- 25 Nov, 2008 5 commits
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Wim Taymans authored
Original commit message from CVS: * gst-libs/gst/rtp/gstrtcpbuffer.c: Fix typo in the docs.
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Wim Taymans authored
Original commit message from CVS: * ext/ogg/gstoggdemux.c: (gst_ogg_demux_sink_event): If no stream was found before receiving EOS, post an error message. Fixes #561924.
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Wim Taymans authored
Original commit message from CVS: * ext/theora/gsttheoraenc.h: * ext/theora/theoraenc.c: (gst_theora_enc_init), (theora_buffer_from_packet), (theora_push_packet), (theora_enc_sink_event), (theora_enc_is_discontinuous), (theora_enc_chain): Parse segment events. Pass incomming buffer timestamps to outgoing buffers. Use the running_time to construct the granulepos. Fixes #562163.
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Wim Taymans authored
Original commit message from CVS: * gst/playback/gstplaybin2.c: (activate_group): Fix buffer-duration property.
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Wim Taymans authored
gst-libs/gst/audio/gstbaseaudiosink.c: Really fix audiosink drain handling by keeping track of the running_time of th... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_drain), (gst_base_audio_sink_event), (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): Really fix audiosink drain handling by keeping track of the running_time of the last sample.
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- 24 Nov, 2008 6 commits
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Michael Smith authored
gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes. Original commit message from CVS: * gst/playback/gstplaybin2.c: Add notification of current stream. Add ability to configure buffer sizes. * gst/playback/gsturidecodebin.c: Add ability to configure buffer sizes for streaming mode. Bug #561734.
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Stefan Kost authored
gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks n... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: Time is already in running_time. Remove base_time handling. Fixes audiosinks not draining and thus chopping some audio in the end.
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David Schleef authored
Original commit message from CVS: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggmux.h: If we're muxing a dirac stream, flush the page after every picture.
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Stefan Kost authored
gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the ... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: Add one log message to check for audio_drained. Sync one log message with the condition. Send EOS after draining audio in pull mode.
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Sebastian Dröge authored
ext/: Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstr... Original commit message from CVS: * ext/gio/gstgiobasesrc.c: (gst_gio_base_src_create): * ext/gnomevfs/gstgnomevfssrc.c: (gst_gnome_vfs_src_create): Use gst_buffer_try_new_and_alloc() and fail properly if the allocation failed. This prevents abort() if downstream elements request an insane amount of memory.
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Jon Trowbridge authored
Original commit message from CVS: * gst/volume/gstvolume.c: (volume_choose_func), (volume_update_volume), (gst_volume_set_volume), (gst_volume_get_volume), (gst_volume_set_mute), (gst_volume_class_init), (gst_volume_init), (volume_process_double), (volume_process_float), (volume_process_int32), (volume_process_int32_clamp), (volume_process_int24), (volume_process_int24_clamp), (volume_process_int16), (volume_process_int16_clamp), (volume_process_int8), (volume_process_int8_clamp), (volume_setup), (volume_transform_ip), (volume_set_property), (volume_get_property): * gst/volume/gstvolume.h: Cleanup volume, define and use default values. Recalculate new volume and mute setup before processing. Fixes #561789. * tests/check/elements/volume.c: (GST_START_TEST), (volume_suite): Add controller unit test. Patch by: Jonathan Matthew Fix bogus test that messed with basetransform's internal state.
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- 22 Nov, 2008 3 commits
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Sebastian Dröge authored
tests/check/elements/speexresample.c: Make the unit test a bit faster to prevent timeouts, especially with valgrind. Original commit message from CVS: * tests/check/elements/speexresample.c: (GST_START_TEST): Make the unit test a bit faster to prevent timeouts, especially with valgrind.
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Wim Taymans authored
Original commit message from CVS: * gst/videorate/gstvideorate.c: Add jpeg and png image media types to the caps. Fixes #561436.
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Wim Taymans authored
gst/playback/gstplaysink.c: Don't post an error when we can't configure the volume but post a warning instead. Fixes ... Original commit message from CVS: * gst/playback/gstplaysink.c: (gen_audio_chain): Don't post an error when we can't configure the volume but post a warning instead. Fixes #561780.
