Commit graph

1833 commits

Author SHA1 Message Date
Wim Taymans
fe26e8d94c gst/rtp/gstrtpL16pay.c: Removed some unused code.
Original commit message from CVS:
* gst/rtp/gstrtpL16pay.c: (gst_rtp_L16_pay_handle_buffer):
Removed some unused code.
* gst/rtp/gstrtpamrpay.c: (gst_rtp_amr_pay_handle_buffer):
* gst/rtp/gstrtpgsmpay.c: (gst_rtp_gsm_pay_handle_buffer):
* gst/rtp/gstrtpmp2tpay.c: (gst_rtp_mp2t_pay_handle_buffer):
* gst/rtp/gstrtpspeexpay.c: (gst_rtp_speex_pay_handle_buffer):
* gst/rtp/gstrtptheorapay.c: (gst_rtp_theora_pay_init_packet),
(gst_rtp_theora_pay_flush_packet):
* gst/rtp/gstrtpvorbispay.c: (gst_rtp_vorbis_pay_flush_packet):
Try to preserve the incomming buffer duration on the outgoing
packets. Fixes #478244.
2007-09-19 16:24:09 +00:00
Stefan Kost
098c8faefb ChangeLog: Add missing newline.
Original commit message from CVS:
* ChangeLog:
Add missing newline.
* gst/librfb/rfbdecoder.c:
Fix the build (missing stdlib.h).
* gst/spectrum/gstspectrum.c:
* gst/spectrum/gstspectrum.h:
Use basetransform segment so that it is correctly managed on flushes
and start/stop. Report message timestamp as stream time, which is what
an application can understand. (Yes these are adapted from wim recent
level element changes)
2007-09-18 11:45:06 +00:00
Jan Schmidt
216f6e0593 gst/: Fix compiler warnings shown with Forte.
Original commit message from CVS:
* gst/audiofx/audiodynamic.c: (gst_audio_dynamic_class_init):
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_create_stream),
(new_session_pad), (request_pt_map), (gst_rtspsrc_do_stream_eos),
(gst_rtspsrc_loop_interleaved), (gst_rtspsrc_parse_rtpinfo),
(gst_rtspsrc_handle_message):
Fix compiler warnings shown with Forte.
2007-09-17 17:35:13 +00:00
Wim Taymans
7eb37e2575 gst/rtsp/gstrtspsrc.c: Give meaningfull error when all streams failed to configure for some reason.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_setup_streams),
(gst_rtspsrc_dup_printf):
Give meaningfull error when all streams failed to configure for some
reason.
2007-09-17 02:05:14 +00:00
Wim Taymans
e9f273126b gst/rtp/README: Update README with the design for synchronisation rules of RTP on sender and receiver.
Original commit message from CVS:
* gst/rtp/README:
Update README with the design for synchronisation rules of RTP on
sender and receiver.
2007-09-16 19:13:58 +00:00
Sebastian Dröge
233644df33 gst/wavparse/gstwavparse.c: Don't push EOS from the chain function, the element driving the pipeline is responsible f...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_loop),
(gst_wavparse_chain):
Don't push EOS from the chain function, the element
driving the pipeline is responsible for this. The bug
this was meant to fix seems to be queue not forwarding
EOS in all cases (see #476514).
2007-09-14 09:40:49 +00:00
Wim Taymans
80dc806b65 gst/level/gstlevel.*: Use basetransform segment so that it is correctly managed on flushes and start/stop.
Original commit message from CVS:
* gst/level/gstlevel.c: (gst_level_class_init), (gst_level_start),
(gst_level_transform_ip):
* gst/level/gstlevel.h:
Use basetransform segment so that it is correctly managed on flushes and
start/stop.
Report message timestamp as stream time, which is what an application
can understand.
2007-09-13 17:31:16 +00:00
Sebastian Dröge
d78b9e274b gst/wavparse/gstwavparse.c: Add EOS logic for the push-based mode too. Fixes #476514.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c: (gst_wavparse_perform_eos),
(gst_wavparse_loop), (gst_wavparse_chain):
Add EOS logic for the push-based mode too. Fixes #476514.
2007-09-13 12:37:56 +00:00
Wim Taymans
8a6f9aa51a gst/law/: Fix law encoder timestamps.
Original commit message from CVS:
* gst/law/alaw-encode.c: (gst_alawenc_init), (gst_alawenc_chain):
* gst/law/alaw-encode.h:
* gst/law/mulaw-encode.c: (gst_mulawenc_init),
(gst_mulawenc_chain):
* gst/law/mulaw-encode.h:
Fix law encoder timestamps.
