesrc_change_state):
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* gst/oldcore/gstmultifilesrc.h:
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|
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* gst/oldcore/gstpipefilter.c: (gst_pipefilter_init),
|
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(gst_pipefilter_open_file), (gst_pipefilter_close_file),
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(gst_pipefilter_change_state):
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|
|
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* gst/oldcore/gstpipefilter.h:
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|
|
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* gst/smoothwave/gstsmoothwave.c: (gst_smoothwave_init):
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|
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* gst/videomixer/videomixer.c: (gst_videomixer_init):
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* sys/osxaudio/gstosxaudiosink.c: (gst_osxaudiosink_init):
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* sys/osxaudio/gstosxaudiosink.h:
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* sys/osxaudio/gstosxaudiosrc.h:
|
2005-10-23 22:43:08 +00:00
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renamed GST_FLAGS macros to GST_OBJECT_FLAGS
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moved bitshift from macro to enum definition
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2005-10-12 14:29:43 +00:00
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2005-10-12 03:12:57 +00:00
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2005-10-12 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/Makefile.am:
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* ext/cairo/Makefile.am:
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* ext/cairo/gstcairo.c: (plugin_init):
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|
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* ext/cairo/gsttextoverlay.c: (gst_textoverlay_change_state):
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* ext/cairo/gsttimeoverlay.c: (gst_timeoverlay_update_font_height),
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(gst_timeoverlay_setup), (gst_timeoverlay_planar411):
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* ext/cairo/gsttimeoverlay.h:
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update of cairo-based timeoverlay to 1.0 Cairo API
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|
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doesn't work yet for resizing of output sink
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|
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2005-10-11 17:38:29 +00:00
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2005-10-11 Wim Taymans <wim@fluendo.com>
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|
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* ext/speex/gstspeexdec.c: (speex_dec_event), (speex_dec_chain):
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newsegment API fix.
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2005-10-11 16:33:08 +00:00
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2005-10-11 Wim Taymans <wim@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_handle_sink_event),
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(gst_dvdemux_demux_frame):
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|
|
|
* ext/flac/gstflacdec.c: (gst_flacdec_write):
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_chain):
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_stream_header),
|
|
|
|
(gst_avi_demux_handle_seek):
|
|
|
|
* gst/goom/gstgoom.c: (gst_goom_event):
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|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_stop_file):
|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_handle_seek),
|
|
|
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(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
|
|
|
|
(gst_wavparse_loop), (gst_wavparse_pad_convert),
|
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(gst_wavparse_srcpad_event):
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newsegment API update.
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2005-10-11 10:07:35 +00:00
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2005-10-11 Andy Wingo <wingo@pobox.com>
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|
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* ext/speex/gstspeexenc.c: Signedness cleanups.
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2005-10-10 19:57:40 +00:00
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2005-10-10 Edgard Lima <edgard.lima@indt.org.br>
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* PORTED_09:
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* ext/speex/Makefile.am:
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* ext/speex/gstspeex.c:
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* ext/speex/gstspeexenc.c:
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|
Speexenc ported to 0.9
|
|
|
|
|
sys/oss/: Cleanups, make device configurable in the sink, handle and report errors.
Original commit message from CVS:
* sys/oss/gstosssink.c: (gst_oss_sink_class_init),
(gst_oss_sink_init), (gst_oss_sink_set_property),
(gst_oss_sink_get_property), (gst_oss_sink_open),
(gst_oss_sink_prepare), (gst_oss_sink_reset):
* sys/oss/gstosssink.h:
* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
(gst_oss_src_set_property), (gst_oss_src_init), (gst_oss_src_open),
(gst_oss_src_prepare):
Cleanups, make device configurable in the sink, handle and report
errors.
2005-10-10 14:16:21 +00:00
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2005-10-10 Wim Taymans <wim@fluendo.com>
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|
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* sys/oss/gstosssink.c: (gst_oss_sink_class_init),
|
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(gst_oss_sink_init), (gst_oss_sink_set_property),
|
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|
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(gst_oss_sink_get_property), (gst_oss_sink_open),
|
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(gst_oss_sink_prepare), (gst_oss_sink_reset):
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* sys/oss/gstosssink.h:
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* sys/oss/gstosssrc.c: (gst_oss_src_class_init),
|
|
|
|
(gst_oss_src_set_property), (gst_oss_src_init), (gst_oss_src_open),
|
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(gst_oss_src_prepare):
|
|
|
|
Cleanups, make device configurable in the sink, handle and report
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errors.
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2005-10-10 12:31:07 +00:00
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2005-10-10 Wim Taymans <wim@fluendo.com>
|
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|
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* ext/gconf/gstgconfaudiosink.c: (gst_gconf_audio_sink_reset):
|
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* ext/gconf/gstgconfvideosink.c: (gst_gconf_video_sink_reset):
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Make sure element is NULL before removing from the bin.
