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114 commits

Author SHA1 Message Date
Thomas Vander Stichele d9d1b4a934 some documentation for audioconvert
Original commit message from CVS:
some documentation for audioconvert
2005-09-23 14:41:31 +00:00
Wim Taymans b6bc76642a gst/audioconvert/gstaudioconvert.c: And enable 24 bits mode as well..
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
And enable 24 bits mode as well..
2005-09-15 13:52:27 +00:00
Tim-Philipp Müller 32f976bfea gst/audioconvert/Makefile.am: Audioconvert derives from GstBaseTransform and should link to the library with our base...
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
Audioconvert derives from GstBaseTransform and should
link to the library with our base elements to avoid
unresolved symbols. Makes things work with MinGW (#316160)
* gst/playback/test4.c: (main):
Fix MinGW build problem and use g_usleep() instead of
sleep() (#316162)
2005-09-13 13:52:59 +00:00
Wim Taymans 1237e1e701 gst/audioconvert/audioconvert.*: Cleanups, speedups, simplifications, added back support for 24 bits.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (float),
(audio_convert_prepare_context), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
Cleanups, speedups, simplifications, added back support
for 24 bits.
2005-09-12 11:38:05 +00:00
Jan Schmidt 0f4fa24d8e check/: Add extra tests for basetransform based components.
Original commit message from CVS:
* check/Makefile.am:
* check/pipelines/simple_launch_lines.c: (setup_pipeline),
(run_pipeline), (GST_START_TEST), (simple_launch_lines_suite):
Add extra tests for basetransform based components.
Comment out the test_element_negotiation test until we decide
if it's testing correct behaviour.
* ext/libvisual/visual.c: (gst_visual_init), (get_buffer),
(gst_visual_chain), (gst_visual_change_state):
Slightly more correct but still bogus timestamping.
Fix state change function.
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init):
* gst/audioresample/gstaudioresample.c:
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_class_init):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_class_init),
(gst_videoscale_prepare_size), (gst_videoscale_set_caps),
(gst_videoscale_prepare_image):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform_ip):
Basetransform updates. Enable passthrough modes.
* sys/ximage/ximagesink.c: (gst_ximage_buffer_init),
(gst_ximagesink_renegotiate_size), (gst_ximagesink_xcontext_get),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
Negotiation fix that allows the window to return to the original
size and renegotiate passthrough upstream. Extra debug output.
2005-09-09 17:53:47 +00:00
Thomas Vander Stichele a23795d4d3 disable 24 bit until it gets fixed
Original commit message from CVS:
disable 24 bit until it gets fixed
2005-09-02 23:16:15 +00:00
Andy Wingo 6665c3084c All plugins updated for element state changes.
Original commit message from CVS:
2005-09-02  Andy Wingo  <wingo@pobox.com>

* All plugins updated for element state changes.
2005-09-02 15:43:18 +00:00
Andy Wingo c32721723b Updates for two-arg init from GST_BOILERPLATE_FULL.
Original commit message from CVS:
2005-08-28  Andy Wingo  <wingo@pobox.com>

* Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-28 17:52:45 +00:00
Wim Taymans b6c368ce67 gst/audioconvert/audioconvert.c: Cleanups.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
Cleanups.
2005-08-26 18:57:30 +00:00
Wim Taymans ddec57c089 gst/audioconvert/audioconvert.c: More elegant and working temp buffer selection algo.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(audio_convert_convert):
More elegant and working temp buffer selection algo.
2005-08-26 18:43:02 +00:00
Wim Taymans 123aa7de1a gst/audioconvert/audioconvert.c: Use realloc else we lose our original data.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
Use realloc else we lose our original data.
2005-08-26 17:46:45 +00:00
Wim Taymans 98fbd82d1c gst/audioconvert/: Oops, allocate enough space to perform the channel mix.
Original commit message from CVS:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_parse_caps), (gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_transform):
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_mix):
Oops, allocate enough space to perform the channel mix.
2005-08-26 17:30:41 +00:00
Wim Taymans ceb84de916 gst/audioconvert/: Cleanups, librarify a bit, optimize, better negotiation and more.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/audioconvert.c: (if), (float),
(audio_convert_get_func_index), (check_default),
(audio_convert_clean_fmt), (audio_convert_prepare_context),
(audio_convert_clean_context), (audio_convert_get_sizes),
(get_temp_buffer), (audio_convert_convert):
* gst/audioconvert/audioconvert.h:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_parse_caps),
(gst_audio_convert_get_unit_size),
(gst_audio_convert_transform_caps),
(gst_audio_convert_fixate_caps), (gst_audio_convert_set_caps),
(gst_audio_convert_transform_ip), (gst_audio_convert_transform):
* gst/audioconvert/gstaudioconvert.h:
* gst/audioconvert/gstchannelmix.c: (gst_channel_mix_unset_matrix),
(gst_channel_mix_fill_identical),
(gst_channel_mix_fill_compatible), (gst_channel_mix_detect_pos),
(gst_channel_mix_fill_one_other), (gst_channel_mix_fill_others),
(gst_channel_mix_fill_normalize), (gst_channel_mix_fill_matrix),
(gst_channel_mix_setup_matrix), (gst_channel_mix_passthrough),
(gst_channel_mix_mix):
* gst/audioconvert/gstchannelmix.h:
Cleanups, librarify a bit, optimize, better negotiation and more.
2005-08-26 15:43:56 +00:00
Thomas Vander Stichele 43332aed85 plug some leaks
Original commit message from CVS:
plug some leaks
2005-08-25 17:32:34 +00:00
Thomas Vander Stichele 41a43b86a8 port audioconvert to basetransform fix ffmpegcsp and videoscale for basetransform changes
Original commit message from CVS:
port audioconvert to basetransform
fix ffmpegcsp and videoscale for basetransform changes
2005-08-24 13:32:52 +00:00
Thomas Vander Stichele 3e589b0977 rename plugin
Original commit message from CVS:
rename plugin
2005-08-15 14:15:40 +00:00
Thomas Vander Stichele e571f069d1 renamed to actual element names, so much nicer to look at
Original commit message from CVS:

* common/gtk-doc-plugins.mak:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
renamed to actual element names, so much nicer to look at
* docs/plugins/tmpl/gstmultifdsink.sgml:
remove
* docs/plugins/tmpl/multifdsink.sgml:
* docs/plugins/tmpl/tcpserversink.sgml:
add
* ext/alsa/gstalsa.c:
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_get_property):
* ext/ogg/gstoggmux.c:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain):
* gst/playback/gstdecodebin.c:
* gst/tcp/gstmultifdsink.c: (gst_multifdsink_class_init):
* gst/tcp/gsttcpserversink.c:
various fixes and documentation additions
2005-08-05 18:51:29 +00:00
Wim Taymans a6d89f51fa gst/audioconvert/gstaudioconvert.c: Convert me to BaseTransform!! help..
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link_src):
Convert me to BaseTransform!! help..
2005-07-29 17:07:39 +00:00
Wim Taymans 567802ca2c gst/audioconvert/gstaudioconvert.c: Timestamp buffers correctly.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
Timestamp buffers correctly.

* gst/playback/gstplaybin.c: (gen_video_element):
Make internal fakesink silent.
2005-07-16 13:58:21 +00:00
Thomas Vander Stichele 1ea0574af4 make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be added manually to each Makefile.am so we are sure it goes
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
2005-07-13 17:58:07 +00:00
Wim Taymans a46a991d26 ext/theora/theoradec.c: Prepare for better timestamp fix later.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_getcaps),
(theora_dec_push), (theora_handle_data_packet):
Prepare for better timestamp fix later.

* gst/audioconvert/gstaudioconvert.c:
List most accurate caps first

* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_loop):
Use proper pad task function.

* sys/xvimage/xvimagesink.c: (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_show_frame):
Fix deadlock when alloc failed.
2005-07-06 15:14:38 +00:00
Andy Wingo 3a7f5a2e06 gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate): No refcount leakage.
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/gstaudioconvert.c (gst_audio_convert_fixate):
No refcount leakage.
2005-07-04 10:40:17 +00:00
Andy Wingo 1f40231de5 configure.ac: Enable -Werror.
Original commit message from CVS:
2005-07-04  Andy Wingo  <wingo@pobox.com>

* configure.ac: Enable -Werror.