2007-09-12 22:01:59 +00:00
Peter Kjellerstedt
eb2aee1b34 gst/: Printf format fixes (#476128).
Original commit message from CVS:
Patch by: Peter Kjellerstedt  <pkj at axis com>
* gst-libs/gst/app/gstappsink.c:
* gst/flv/gstflvdemux.c:
* gst/flv/gstflvparse.c:
* gst/interleave/deinterleave.c:
* gst/switch/gstswitch.c:
Printf format fixes (#476128).
2007-09-12 08:38:21 +00:00
Wim Taymans
4b25ca6267 gst/udp/gstudpsrc.c: Make udpsrc timestamp outgoing buffers based on when they were received.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_init), (gst_udpsrc_create):
Make udpsrc timestamp outgoing buffers based on when they were received.
Also make it output a segment in time.
2007-09-10 19:53:28 +00:00
Stefan Kost
2d15f70302 gst/avi/gstavidemux.c: Plug a little leak. Little code cleanups.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Plug a little leak. Little code cleanups.
2007-09-10 06:49:32 +00:00
Haakon Sporsheim
5e39863fca gst/rtp/gstrtph263pay.c: Fix up header structure so that compilers don't add padding between the structure fields, si...
Original commit message from CVS:
Patch by: Haakon Sporsheim  <haakon.sporsheim at tandberg com>
* gst/rtp/gstrtph263pay.c:
Fix up header structure so that compilers don't add padding
between the structure fields, since that would lead to us
sending RTP packets with broken headers (as is currently the
case when compiling with MSVC). Also see similar fixes in
libgstrtp in gst-plugins-base. (#474616; #471194)
2007-09-07 18:04:41 +00:00
Sebastian Dröge
1b98dfee5e gst/spectrum/gstspectrum.c: Use the correct parameter order for the memset calls.
Original commit message from CVS:
* gst/spectrum/gstspectrum.c: (gst_spectrum_start),
(gst_spectrum_transform_ip):
Use the correct parameter order for the memset calls.
Thanks to Christian Schaller for noticing.
2007-09-07 15:54:38 +00:00
Sebastian Dröge
f5a3e61e69 Port GstSpectrum to GstAudioFilter and libgstfft, add support for int32, float and double, use floats for the message...
Original commit message from CVS:
* configure.ac:
* gst/spectrum/Makefile.am:
* gst/spectrum/demo-audiotest.c: (draw_spectrum),
(message_handler), (main):
* gst/spectrum/demo-osssrc.c: (draw_spectrum), (message_handler):
* gst/spectrum/gstspectrum.c: (gst_spectrum_base_init),
(gst_spectrum_class_init), (gst_spectrum_init),
(gst_spectrum_dispose), (gst_spectrum_set_property),
(gst_spectrum_get_property), (gst_spectrum_start),
(gst_spectrum_setup), (gst_spectrum_message_new),
(gst_spectrum_transform_ip):
* gst/spectrum/gstspectrum.h:
Port GstSpectrum to GstAudioFilter and libgstfft, add support
for int32, float and double, use floats for the message contents,
average all FFTs done in one interval for better results, use
a better windowing function, allow posting the phase in the message
and actually do an FFT with the requested number of bands instead
of interpolating.
* tests/check/elements/spectrum.c: (GST_START_TEST),
(spectrum_suite):
Improve the units tests by checking for a 11025Hz sine wave
and add unit tests for all 4 supported sample types.
2007-09-06 07:21:22 +00:00
Tim-Philipp Müller
c8af2199d3 gst/qtdemux/: Don't assume tags are encoded as UTF-8 (#473670).
Original commit message from CVS:
* gst/qtdemux/Makefile.am:
* gst/qtdemux/qtdemux.c:
Don't assume tags are encoded as UTF-8 (#473670).
2007-09-05 16:23:21 +00:00
Wim Taymans
93e1176891 gst/udp/gstmultiudpsink.c: Add property do configure destination address/port pairs
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_set_clients_string),
(gst_multiudpsink_get_clients_string),
(gst_multiudpsink_set_property), (gst_multiudpsink_get_property),
(gst_multiudpsink_init_send), (gst_multiudpsink_add_internal),
(gst_multiudpsink_add), (gst_multiudpsink_clear_internal),
(gst_multiudpsink_clear):
Add property do configure destination address/port pairs
API:GstMultiUDPSink::clients
2007-09-04 22:42:21 +00:00
Stefan Kost
5248639cc1 gst/wavparse/gstwavparse.c: More code cleanups. Add some more comment and improve debugs logs.