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|
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|
2005-10-07 13:17:53 +00:00
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2005-10-07 Andy Wingo <wingo@pobox.com>
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|
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2005-10-07 15:24:24 +00:00
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* ext/raw1394/gstdv1394src.c: Make interruptible, so it won't
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block forever in a read().
|
2005-10-07 16:22:59 +00:00
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(gst_dv1394src_bus_reset): Post a message when the cable is
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unplugged.
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(gst_dv1394src_create, gst_dv1394src_unlock): Remove some prints.
|
2005-10-23 22:43:08 +00:00
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(gst_dv1394src_bus_reset): Don't unref the message.
|
2005-10-07 15:24:24 +00:00
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2005-10-07 13:17:53 +00:00
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* ext/raw1394/gstdv1394src.c: Clean up for style before doing some
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hacking. The only change should be that the state change stuff was
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put into basesrc's start() and stop() routines, which coalesces
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some steps.
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2005-10-07 11:30:41 +00:00
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2005-10-07 Tim-Philipp Müller <tim at centricular dot net>
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* configure.ac:
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Add check for mmap
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* gst/debug/Makefile.am:
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Only compile efence plugin on systems that have mmap.
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2005-10-05 11:38:29 +00:00
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2005-10-05 Tim-Philipp Müller <tim at centricular dot net>
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* gst/debug/Makefile.am:
|
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* gst/debug/breakmydata.c:
|
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* gst/debug/gstdebug.c:
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* gst/debug/gstnavigationtest.c:
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* gst/debug/gstnavseek.c:
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* gst/debug/gstnavseek.h:
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* gst/debug/progressreport.c:
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* gst/debug/testplugin.c:
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Port progressreport, navseek, navigationtest, testsink and
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breakmydata.
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2005-10-05 11:15:23 +00:00
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2005-10-05 Edward Hervey <edward@fluendo.com>
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* ext/dv/gstdvdemux.c: (gst_dvdemux_src_convert),
|
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(gst_dvdemux_src_query):
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|
Fixes for better conversion
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2005-10-04 17:58:40 +00:00
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2005-10-04 Michael Smith <msmith@fluendo.com>
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* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_reset),
|
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(gst_auto_audio_sink_find_best), (gst_auto_audio_sink_detect):
|
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* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_reset),
|
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(gst_auto_video_sink_find_best), (gst_auto_video_sink_detect):
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Set state of elements to NULL before removing from bins.
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Set state of test element to NULL if we failed to move it to READY
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2005-10-04 17:44:43 +00:00
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2005-10-04 Edward Hervey <edward@fluendo.com>
|
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|
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* ext/dv/Makefile.am:
|
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|
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* ext/dv/gstdvdemux.c: (gst_dvdemux_src_query), (gst_dvdemux_src_conver):
|
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Added DEFAULT <==> BYTES, TIME conversions on srcpad,
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Corrected the query function for position so it doesn't forget what
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|
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format was asked, and calls the conversion functions on the correct pad.
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2005-10-03 17:59:18 +00:00
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2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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back to development
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2005-10-03 17:48:57 +00:00
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=== release 0.9.3 ===
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2005-10-03 Thomas Vander Stichele <thomas at apestaart dot org>
|
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|
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* configure.ac:
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Releasing 0.9.3, "Aramis"
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2005-10-02 23:08:35 +00:00
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2005-10-03 Andy Wingo <wingo@pobox.com>
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* ext/flac/gstflacdec.c (gst_flacdec_write): Deal with pad_alloc
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error returns.
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2005-10-02 15:33:14 +00:00
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2005-10-02 Andy Wingo <wingo@pobox.com>
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* configure.ac (GST_PLUGIN_LDFLAGS): Change to be like -base.
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|
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* ext/flac/gstflacenc.c: Ported to 0.9.
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|
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* ext/flac/gstflacdec.c (gst_flacdec_loop): Handle errors better.
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* ext/flac/Makefile.am: Add the GST_PLUGINS_BASE cflags and libs,
|
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|
and link to gsttagedit. Enable flacenc.
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* ext/flac/gstflacdec.c: Re-enable tag reading.
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|
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2005-09-30 16:36:49 +00:00
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2005-09-30 Wim Taymans <wim@fluendo.com>
|
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* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
|
|
|
|
* gst/rtp/gstrtpgsmparse.c:
|
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|
|
* gst/rtp/gstrtph263penc.c:
|
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|
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* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
|
|
|
|
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
|
|
|
|
(gst_rtpmp4venc_set_property):
|
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|
|
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
|
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|
Various class and caps fixes from Andre Magalhaes (andrunko)
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2005-09-29 13:08:41 +00:00
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2005-09-29 Wim Taymans <wim@fluendo.com>
|
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* gst/level/level-example.c: (main):
|
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|
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Update for new bus API.