* ext/theora/theoradec.c (theora_dec_src_getcaps):
* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate):
* gst/audioconvert/gstaudioconvert.c (_fixate_caps_to_int)
(gst_audio_convert_fixate):
* gst/sine/gstsinesrc.c (gst_sinesrc_src_fixate)
(gst_sinesrc_create): Fixate func changes.

* sys/ximage/ximagesink.c: (gst_ximagesink_renegotiate_size),
(gst_ximagesink_buffer_alloc): Unused var.
2005-07-04 10:37:25 +00:00
Andy Wingo d0bc038021 *.c: Don't cast to GstObject before reffing/unreffing.
Original commit message from CVS:
2005-06-28  Andy Wingo  <wingo@pobox.com>

* *.c: Don't cast to GstObject before reffing/unreffing.
2005-06-28 10:16:13 +00:00
Andy Wingo 55d437af1c gst/: Ghost pad API fixes.
Original commit message from CVS:
2005-06-09  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/gconf/gconf.c:
* gst/playback/test.c:
* gst/playback/gstplaybin.c (gen_video_element): Ghost pad API
fixes.

* gst/audioconvert/gstaudioconvert.c: RPAD fixes.

* ext/theora/theoraenc.c (theora_enc_chain):
* ext/theora/theoradec.c (theora_handle_data_packet): GCC4 fixes.

* ext/ogg/gstoggdemux.c (GstOggPad): Derive from GstPad, not
RealPad.
2005-06-08 22:18:05 +00:00
Wim Taymans 6db639089a Cleanups and buffer alloc.
Original commit message from CVS:
* ext/ogg/README:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_submit_packet),
(gst_ogg_demux_activate_chain), (gst_ogg_demux_clear_chains):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page):
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_data_packet):
* gst/audioconvert/bufferframesconvert.c:
(buffer_frames_convert_chain):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_bufferalloc),
(gst_ffmpegcsp_chain):
* gst/videorate/gstvideorate.c: (gst_videorate_transformcaps),
(gst_videorate_getcaps), (gst_videorate_setcaps),
(gst_videorate_event), (gst_videorate_chain):
* gst/videotestsrc/gstvideotestsrc.c: (gst_videotestsrc_activate),
(gst_videotestsrc_src_query), (gst_videotestsrc_loop):
* sys/ximage/ximagesink.c: (gst_ximagesink_ximage_new),
(gst_ximagesink_setcaps), (gst_ximagesink_buffer_alloc):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
(gst_xvimagesink_xvimage_new), (gst_xvimagesink_xvimage_put),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc):
Cleanups and buffer alloc.
2005-06-02 09:46:40 +00:00
Wim Taymans 5474600d4f gst-libs/gst/audio/: Various small cleanups.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_get_type), (gst_audioringbuffer_class_init),
(audioringbuffer_thread_func), (gst_audioringbuffer_init),
(gst_audioringbuffer_acquire), (gst_audioringbuffer_release),
(gst_audioringbuffer_play), (gst_audioringbuffer_stop),
(gst_audioringbuffer_delay), (gst_audiosink_class_init),
(gst_audiosink_create_ringbuffer):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_init),
(gst_baseaudiosink_get_clock), (gst_baseaudiosink_get_time),
(gst_baseaudiosink_set_property), (build_linear_format),
(debug_spec_caps), (debug_spec_buffer),
(gst_baseaudiosink_setcaps), (gst_baseaudiosink_get_times),
(gst_baseaudiosink_event), (gst_baseaudiosink_preroll),
(gst_baseaudiosink_render), (gst_baseaudiosink_create_ringbuffer),
(gst_baseaudiosink_callback), (gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_get_type),
(gst_ringbuffer_set_callback), (gst_ringbuffer_acquire),
(gst_ringbuffer_release), (gst_ringbuffer_is_acquired),
(gst_ringbuffer_play), (gst_ringbuffer_pause),
(gst_ringbuffer_stop), (gst_ringbuffer_delay),
(gst_ringbuffer_played_samples), (gst_ringbuffer_set_sample),
(wait_segment), (gst_ringbuffer_commit),
(gst_ringbuffer_prepare_read), (gst_ringbuffer_advance),
(gst_ringbuffer_clear):
Various small cleanups.

* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_change_state):
* gst/subparse/gstsubparse.c: (gst_subparse_chain):
No need to take the locks anymore.
2005-05-25 19:52:14 +00:00
Ronald S. Bultje 78016b40cf gst/audioconvert/gstaudioconvert.c: Implement instant setup switching.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_chain), (gst_audio_convert_link_src),
(gst_audio_convert_setcaps):
Implement instant setup switching.
2005-05-23 17:28:02 +00:00
Wim Taymans 04fa67937d Leak fixes in oggdemux.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_internal_chain),
(gst_ogg_demux_submit_buffer), (gst_ogg_demux_get_data),
(gst_ogg_demux_chain_unlocked):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_change_state):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_init),
(gst_ffmpegcsp_bufferalloc), (gst_ffmpegcsp_chain),
(gst_ffmpegcsp_change_state), (gst_ffmpegcsp_set_property),
(gst_ffmpegcsp_get_property):
* sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_destroy),
(gst_xvimage_buffer_finalize), (gst_xvimage_buffer_free),
(gst_xvimage_buffer_class_init), (gst_xvimage_buffer_get_type),
(gst_xvimagesink_check_xshm_calls), (gst_xvimagesink_xvimage_new),
(gst_xvimagesink_xvimage_put), (gst_xvimagesink_imagepool_clear),
(gst_xvimagesink_setcaps), (gst_xvimagesink_change_state),
(gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_free),
(gst_xvimagesink_buffer_alloc), (gst_xvimagesink_set_xwindow_id):
Leak fixes in oggdemux.
Some cleanups in audioconvert.
Make passthrough work along with buffer_alloc etc.
Make buffer_alloc and buffer recycling actually work in
xvimagesink.
2005-05-17 17:41:32 +00:00
David Schleef d90ee5bfa3 Port from GstData to GstMiniObject.
Original commit message from CVS:
Port from GstData to GstMiniObject.
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_dispose):
* ext/ogg/gstoggmux.c: (gst_ogg_mux_buffer_from_page),
(gst_ogg_mux_queue_pads), (gst_ogg_mux_set_header_on_caps),
(gst_ogg_mux_collected):
* ext/ogg/gstogmparse.c: (gst_ogm_parse_chain):
* ext/theora/theoradec.c: (theora_handle_comment_packet),
(theora_handle_data_packet):
* ext/theora/theoraenc.c: (theora_buffer_from_packet),
(theora_set_header_on_caps), (theora_enc_chain):
* ext/vorbis/vorbisdec.c: (vorbis_dec_sink_event),
(vorbis_handle_comment_packet):
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_set_header_on_caps):
* ext/vorbis/vorbisparse.c: (vorbis_parse_set_header_on_caps):
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_chain):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_chain):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_get_buffer):
* gst/playback/gstdecodebin.c: (gst_decode_bin_class_init):
* gst/playback/gstplaybasebin.c: (check_queue), (probe_triggered),
(mute_stream), (silence_stream):
* gst/playback/gstplaybin.c: (gst_play_bin_class_init):
* gst/volume/gstvolume.c: (volume_transform):
* sys/ximage/ximagesink.c: (gst_ximage_buffer_finalize),
(gst_ximage_buffer_init), (gst_ximage_buffer_class_init),
(gst_ximage_buffer_get_type), (gst_ximagesink_check_xshm_calls),
(gst_ximagesink_ximage_new), (gst_ximagesink_ximage_destroy),
(gst_ximagesink_ximage_put), (gst_ximagesink_imagepool_clear),
(gst_ximagesink_show_frame), (gst_ximagesink_buffer_free),
(gst_ximagesink_buffer_alloc):
* sys/ximage/ximagesink.h:
2005-05-16 15:35:52 +00:00
Wim Taymans 8aeaf8ed14 Make caps writable before writing to it.
Original commit message from CVS:
* ext/vorbis/vorbisenc.c: (gst_vorbisenc_sink_setcaps),
(gst_vorbisenc_src_query), (gst_vorbisenc_sink_query),
(gst_vorbisenc_set_header_on_caps), (gst_vorbisenc_sink_event),
(gst_vorbisenc_chain):
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
Make caps writable before writing to it.
Fix negotiation in audioconvert some more.
2005-05-09 17:07:27 +00:00
Wim Taymans 3ec8704c9e Fixed negotiation wrt _peer_get_caps()
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_getcaps), (gst_audio_convert_setcaps),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_getcaps), (gst_ffmpegcsp_configure_context),
(gst_ffmpegcsp_setcaps), (gst_ffmpegcsp_chain):
* gst/sine/Makefile.am:
* gst/sine/gstsinesrc.c: (gst_sinesrc_get_type),
(gst_sinesrc_class_init), (gst_sinesrc_init),
(gst_sinesrc_src_fixate), (gst_sinesrc_setcaps),
(gst_sinesrc_src_query), (gst_sinesrc_create), (gst_sinesrc_start),
(gst_sinesrc_update_freq):
* gst/sine/gstsinesrc.h:
* gst/tcp/gstmultifdsink.c:
* sys/xvimage/xvimagesink.c:
Fixed negotiation wrt _peer_get_caps()
Some cleanups.
2005-05-06 17:13:49 +00:00
Andy Wingo 790c059867 gst/: Some GCC4 fixes
Original commit message from CVS:
2005-05-05  Andy Wingo  <wingo@pobox.com>

* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_from_vorbiscomment_buffer), (gst_vorbis_tag_chain):
* gst/adder/gstadder.h:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_one_other):
* gst/audiorate/gstaudiorate.c: (gst_audiorate_setcaps),
(gst_audiorate_init), (gst_audiorate_chain):
* gst/playback/gstplaybasebin.c: (setup_source):
* gst/playback/test3.c: (update_scale):
Some GCC4 fixes

* 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: Foo
2005-05-05 14:57:20 +00:00
Wim Taymans 3e5febcdd1 gst/: Don't ignore _push() return values.
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_change_state), (gst_audio_convert_channels):
* gst/videotestsrc/gstvideotestsrc.c:
(gst_videotestsrc_src_negotiate), (gst_videotestsrc_src_link),
(gst_videotestsrc_parse_caps), (gst_videotestsrc_src_accept_caps),
(gst_videotestsrc_setcaps), (gst_videotestsrc_activate),
(gst_videotestsrc_init), (gst_videotestsrc_loop):
Don't ignore _push() return values.
Make sure no processing is done when shutting down.
Videotestsrc pad activation fix.
2005-05-05 09:49:08 +00:00
Thomas Vander Stichele 37822dc3fb Fix part of the build. Come on guys, autogen didn't even work :)
Original commit message from CVS:
Fix part of the build.  Come on guys, autogen didn't even work :)
2005-04-25 13:29:40 +00:00
David Schleef 129c7e8af1 configure.ac: Remove idct and resample libs
Original commit message from CVS:
* configure.ac: Remove idct and resample libs
* gst-libs/gst/Makefile.am: same
Remove usage of gst_library_load():
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/libvisual/visual.c: (plugin_init):
* ext/ogg/gstogg.c: (plugin_init):
* ext/theora/theora.c: (plugin_init):
* ext/vorbis/vorbis.c: (plugin_init):
* gst-libs/gst/audio/gstaudiofiltertemplate.c: (plugin_init):
* gst/audioscale/gstaudioscale.c:
* gst/adder/gstadder.c: (plugin_init):
* gst/audioconvert/plugin.c: (plugin_init):
* sys/ximage/ximagesink.c: (plugin_init):
* sys/xvimage/xvimagesink.c: (plugin_init):
* gst/tcp/gsttcpplugin.c: (plugin_init):
Link plugins against libraries:
* ext/ogg/Makefile.am:
* ext/theora/Makefile.am:
* ext/vorbis/Makefile.am:
* gst/audioconvert/Makefile.am:
Create proper libraries:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/video/Makefile.am:
Move resample library to audioscale plugin directory:
* gst-libs/gst/resample/Makefile.am:
* gst-libs/gst/resample/README:
* gst-libs/gst/resample/dtof.c:
* gst-libs/gst/resample/dtos.c:
* gst-libs/gst/resample/functable.c:
* gst-libs/gst/resample/private.h:
* gst-libs/gst/resample/resample.c:
* gst-libs/gst/resample/resample.h:
* gst-libs/gst/resample/resample.vcproj:
* gst-libs/gst/resample/test.c:
* gst/audioscale/Makefile.am:
* gst/audioscale/README:
* gst/audioscale/dtof.c:
* gst/audioscale/dtos.c:
* gst/audioscale/functable.c:
* gst/audioscale/private.h:
* gst/audioscale/resample.c:
* gst/audioscale/resample.h:
* gst/audioscale/test.c:
Move tagedit library to gst-libs:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstid3tag.c:
* gst-libs/gst/tag/gsttagediting.c:
* gst-libs/gst/tag/gsttageditingprivate.h:
* gst-libs/gst/tag/gstvorbistag.c:
* gst/tags/Makefile.am:
* gst/tags/gstid3tag.c:
* gst/tags/gstvorbistag.c:
Fix for core changes:
* gst/sine/gstsinesrc.c: (gst_sinesrc_class_init),
(gst_sinesrc_init), (gst_sinesrc_src_fixate), (gst_sinesrc_link),
(gst_sinesrc_getrange):
2005-04-25 00:23:06 +00:00
Wim Taymans 468d6d4326 gst/audioconvert/: Link against audio libs.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_chain),
(gst_audio_convert_caps_remove_format_info),
(gst_audio_convert_setcaps), (gst_audio_convert_fixate),
(gst_audio_convert_channels):
Link against audio libs.
Fix audio convert plugin.
2005-04-20 09:45:33 +00:00
Wim Taymans 1dae961cbf Plugin port to 0.9, ogg/theora playback should work in the seek example now.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
2005-03-31 09:43:49 +00:00
Tim-Philipp Müller 797a080e39 Don't leak othercaps in link function (fixes #167878)
Original commit message from CVS:
Don't leak othercaps in link function (fixes #167878)
2005-02-19 20:01:36 +00:00
Benjamin Otte c196377ff6 gst/audioconvert/gstaudioconvert.c: create channel conversion matrix when linking
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_link),
(gst_audio_convert_channels):
create channel conversion matrix when linking
* gst/audioconvert/.cvsignore:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/channelmixtest.c: (main):
add (ugly) test that ensures stereo <=> mono conversion works
correctly
2005-02-13 17:39:22 +00:00
Benjamin Otte f9cf10c748 gst/audioconvert/gstchannelmix.h: include missing header file
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.h:
include missing header file
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_compatible):
use same sign for both channels when converting to/from compatible
channel. Previously used different signs made the signals cancel
each other out and appear like silence. (fixes #167269)
2005-02-13 16:10:16 +00:00
Jan Schmidt 83e3fe189c configure.ac: Add dvdlpcmdec
Original commit message from CVS:

* configure.ac:
Add dvdlpcmdec

* ext/mpeg2dec/gstmpeg2dec.c: (gst_mpeg2dec_reset),
(free_all_buffers), (gst_mpeg2dec_alloc_buffer):
Don't push buffers if the src pad isn't negotiated yet.

* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_buffer_to_default_format),
(gst_audio_convert_buffer_from_default_format):
Add support for 24-bit width.

* gst/dvdlpcmdec/.cvsignore:
* gst/dvdlpcmdec/Makefile.am:
* gst/dvdlpcmdec/gstdvdlpcmdec.c: (gst_dvdlpcmdec_get_type),
(gst_dvdlpcmdec_base_init), (gst_dvdlpcmdec_class_init),
(gst_dvdlpcm_reset), (gst_dvdlpcmdec_init), (gst_dvdlpcmdec_link),
(gst_dvdlpcmdec_chain), (gst_dvdlpcmdec_change_state),
(plugin_init):
* gst/dvdlpcmdec/gstdvdlpcmdec.h:
New decoder for rearranging DVD LPCM into our audio/x-raw-int
format. Needs support for the channels maps if someone can find
a DVD LPCM track with > 2 channels.

* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_handle_dvd_event),
(gst_dvd_demux_send_discont), (gst_dvd_demux_handle_discont),
(gst_dvd_demux_get_audio_stream), (gst_dvd_demux_process_private):
* gst/mpegstream/gstdvddemux.h:
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_send_discont),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_init_stream),
(gst_mpeg_demux_send_subbuffer), (gst_mpeg_demux_handle_src_query):
* gst/mpegstream/gstmpegdemux.h:
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_reset),
(gst_mpeg_parse_parse_packhead), (gst_mpeg_parse_loop),
(gst_mpeg_parse_get_rate), (gst_mpeg_parse_convert_src),
(gst_mpeg_parse_handle_src_query),
(gst_mpeg_parse_handle_src_event):
Use audio/x-dvd-lpcm for LPCM output.
Add DTS output.
2005-02-08 11:08:15 +00:00
Andy Wingo 9c1c858d24 gst/audioconvert/bufferframesconvert.c
Original commit message from CVS:
2005-02-04  Andy Wingo  <wingo@pobox.com>

* gst/audioconvert/bufferframesconvert.c
(buffer_frames_convert_fixate): New function, fixates to 256
frames per buffer by default. (Much better than 1.)
(buffer_frames_convert_init): Set the fixate function for both src
and sink pad.
(buffer_frames_convert_link): After success setting nonfixed caps,
get the negotiated caps so we can know how many buffer-frames it
will be. No idea how this worked at all before.
2005-02-04 15:40:38 +00:00
Ronald S. Bultje 97dd32f748 gst/audioconvert/gstaudioconvert.c: The return value of fixate_to does not imply that the requested value was set, so...
Original commit message from CVS:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
The return value of fixate_to does not imply that the requested
value was set, so don't assume.
2005-01-07 18:17:52 +00:00
Thomas Vander Stichele c1e1206414 fix for glib < 2.4
Original commit message from CVS:
fix for glib < 2.4
2004-12-24 15:12:56 +00:00
Benjamin Otte 05103c18ca gst/audioconvert/gstchannelmix.c: more overwriting protection due to modifying channels one by one instead of all at ...
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
more overwriting protection due to modifying channels one by one
instead of all at once
2004-11-28 20:41:23 +00:00
Ronald S. Bultje 836ea71ea1 gst/audioconvert/gstchannelmix.c: Normalize using absolute values.
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_fill_normalize):
Normalize using absolute values.
2004-11-28 19:22:07 +00:00
Benjamin Otte 110fd90222 gst/audioconvert/gstchannelmix.c: walk the samples backwards if out_channels > in_channels so we don't overwrite data
Original commit message from CVS:
* gst/audioconvert/gstchannelmix.c: (gst_audio_convert_mix):
walk the samples backwards if out_channels > in_channels so we don't
overwrite data
2004-11-28 18:26:36 +00:00
Ronald S. Bultje 4fdf8d5b16 gst/audioconvert/: Implement a channel mixer.
Original commit message from CVS:
* gst/audioconvert/Makefile.am:
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_init),
(gst_audio_convert_link), (gst_audio_convert_change_state),
(gst_audio_convert_channels):
* gst/audioconvert/gstchannelmix.c:
(gst_audio_convert_unset_matrix),
(gst_audio_convert_fill_identical),
(gst_audio_convert_fill_compatible),
(gst_audio_convert_detect_pos), (gst_audio_convert_fill_one_other),
(gst_audio_convert_fill_others),
(gst_audio_convert_fill_normalize),
(gst_audio_convert_fill_matrix), (gst_audio_convert_setup_matrix),
(gst_audio_convert_passthrough), (gst_audio_convert_mix):
* gst/audioconvert/gstchannelmix.h:
Implement a channel mixer.
2004-11-28 16:09:13 +00:00
Ronald S. Bultje 1de3e19fdb ext/a52dec/gsta52dec.c: Don't do sample adjusting anymore, we use float audio now.
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_loop),
(gst_a52dec_change_state):
Don't do sample adjusting anymore, we use float audio now.
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_fixate):
Don't fixate to non-existing properties.
2004-11-27 20:22:42 +00:00