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
More code cleanups. Add some more comment and improve debugs logs.
2007-09-04 14:37:22 +00:00
Stefan Kost
43b18b3f43 gst/wavparse/gstwavparse.*: Implement seek-query. Refactor duration calculations. Appropriate use of uint64_scale_int...
Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
* gst/wavparse/gstwavparse.h:
Implement seek-query. Refactor duration calculations. Appropriate use
of uint64_scale_int and uint64_scale. Move repeadedly calculated stuff
out of loops.
2007-09-04 07:58:36 +00:00
Stefan Kost
c1b2242e77 gst/avi/gstavidemux.c: Implement seek-query.
Original commit message from CVS:
* gst/avi/gstavidemux.c:
Implement seek-query.
2007-09-03 07:44:34 +00:00
Wim Taymans
14e218c083 gst/rtsp/gstrtspsrc.c: Use new basesink async property to make sparse RTCP packet not wait for preroll.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_stream_configure_udp_sink),
(gst_rtspsrc_dup_printf):
Use new basesink async property to make sparse RTCP packet not wait for
preroll.
2007-08-29 21:43:08 +00:00
Jan Schmidt
32621485d5 gst/audiofx/Makefile.am: Dist the right file.
Original commit message from CVS:
* gst/audiofx/Makefile.am:
Dist the right file.
2007-08-27 14:44:19 +00:00
Wim Taymans
a221e91936 gst/rtsp/gstrtspsrc.c: Make sure we generate and parse floating point values in the POSIX locale instead of the curre...
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_dup_printf),
(gst_rtspsrc_get_float), (gst_rtspsrc_play):
Make sure we generate and parse floating point values in the POSIX
locale instead of the current locale.
2007-08-23 16:27:36 +00:00
Wim Taymans
5592bdd459 gst/rtsp/gstrtspsrc.*: Fix method detection again.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_seek),
(gst_rtspsrc_parse_methods), (gst_rtspsrc_open),
(gst_rtspsrc_play):
* gst/rtsp/gstrtspsrc.h:
Fix method detection again.
Keep track of when we must send a Range header.
Use segment values for Range, Speed and Scale headers.
Parse Speed and Scale headers to update the segment values.
2007-08-22 15:01:29 +00:00
Wim Taymans
7d92376d3b 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
Sebastian Dröge
45ac408d0a gst/filter/: Add small comparision with the chebyshev filters in the docs.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstlpwsinc.c:
Add small comparision with the chebyshev filters in the docs.
2007-08-19 19:16:33 +00:00
Sebastian Dröge
5f32a4bac6 gst/audiofx/: Add small comparision with the windowed sinc filters in the docs.
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
* gst/audiofx/audiochebyshevfreqlimit.c:
Add small comparision with the windowed sinc filters in the docs.
2007-08-19 19:11:04 +00:00
Wim Taymans
60bf53248b gst/rtsp/gstrtspsrc.c: Refactor the udp and interleaved loop function a bit.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_loop_interleaved),
(gst_rtspsrc_loop_udp), (gst_rtspsrc_loop_send_cmd),
(gst_rtspsrc_loop):
Refactor the udp and interleaved loop function a bit.
2007-08-18 19:44:55 +00:00
Wim Taymans
0dcafb0635 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
Wim Taymans
4d581cb606 gst/debug/rndbuffersize.c: Fix debug statement.
Original commit message from CVS:
* gst/debug/rndbuffersize.c: (gst_rnd_buffer_size_loop):
Fix debug statement.
2007-08-17 15:30:39 +00:00
Wim Taymans
98fb7c070f gst/rtsp/gstrtspsrc.c: Fix stray %u in debug line as spotted by Saur on IRC.
Original commit message from CVS:
* gst/rtsp/gstrtspsrc.c: (gst_rtspsrc_do_stream_eos):
Fix stray %u in debug line as spotted by Saur on IRC.
2007-08-17 15:28:40 +00:00
Sebastian Dröge
1301d15e4f Use generator macros for the process functions for the different sample types, add lower upper boundaries for the GOb...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_set_property), (bpwsinc_get_property):
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (lpwsinc_build_kernel), (lpwsinc_set_property),
(lpwsinc_get_property):
* gst/filter/gstlpwsinc.h:
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and change frequency
properties to floats to save a bit of memory, even ints would in
theory be enough. Also rename frequency to cutoff for consistency
reasons.