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|
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|
2005-09-28 09:45:00 +00:00
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|
2005-09-28 Tim-Philipp Müller <tim at centricular dot net>
|
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|
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|
|
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* configure.ac:
|
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|
|
Fix unexpanded autoconf macro GST_DOC, which has been renamed
|
|
|
|
to GST_DOCBOOK_CHECK (see common/m4/gst-doc.m4) (#316202).
|
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2005-09-27 15:12:45 +00:00
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2005-09-27 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
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* sys/oss/gstosssink.c: (gst_oss_sink_prepare):
|
|
|
|
Fix playback of mono streams (bytes_per_sample should be set
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|
|
|
from the sample width and the number of channels negotiated,
|
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|
|
and not just be set to 4) (#317338)
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|
2005-09-26 14:42:09 +00:00
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2005-09-26 Wim Taymans <wim@fluendo.com>
|
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|
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* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush),
|
|
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(gst_rtpmpaenc_handle_buffer):
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|
Set buffer duration correctly.
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|
2005-09-26 13:06:27 +00:00
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|
2005-09-26 Tim-Philipp Müller <tim at centricular dot net>
|
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|
|
|
|
|
|
* gst/avi/gstavidemux.c: (gst_avi_demux_base_init),
|
|
|
|
(gst_avi_demux_class_init), (gst_avi_demux_parse_stream),
|
|
|
|
(gst_avi_demux_change_state):
|
|
|
|
Don't crash when encountering a stream with an unknown fourcc or
|
|
|
|
codec id. Instead, create a pad of type video/x-avi-unknown or
|
|
|
|
audio/x-avi-unknown, which as a side-effect also results in less
|
|
|
|
confusing error messages in players ('no decoder' vs. 'no streams');
|
|
|
|
minor fixes to state change function and class_init function.
|
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|
2005-09-24 13:10:52 +00:00
|
|
|
2005-09-24 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
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* gst/autodetect/gstautoaudiosink.c: (gst_auto_audio_sink_init):
|
|
|
|
* gst/autodetect/gstautovideosink.c: (gst_auto_video_sink_init):
|
|
|
|
These are sinks.
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|
|
|
|
2005-09-24 12:10:02 +00:00
|
|
|
2005-09-24 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* check/elements/level.c: (GST_START_TEST):
|
|
|
|
fix test for new GstClockTime use
|
|
|
|
* gst/level/gstlevel.c: (gst_level_init), (gst_level_set_caps),
|
|
|
|
(gst_level_transform_ip):
|
|
|
|
* gst/level/gstlevel.h:
|
|
|
|
fix up the decay peak, ensuring the decay peak is never lower
|
|
|
|
than the peak for that interval
|
|
|
|
|
2005-09-23 18:23:04 +00:00
|
|
|
2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* docs/plugins/gst-plugins-good-plugins.args:
|
|
|
|
* docs/plugins/inspect/plugin-alpha.xml:
|
|
|
|
* docs/plugins/inspect/plugin-rtp.xml:
|
|
|
|
* gst/level/gstlevel.c: (gst_level_set_caps),
|
|
|
|
(gst_level_transform_ip):
|
|
|
|
updating docs
|
|
|
|
|
2005-09-23 18:15:51 +00:00
|
|
|
2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* Makefile.am:
|
|
|
|
* check/elements/level.c: (GST_START_TEST):
|
|
|
|
* gst/level/Makefile.am:
|
|
|
|
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_init),
|
|
|
|
(gst_level_set_property), (gst_level_get_property),
|
|
|
|
(gst_level_set_caps), (gst_level_message_new),
|
|
|
|
(gst_level_transform_ip):
|
|
|
|
* gst/level/gstlevel.h:
|
|
|
|
* gst/level/level-example.c: (message_handler), (main):
|
|
|
|
convert to using GstClockTime for all time values, finally.
|
|
|
|
|
2005-09-23 04:23:00 +00:00
|
|
|
2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
|
|
|
|
|
|
|
|
* gst/auparse/Makefile.am:
|
|
|
|
* gst/auparse/gstauparse.c: (gst_auparse_chain):
|
|
|
|
fix up build
|
|
|
|
* configure.ac:
|
|
|
|
* ext/cairo/Makefile.am:
|
|
|
|
* ext/dv/Makefile.am:
|
|
|
|
* ext/esd/Makefile.am:
|
|
|
|
* ext/flac/Makefile.am:
|
|
|
|
* ext/gconf/Makefile.am:
|
|
|
|
* ext/gdk_pixbuf/Makefile.am:
|
|
|
|
* ext/jpeg/Makefile.am:
|
|
|
|
* ext/ladspa/Makefile.am:
|
|
|
|
* ext/libcaca/Makefile.am:
|
|
|
|
* ext/libmng/Makefile.am:
|
|
|
|
* ext/libpng/Makefile.am:
|
|
|
|
* ext/mikmod/Makefile.am:
|
|
|
|
* ext/pango/Makefile.am:
|
|
|
|
* ext/raw1394/Makefile.am:
|
|
|
|
* ext/shout2/Makefile.am:
|
|
|
|
* ext/speex/Makefile.am:
|
|
|
|
* gst/alpha/Makefile.am:
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* gst/autodetect/Makefile.am:
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* gst/avi/Makefile.am:
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|
* gst/cutter/Makefile.am:
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* gst/debug/Makefile.am:
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* gst/effectv/Makefile.am:
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* gst/flx/Makefile.am:
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* gst/goom/Makefile.am:
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* gst/interleave/Makefile.am:
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* gst/law/Makefile.am:
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* gst/matroska/Makefile.am:
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* gst/median/Makefile.am:
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* gst/monoscope/Makefile.am:
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* gst/multipart/Makefile.am:
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* gst/oldcore/Makefile.am:
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* gst/rtp/Makefile.am:
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* gst/rtsp/Makefile.am:
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* gst/smoothwave/Makefile.am:
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* gst/smpte/Makefile.am:
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* gst/videobox/Makefile.am:
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* gst/videofilter/Makefile.am:
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* gst/videomixer/Makefile.am:
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* gst/wavenc/Makefile.am:
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|
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* gst/wavparse/Makefile.am:
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|
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* sys/oss/Makefile.am:
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* sys/osxaudio/Makefile.am:
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* sys/osxvideo/Makefile.am:
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fix up GST_LIBS use
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2005-09-23 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst/auparse/Makefile.am:
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* gst/auparse/gstauparse.c: (gst_auparse_chain):
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fix build of auparse
|
2005-09-22 22:38:48 +00:00
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2005-09-22 Edgard Lima (edgard.lima@indt.org.br)
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* configure.ac:
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* PORTED_09:
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* gst/auparse/gstauparse.c:
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* gst/auparse/gstauparse.h:
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Auparse ported to 0.9.
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2005-09-22 14:13:36 +00:00
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2005-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/TODO:
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* gst/rtp/gstrtpdec.c: (gst_rtpdec_getcaps):
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* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
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(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
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(gst_rtpmp4venc_set_property):
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* gst/rtp/gstrtpmp4venc.h:
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|
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* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_handle_buffer):
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|
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* gst/rtp/gstrtpmpaenc.h:
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Use is_filled to both check MTU and max-ptime of base class.
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2005-09-22 11:28:23 +00:00
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2005-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
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(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer),
|
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(gst_rtpmp4venc_set_property):
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Don't fragment packets with multiple frames.
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|
2005-09-22 10:39:11 +00:00
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2005-09-22 Wim Taymans <wim@fluendo.com>
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* gst/rtp/TODO:
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* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
|
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* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_class_init),
|
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(gst_rtpmp4venc_init), (gst_rtpmp4venc_parse_data),
|
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(gst_rtpmp4venc_handle_buffer), (gst_rtpmp4venc_set_property),
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(gst_rtpmp4venc_get_property):
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* gst/rtp/gstrtpmp4venc.h:
|
|
|
|
Remove g_print.
|
|
|
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Update TODO
|
|
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Make payload encoder a bit smarter and more correct with
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timestamps.
|
|
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Added option in payloader to include config string in-band.
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2005-09-21 19:41:45 +00:00
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2005-09-21 Wim Taymans <wim@fluendo.com>
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
|
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(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
|
|
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(gst_rtspsrc_send):
|
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|
|
Strip spaces for key/value pairs.
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2005-09-21 17:53:26 +00:00
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|
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2005-09-21 Wim Taymans <wim@fluendo.com>
|
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* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_parse_rtpmap),
|
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(gst_rtspsrc_media_to_caps), (gst_rtspsrc_stream_setup_rtp),
|
|
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|
(gst_rtspsrc_stream_configure_transport), (gst_rtspsrc_send),
|
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(gst_rtspsrc_change_state):
|
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|
|
More SDP parsing and caps setting.
|
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|
Do NO_PREROLL differently.
|
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add pads only after negotiated.
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* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init),
|
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(gst_udpsrc_getcaps):
|
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|
Implement the getcaps function.
|
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|
2005-09-21 17:50:29 +00:00
|
|
|
2005-09-21 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
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* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps),
|
|
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|
(gst_rtpamrdec_chain):
|
|
|
|
Handle multiple AMr packets per payload. Handle CRC and
|
|
|
|
parse ILL/ILP.
|
|
|
|
|
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* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_setcaps):
|
|
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|
Make caps params strings for easy SDP mapping.
|
|
|
|
|
|
|
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* gst/rtp/gstrtpdec.c: (gst_rtpdec_init), (gst_rtpdec_getcaps):
|
|
|
|
Handle capsnego better.
|
|
|
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|
|
|
|
* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps):
|
|
|
|
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_new_caps):
|
|
|
|
Generate and parse config string in the caps.
|
|
|
|
|
2005-09-21 12:19:24 +00:00
|
|
|
2005-09-21 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/README:
|
|
|
|
Update README
|
|
|
|
|
|
|
|
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_sink_setcaps):
|
|
|
|
Make extra params as strings.
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_set_state),
|
|
|
|
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
|
|
|
|
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send):
|
|
|
|
Make state change return NO_PREROLL as this is a live
|
|
|
|
source.
|
|
|
|
|
|
|
|
* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
|
|
|
|
Don't unref old caps when NULL.
|
|
|
|
|
gst/rtsp/: Add URI handler.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send),
(gst_rtspsrc_open), (gst_rtspsrc_uri_get_type),
(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri),
(gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init):
* gst/rtsp/sdpmessage.c: (sdp_media_get_format):
* gst/rtsp/sdpmessage.h:
Add URI handler.