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
Regenerated for the above changes.
2007-08-17 15:05:17 +00:00
Sebastian Dröge
f86bfaf5f9 gst/audiofx/: Use generator macros for the process functions for the different sample types, add lower upper boundari...
Original commit message from CVS:
* gst/audiofx/audiochebyshevfreqband.c:
(gst_audio_chebyshev_freq_band_class_init):
* gst/audiofx/audiochebyshevfreqlimit.c:
(gst_audio_chebyshev_freq_limit_class_init):
Use generator macros for the process functions for the different
sample types, add lower upper boundaries for the GObject properties
so automatically generated UIs can use sliders and add a note about
the number of poles as a too high number of poles combined with
very low or very high frequencies will produce only noise.
* docs/plugins/gst-plugins-good-plugins.args:
Regenerated for the property changes.
2007-08-17 14:43:33 +00:00
Wim Taymans
6ef7055041 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
Wim Taymans
2e599ab037 gst/udp/gstmultiudpsink.*: Add support for getting and setting the socket to use.
Original commit message from CVS:
* gst/udp/gstmultiudpsink.c: (gst_multiudpsink_class_init),
(gst_multiudpsink_init), (gst_multiudpsink_set_property),
(gst_multiudpsink_get_property), (gst_multiudpsink_init_send),
(gst_multiudpsink_close), (gst_multiudpsink_add):
* gst/udp/gstmultiudpsink.h:
Add support for getting and setting the socket to use.
* gst/udp/gstudpsrc.c: (gst_udpsrc_class_init), (gst_udpsrc_init),
(gst_udpsrc_create), (gst_udpsrc_get_property):
Add support for getting the currently used socket.
2007-08-17 13:59:15 +00:00
Sebastian Dröge
fc8a487616 gst/filter/gstbpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(gst_bpwsinc_init), (process_32), (process_64),
(bpwsinc_build_kernel), (bpwsinc_push_residue),
(bpwsinc_transform), (bpwsinc_start), (bpwsinc_query),
(bpwsinc_query_type), (bpwsinc_event), (bpwsinc_set_property):
* gst/filter/gstbpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/bpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Reset residue length only when actually creating a residue.
2007-08-16 19:22:48 +00:00
Sebastian Dröge
842451a720 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
Stefan Kost
22bcaa904c Make ro memory to share.
Original commit message from CVS:
* ext/annodex/gstcmmltag.c:
* gst/rtp/gstrtpvorbispay.c:
Make ro memory to share.
2007-08-16 12:15:06 +00:00
Wim Taymans
042d3a461c gst/udp/gstudpsrc.c: Improve UDP performance by avoiding a select() when we have data available immediatly.
Original commit message from CVS:
* gst/udp/gstudpsrc.c: (gst_udpsrc_create):
Improve UDP performance by avoiding a select() when we have data
available immediatly.
2007-08-16 11:49:01 +00:00
Wim Taymans
41f0496738 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
Sebastian Dröge
a490cffe5f gst/filter/gstlpwsinc.*: Implement latency query and only forward those samples downstream that actually contain the ...
Original commit message from CVS:
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(gst_lpwsinc_init), (process_32), (process_64),
(lpwsinc_build_kernel), (lpwsinc_push_residue),
(lpwsinc_transform), (lpwsinc_start), (lpwsinc_query),
(lpwsinc_query_type), (lpwsinc_event), (lpwsinc_set_property):
* gst/filter/gstlpwsinc.h:
Implement latency query and only forward those samples downstream
that actually contain the data we want, i.e. drop kernel_length/2
in the beginning and append kernel_length/2 (created by convolving
the filter kernel with zeroes) to the end.
* tests/check/elements/lpwsinc.c: (GST_START_TEST):
Adjust the unit test for this slightly changed behaviour.
2007-08-16 09:48:27 +00:00
Stefan Kost
647e2dd7c0 gst/debug/rndbuffersize.c: Fix da leak.
Original commit message from CVS:
* gst/debug/rndbuffersize.c:
Fix da leak.
2007-08-16 07:40:48 +00:00
Stefan Kost
e949d1989b gst/debug/: Add new test element and clean-up the others a little.
Original commit message from CVS:
* gst/debug/Makefile.am:
* gst/debug/breakmydata.c:
* gst/debug/gstdebug.c:
* gst/debug/negotiation.c:
* gst/debug/progressreport.c:
* gst/debug/rndbuffersize.c:
* gst/debug/testplugin.c:
Add new test element and clean-up the others a little.