Parse SDP and create caps.
2005-09-20 17:35:11 +00:00
|
|
|
2005-09-20 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_get_type),
|
|
|
|
(gst_rtspsrc_parse_rtpmap), (gst_rtspsrc_media_to_caps),
|
|
|
|
(gst_rtspsrc_stream_setup_rtp), (gst_rtspsrc_send),
|
|
|
|
(gst_rtspsrc_open), (gst_rtspsrc_uri_get_type),
|
|
|
|
(gst_rtspsrc_uri_get_protocols), (gst_rtspsrc_uri_get_uri),
|
|
|
|
(gst_rtspsrc_uri_set_uri), (gst_rtspsrc_uri_handler_init):
|
|
|
|
* gst/rtsp/sdpmessage.c: (sdp_media_get_format):
|
|
|
|
* gst/rtsp/sdpmessage.h:
|
|
|
|
Add URI handler.
|
|
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|
Parse SDP and create caps.
|
|
|
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|
2005-09-20 07:30:31 +00:00
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|
|
2005-09-20 Stefan Kost <ensonic@users.sf.net>
|
|
|
|
|
|
|
|
* gst/alpha/gstalpha.c:
|
2005-10-03 17:48:57 +00:00
|
|
|
fix element description
|
2005-09-20 07:30:31 +00:00
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2005-09-19 17:03:55 +00:00
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2005-09-19 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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|
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_chain):
|
|
|
|
Don't check payload for now.
|
|
|
|
|
gst/wavparse/gstwavparse.*: Fix wavparse some more.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
(gst_wavparse_init), (gst_wavparse_parse_file_header),
(gst_wavparse_stream_init), (gst_wavparse_handle_seek),
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
(gst_wavparse_loop), (gst_wavparse_pad_convert),
(gst_wavparse_pad_query), (gst_wavparse_srcpad_event),
(gst_wavparse_change_state):
* gst/wavparse/gstwavparse.h:
Fix wavparse some more.
2005-09-19 16:26:30 +00:00
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2005-09-19 Wim Taymans <wim@fluendo.com>
|
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|
|
|
|
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* gst/wavparse/gstwavparse.c: (gst_wavparse_reset),
|
|
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(gst_wavparse_init), (gst_wavparse_parse_file_header),
|
|
|
|
(gst_wavparse_stream_init), (gst_wavparse_handle_seek),
|
|
|
|
(gst_wavparse_stream_headers), (gst_wavparse_stream_data),
|
|
|
|
(gst_wavparse_loop), (gst_wavparse_pad_convert),
|
|
|
|
(gst_wavparse_pad_query), (gst_wavparse_srcpad_event),
|
|
|
|
(gst_wavparse_change_state):
|
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|
|
* gst/wavparse/gstwavparse.h:
|
|
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|
Fix wavparse some more.
|
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|
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|
2005-09-19 11:48:13 +00:00
|
|
|
2005-09-19 Wim Taymans <wim@fluendo.com>
|
|
|
|
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* check/elements/level.c: (GST_START_TEST):
|
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Fix for bus API change.
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2005-09-19 11:38:10 +00:00
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2005-09-19 Wim Taymans <wim@fluendo.com>
|
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|
|
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* gst/level/level-example.c: (main):
|
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|
|
Fix for new bus API.
|
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|
|
|
|
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* gst/udp/gstudpsrc.c: (gst_udpsrc_set_property):
|
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Set caps on pads.
|
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|
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|
2005-09-15 13:57:56 +00:00
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|
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2005-09-15 Wim Taymans <wim@fluendo.com>
|
|
|
|
|
|
|
|
* gst/rtp/Makefile.am:
|
|
|
|
* gst/rtp/README:
|
|
|
|
* gst/rtp/gstrtp.c: (plugin_init):
|
|
|
|
* gst/rtp/gstrtpamrdec.c: (gst_rtpamrdec_init),
|
|
|
|
(gst_rtpamrdec_sink_setcaps), (gst_rtpamrdec_chain),
|
|
|
|
(gst_rtpamrdec_change_state):
|
|
|
|
* gst/rtp/gstrtpamrenc.c: (gst_rtpamrenc_get_type),
|
|
|
|
(gst_rtpamrenc_class_init), (gst_rtpamrenc_init),
|
|
|
|
(gst_rtpamrenc_setcaps), (gst_rtpamrenc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpamrenc.h:
|
|
|
|
* gst/rtp/gstrtpgsmenc.c: (gst_rtpgsmenc_get_type),
|
|
|
|
(gst_rtpgsmenc_class_init), (gst_rtpgsmenc_init),
|
|
|
|
(gst_rtpgsmenc_setcaps), (gst_rtpgsmenc_handle_buffer):
|
|
|
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* gst/rtp/gstrtpgsmenc.h:
|
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|
|
* gst/rtp/gstrtpgsmparse.c: (gst_rtpgsmparse_class_init),
|
|
|
|
(gst_rtpgsm_caps_nego), (gst_rtpgsmparse_chain),
|
|
|
|
(gst_rtpgsmparse_set_property), (gst_rtpgsmparse_get_property),
|
|
|
|
(gst_rtpgsmparse_change_state):
|
|
|
|
* gst/rtp/gstrtpgsmparse.h:
|
|
|
|
* gst/rtp/gstrtph263pdec.c: (gst_rtph263pdec_class_init),
|
|
|
|
(gst_rtph263pdec_finalize), (gst_rtph263pdec_change_state):
|
|
|
|
* gst/rtp/gstrtph263penc.c: (gst_rtph263penc_get_type),
|
|
|
|
(gst_rtph263penc_class_init), (gst_rtph263penc_init),
|
|
|
|
(gst_rtph263penc_finalize), (gst_rtph263penc_setcaps),
|
|
|
|
(gst_rtph263penc_flush), (gst_rtph263penc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtph263penc.h:
|
|
|
|
* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_setcaps),
|
|
|
|
(gst_rtpmp4vdec_chain), (gst_rtpmp4vdec_change_state):
|
|
|
|
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_get_type),
|
|
|
|
(gst_rtpmp4venc_class_init), (gst_rtpmp4venc_init),
|
|
|
|
(gst_rtpmp4venc_finalize), (gst_rtpmp4venc_new_caps),
|
|
|
|
(gst_rtpmp4venc_setcaps), (gst_rtpmp4venc_flush),
|
|
|
|
(gst_rtpmp4venc_parse_data), (gst_rtpmp4venc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpmp4venc.h:
|
|
|
|
* gst/rtp/gstrtpmpadec.c:
|
|
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* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_get_type),
|
|
|
|
(gst_rtpmpaenc_class_init), (gst_rtpmpaenc_init),
|
|
|
|
(gst_rtpmpaenc_finalize), (gst_rtpmpaenc_setcaps),
|
|
|
|
(gst_rtpmpaenc_flush), (gst_rtpmpaenc_handle_buffer):
|
|
|
|
* gst/rtp/gstrtpmpaenc.h:
|
|
|
|
Updates to payloader/depayloaders, make payloaders use
|
|
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|
the base classes.
|
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|
|
Updated README with suggested RTP caps and how to convert
|
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|
to/from SDP.
|
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|
|
Added config descriptor in mp4v payloader.
|
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|
|
|
2005-09-15 10:47:58 +00:00
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|
2005-09-15 Andy Wingo <wingo@pobox.com>
|
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|
|
|
|
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* gst/autodetect/gstautoaudiosink.c (gst_auto_audio_sink_find_best):
|
|
|
|
* gst/autodetect/gstautovideosink.c
|
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(gst_auto_video_sink_find_best): Update for new registry API.
|
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|
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|
2005-09-14 20:51:47 +00:00
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|
2005-09-14 Thomas Vander Stichele <thomas at apestaart dot org>
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|
|
|
|
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|
* common/c-to-xml.py:
|
|
|
|
* common/gtk-doc-plugins.mak:
|
|
|
|
a simple py script to generate valid xml from a C example
|
|
|
|
probably also need to strip an MIT license when we decide
|
|
|
|
* docs/plugins/Makefile.am:
|
|
|
|
* gst/level/Makefile.am:
|
|
|
|
* gst/level/gstlevel.c: (gst_level_init):
|
|
|
|
* gst/level/level-example.c: (message_handler), (main):
|
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|
|
add an example to level that will show up in the docs
|
|
|
|
* gst/rtp/TODO:
|
|
|
|
add a note for the future
|
|
|
|
|
2005-09-14 11:44:11 +00:00
|
|
|
2005-09-14 Michael Smith <msmith@fluendo.com>
|
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|
|
|
|
|
|
* gst/wavenc/gstwavenc.c: (gst_wavenc_class_init):
|
|
|
|
Actually define the debug object being used in wavenc. Fixes #316205
|
|
|
|
|
2005-09-14 11:23:44 +00:00
|
|
|
2005-09-14 Michael Smith <msmith@fluendo.com>
|
|
|
|
|
|
|
|
* gst/smpte/Makefile.am:
|
|
|
|
Link against GST_BASE_LIBS, to get it linked against libgstbase;
|
|
|
|
needed to build on windows (bug 316204)
|
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|
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|
2005-09-12 16:14:48 +00:00
|
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|
2005-09-12 Andy Wingo <wingo@pobox.com>
|
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|
|
|
|
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|
* autogen.sh (package): Now type 'make' to build gst-plugins-good.
|
|
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2005-09-09 17:56:43 +00:00
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2005-09-09 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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* gst/fdsrc/Makefile.am:
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* gst/fdsrc/gstfdsrc.c:
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* gst/fdsrc/gstfdsrc.h:
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Move fdsrc back into gstreamer core elements.
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* gst/level/gstlevel.c: (gst_level_class_init),
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(gst_level_transform_ip):
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* gst/videobox/gstvideobox.c: (gst_video_box_set_property):
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Basetransform changes.