2007-08-14 13:50:43 +00:00
Sebastian Dröge
f944834a11 Add docs for lpwsinc and bpwsinc and integrate them into the build system. While doing that also update all other doc...
Original commit message from CVS:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-bad-plugins-docs.sgml:
* docs/plugins/gst-plugins-bad-plugins-sections.txt:
* docs/plugins/gst-plugins-bad-plugins.args:
* docs/plugins/gst-plugins-bad-plugins.signals:
* docs/plugins/inspect/plugin-bz2.xml:
* docs/plugins/inspect/plugin-cdxaparse.xml:
* docs/plugins/inspect/plugin-dtsdec.xml:
* docs/plugins/inspect/plugin-faac.xml:
* docs/plugins/inspect/plugin-faad.xml:
* docs/plugins/inspect/plugin-filter.xml:
* docs/plugins/inspect/plugin-freeze.xml:
* docs/plugins/inspect/plugin-gsm.xml:
* docs/plugins/inspect/plugin-gstrtpmanager.xml:
* docs/plugins/inspect/plugin-h264parse.xml:
* docs/plugins/inspect/plugin-modplug.xml:
* docs/plugins/inspect/plugin-mpeg2enc.xml:
* docs/plugins/inspect/plugin-musepack.xml:
* docs/plugins/inspect/plugin-musicbrainz.xml:
* docs/plugins/inspect/plugin-nsfdec.xml:
* docs/plugins/inspect/plugin-replaygain.xml:
* docs/plugins/inspect/plugin-soundtouch.xml:
* docs/plugins/inspect/plugin-spcdec.xml:
* docs/plugins/inspect/plugin-spectrum.xml:
* docs/plugins/inspect/plugin-speed.xml:
* docs/plugins/inspect/plugin-tta.xml:
* docs/plugins/inspect/plugin-videosignal.xml:
* docs/plugins/inspect/plugin-xingheader.xml:
* docs/plugins/inspect/plugin-xvid.xml:
* gst/filter/gstbpwsinc.c:
* gst/filter/gstbpwsinc.h:
* gst/filter/gstlpwsinc.c:
* gst/filter/gstlpwsinc.h:
Add docs for lpwsinc and bpwsinc and integrate them
into the build system. While doing that also update
all other docs via make update in docs/plugins.
2007-08-13 13:50:39 +00:00
Sebastian Dröge
e8030a1356 gst/filter/: Improve debugging a bit.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (bpwsinc_build_kernel):
* gst/filter/gstlpwsinc.c: (lpwsinc_build_kernel):
Improve debugging a bit.
2007-08-12 15:41:57 +00:00
Wim Taymans
39321cf1f7 gst/qtdemux/qtdemux.c: Fix parsing of mp4a version 0 atoms. Fixes #465774.
Original commit message from CVS:
* gst/qtdemux/qtdemux.c: (qtdemux_parse_node):
Fix parsing of mp4a version 0 atoms. Fixes #465774.
2007-08-12 14:35:41 +00:00
Sebastian Dröge
a1c029bab5 gst/filter/: Reset the residue in BaseTransform::start to get a clean residue on stream changes.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (gst_bpwsinc_class_init),
(bpwsinc_start):
* gst/filter/gstlpwsinc.c: (gst_lpwsinc_class_init),
(lpwsinc_start):
Reset the residue in BaseTransform::start to get a clean residue
on stream changes.
2007-08-12 12:46:20 +00:00
Sebastian Dröge
6871d561db gst/filter/: Fix processing with buffer sizes that are larger than the filter kernel size.
Original commit message from CVS:
* gst/filter/gstbpwsinc.c: (process_32), (process_64):
* gst/filter/gstlpwsinc.c: (process_32), (process_64):
Fix processing with buffer sizes that are larger than the filter
kernel size.
2007-08-11 15:58:30 +00:00
Stefan Kost
6260b45a1a gst/rtp/gstrtpilbcdepay.c: Include stdlib.
Original commit message from CVS:
* gst/rtp/gstrtpilbcdepay.c:
Include stdlib.
2007-08-10 17:08:01 +00:00
Wim Taymans
e640bc6a4b gst/rtp/gstrtpmpvdepay.c: Set the mpegversion in the caps so that autoplugging does not get confused.
Original commit message from CVS:
* gst/rtp/gstrtpmpvdepay.c:
Set the mpegversion in the caps so that autoplugging does not get
confused.
2007-08-10 16:10:47 +00:00