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2005-09-09 16:11:48 +00:00
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2005-09-09 Thomas Vander Stichele <thomas at apestaart dot org>
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* ext/jpeg/gstsmokeenc.c: (gst_smokeenc_chain):
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* ext/jpeg/smokecodec.c: (find_best_size):
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fix compiler warnings
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2005-09-09 11:09:49 +00:00
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2005-09-09 Thomas Vander Stichele <thomas at apestaart dot org>
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* gst-plugins-good.spec.in:
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spec file fixes
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* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
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(gst_multiudpsink_render), (gst_multiudpsink_add),
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(gst_multiudpsink_clear):
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it actually helps to actually stream if we hook up the
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"add" signal to an actual implementation
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* gst/udp/gstudpsrc.c: (gst_udpsrc_start):
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some debugging
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2005-09-08 16:26:17 +00:00
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2005-09-08 Flavio Oliveira <flavio.oliveira@indt.org.br>
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* ext/jpeg/Makefile.am:
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* ext/jpeg/gstjpeg.c
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* ext/jpeg/gstjpegenc.c:
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* ext/jpeg/gstsmokeenc.c:
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2005-09-07 13:49:37 +00:00
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2005-09-07 Stefan Kost <ensonic@users.sf.net>
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* ext/flac/gstflacdec.c:
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* ext/flac/gstflacenc.c:
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* ext/flac/gstflactag.c:
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* ext/speex/gstspeexenc.c:
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gsttaginterface.h -> gsttagsetter.h
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Port to 0.9 and re-enable efence plugin.
Original commit message from CVS:
* configure.ac:
* gst/debug/Makefile.am:
* gst/debug/efence.c: (gst_efence_class_init), (gst_efence_init),
(gst_efence_chain), (gst_efence_buffer_alloc), (plugin_init),
(gst_fenced_buffer_finalize), (gst_fenced_buffer_copy),
(gst_fenced_buffer_alloc), (gst_fenced_buffer_class_init),
(gst_fenced_buffer_init), (gst_fenced_buffer_get_type):
Port to 0.9 and re-enable efence plugin.
2005-09-06 23:30:03 +00:00
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2005-09-07 Jan Schmidt <thaytan@mad.scientist.com>
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* configure.ac:
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* gst/debug/Makefile.am:
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* gst/debug/efence.c: (gst_efence_class_init), (gst_efence_init),
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(gst_efence_chain), (gst_efence_buffer_alloc), (plugin_init),
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(gst_fenced_buffer_finalize), (gst_fenced_buffer_copy),
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(gst_fenced_buffer_alloc), (gst_fenced_buffer_class_init),
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(gst_fenced_buffer_init), (gst_fenced_buffer_get_type):
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Port to 0.9 and re-enable efence plugin.
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2005-09-06 21:31:25 +00:00
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2005-09-06 Tim-Philipp Müller <tim at centricular dot net>
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* ext/flac/gstflacdec.c: (flac_caps_factory), (raw_caps_factory),
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(gst_flacdec_write), (gst_flacdec_convert_src):
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* ext/flac/gstflacdec.h:
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Add support for flac files with 24/32 bits per sample; and misc.
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minor clean-ups. Seeking is still partly broken (for me at least).
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gst/rtp/: Added mpeg4 video payload encoder/decoder.
Original commit message from CVS:
* gst/rtp/Makefile.am:
* gst/rtp/gstrtp.c: (plugin_init):
* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_get_type),
(gst_rtpmp4vdec_base_init), (gst_rtpmp4vdec_class_init),
(gst_rtpmp4vdec_init), (gst_rtpmp4vdec_setcaps),
(gst_rtpmp4vdec_chain), (gst_rtpmp4vdec_set_property),
(gst_rtpmp4vdec_get_property), (gst_rtpmp4vdec_change_state),
(gst_rtpmp4vdec_plugin_init):
* gst/rtp/gstrtpmp4vdec.h:
* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_get_type),
(gst_rtpmp4venc_base_init), (gst_rtpmp4venc_class_init),
(gst_rtpmp4venc_init), (gst_rtpmp4venc_setcaps),
(gst_rtpmp4venc_flush), (gst_rtpmp4venc_chain),
(gst_rtpmp4venc_set_property), (gst_rtpmp4venc_get_property),
(gst_rtpmp4venc_change_state), (gst_rtpmp4venc_plugin_init):
* gst/rtp/gstrtpmp4venc.h:
* gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_chain):
* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush):
Added mpeg4 video payload encoder/decoder.
Added some docs in mpa payloader.
2005-09-06 15:50:58 +00:00
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2005-09-06 Wim Taymans <wim@fluendo.com>
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* gst/rtp/Makefile.am:
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* gst/rtp/gstrtp.c: (plugin_init):
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* gst/rtp/gstrtpmp4vdec.c: (gst_rtpmp4vdec_get_type),
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(gst_rtpmp4vdec_base_init), (gst_rtpmp4vdec_class_init),
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(gst_rtpmp4vdec_init), (gst_rtpmp4vdec_setcaps),
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(gst_rtpmp4vdec_chain), (gst_rtpmp4vdec_set_property),
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(gst_rtpmp4vdec_get_property), (gst_rtpmp4vdec_change_state),
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(gst_rtpmp4vdec_plugin_init):
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* gst/rtp/gstrtpmp4vdec.h:
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* gst/rtp/gstrtpmp4venc.c: (gst_rtpmp4venc_get_type),
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(gst_rtpmp4venc_base_init), (gst_rtpmp4venc_class_init),
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(gst_rtpmp4venc_init), (gst_rtpmp4venc_setcaps),
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(gst_rtpmp4venc_flush), (gst_rtpmp4venc_chain),
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(gst_rtpmp4venc_set_property), (gst_rtpmp4venc_get_property),
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(gst_rtpmp4venc_change_state), (gst_rtpmp4venc_plugin_init):
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* gst/rtp/gstrtpmp4venc.h:
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* gst/rtp/gstrtpmpadec.c: (gst_rtpmpadec_chain):
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* gst/rtp/gstrtpmpaenc.c: (gst_rtpmpaenc_flush):
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Added mpeg4 video payload encoder/decoder.
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Added some docs in mpa payloader.
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2005-09-06 14:05:33 +00:00
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=== release 0.9.1 ===
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2005-09-06 Thomas Vander Stichele <thomas at apestaart dot org>
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* NEWS:
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* README:
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* RELEASE:
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* autogen.sh:
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* configure.ac:
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releasing 0.9.1, "Blondie"
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2005-09-05 16:28:16 +00:00
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2005-09-05 Andy Wingo <wingo@pobox.com>
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* ext/dv/gstdvdemux.c (gst_dvdemux_chain): Move the pad adding
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here from the state change handler, so we fire signals without
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holding the state lock.
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2005-09-02 Andy Wingo <wingo@pobox.com>
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* All plugins updated for element state changes.
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2005-09-01 20:23:22 +00:00
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2005-09-01 Thomas Vander Stichele <thomas at apestaart dot org>
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* Makefile.am:
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* check/.cvsignore:
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* check/Makefile.am:
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* check/elements/.cvsignore:
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* check/elements/level.c: (setup_level), (cleanup_level),
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(GST_START_TEST), (level_suite), (main):
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add a test for level
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* common/release.mak:
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move the enum checking to release.mak
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* configure.ac:
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add valgrind and check checking
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* gst/level/gstlevel.c: (gst_level_transform):
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* gst/level/gstlevel.h:
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fix Andy's cast bug
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convert a field to int
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fix the arithmetic to int when checking for emit so that a
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100-sample buffer of a 1000Hz stream triggers after 0.1 sec
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add docs to build
Original commit message from CVS:
* Makefile.am:
* configure.ac:
add docs to build
* common/plugins.xsl:
wrap Description into a refsect2
* docs/Makefile.am:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-good-plugins-docs.sgml:
* docs/plugins/gst-plugins-good-plugins-sections.txt:
* gst/goom/Makefile.am:
* gst/goom/gstgoom.c: (gst_goom_get_type), (gst_goom_base_init),
(gst_goom_class_init), (gst_goom_init), (gst_goom_dispose),
(gst_goom_sink_setcaps), (gst_goom_src_setcaps),
(gst_goom_src_negotiate), (gst_goom_event), (gst_goom_chain),
(gst_goom_change_state):
* gst/goom/gstgoom.h:
GstGOOM -> GstGoom
add an example launch line
* gst/level/gstlevel.h:
* gst/monoscope/gstmonoscope.c:
cleanups
2005-09-01 17:55:14 +00:00
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2005-09-01 Thomas Vander Stichele <thomas at apestaart dot org>
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* Makefile.am:
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* configure.ac:
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add docs to build
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* common/plugins.xsl:
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wrap Description into a refsect2
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* docs/Makefile.am:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* gst/goom/Makefile.am:
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* gst/goom/gstgoom.c: (gst_goom_get_type), (gst_goom_base_init),
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(gst_goom_class_init), (gst_goom_init), (gst_goom_dispose),
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(gst_goom_sink_setcaps), (gst_goom_src_setcaps),
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(gst_goom_src_negotiate), (gst_goom_event), (gst_goom_chain),
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(gst_goom_change_state):
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* gst/goom/gstgoom.h:
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GstGOOM -> GstGoom
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add an example launch line
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* gst/level/gstlevel.h:
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* gst/monoscope/gstmonoscope.c:
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cleanups
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2005-08-30 19:01:18 +00:00
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2005-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
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* configure.ac:
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* docs/plugins/Makefile.am:
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* docs/plugins/gst-plugins-good-plugins-docs.sgml:
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* docs/plugins/gst-plugins-good-plugins-sections.txt:
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* docs/plugins/gst-plugins-good-plugins.types:
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document elements and plugins. Shazam !
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2005-08-30 17:32:12 +00:00
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2005-08-30 Thomas Vander Stichele <thomas at apestaart dot org>
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2003-12-16 16:58:59 +00:00
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2005-08-30 17:32:12 +00:00
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* split off gst-plugins-good from gst-plugins. Old ChangeLog can
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be found under doc/random/ChangeLog-0.8
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