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- 21 Nov, 2008 3 commits
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Jonathan Rosser authored
gst/videotestsrc/: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'video... Original commit message from CVS: Patch by: Jonathan Rosser <jonathan.rosser@rd.bbc.co.uk> * gst/videotestsrc/gstvideotestsrc.c: * gst/videotestsrc/gstvideotestsrc.h: * gst/videotestsrc/videotestsrc.c: * gst/videotestsrc/videotestsrc.h: Add a zone plate pattern generator based on BBC R&D Report 1978/23 (yeah *that* 1978). Try 'videotestsrc pattern=zone-plate kx2=20 ky2=20 kt=1'.
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Sebastian Dröge authored
gst/speexresample/gstspeexresample.c: Add a "filter-length" property that maps to the quality values for compatibilty... Original commit message from CVS: * gst/speexresample/gstspeexresample.c: (gst_speex_resample_class_init), (gst_speex_resample_set_property), (gst_speex_resample_get_property): Add a "filter-length" property that maps to the quality values for compatibilty with audioresample.
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Michael Smith authored
Original commit message from CVS: * gst/playback/gstdecodebin2.c: Fix random fat-fingering making this not compile.
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- 20 Nov, 2008 4 commits
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Michael Smith authored
gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching beh... Original commit message from CVS: * gst/playback/gstdecodebin2.c: If the top-level type of the stream is plain text, don't try to decode it, matching behaviour of decodebin. * gst/playback/gstplaysink.c: If we fail to generate a text chain (e.g. due to missing optional plugins), don't crash.
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Michael Smith authored
Original commit message from CVS: * gst-libs/gst/rtsp/gstrtspdefs.c: Fix win32 build. Oops.
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Michael Smith authored
Original commit message from CVS: * gst-libs/gst/rtsp/gstrtspdefs.c: Use WSAGetLastError() rather than errno/h_errno on win32.
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Michael Smith authored
Original commit message from CVS: * gst-libs/gst/riff/riff-media.c: Support WMA Lossless properly.
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- 19 Nov, 2008 1 commit
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David Schleef authored
gst/videotestsrc/: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affect... Original commit message from CVS: * gst/videotestsrc/gstvideotestsrc.c: * gst/videotestsrc/gstvideotestsrc.h: * gst/videotestsrc/videotestsrc.c: * gst/videotestsrc/videotestsrc.h: Add "colorspec" property, specifying whether to generate BT.601 or BT.709 video. This only affects YCbCr values, not RGB, since if you're generating a 709 test pattern, presumably you want 709 RGB primaries, not 601. Also add "smpte75" pattern, which uses 75% colors instead of 100%, since this is often more useful for testing (and also follows the SMPTE EG-1 guideline).
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- 18 Nov, 2008 1 commit
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Alessandro Decina authored
Original commit message from CVS: * gst/playback/gstdecodebin.c: Add a "sink-caps" property to decodebin like it's done for decodebin2. Fixes #560380.
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- 14 Nov, 2008 3 commits
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Jan Schmidt authored
gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arri... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Guard against a NULL dereference I somehow encountered - with a FLUSH_STOP arriving either before basetransform _start(), or after _stop(). * gst/typefind/gsttypefindfunctions.c: Make sure we never jump backwards when typefinding corrupt mov files.
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Jan Schmidt authored
Original commit message from CVS: * gst-libs/gst/interfaces/propertyprobe.c: Fix random type causing a docs warning.
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Stefan Kost authored
Original commit message from CVS: * sys/v4l/gstv4l.c: Give it a minimal rank for autovideosrc.
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- 13 Nov, 2008 4 commits
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Mark Nauwelaerts authored
Original commit message from CVS: * gst/typefind/gsttypefindfunctions.c: (jp2_type_find), (plugin_init): Improve typefinding of ISO JPEG2000 mime types.
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Wim Taymans authored
Original commit message from CVS: * sys/xvimage/xvimagesink.c: (gst_xvimage_buffer_finalize), (gst_xvimagesink_xvimage_put), (gst_xvimagesink_setcaps), (gst_xvimagesink_show_frame), (gst_xvimagesink_buffer_alloc): * sys/xvimage/xvimagesink.h: Avoid typechecking when we do trivial casts. Move error handling out of the main program flow. Sneak in the display-region caps property, not completely correct yet. Cache the width/height in buffer_alloc instead of parsing it from the caps all the time.
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Wim Taymans authored
gst/playback/gstplaybin2.c: don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an... Original commit message from CVS: * gst/playback/gstplaybin2.c: (deactivate_group): don't try to unlink the selector sinkpad when we don't have it yet. This can happen if an error occured before the group was complete.
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Wim Taymans authored
gst-libs/gst/rtp/gstrtpbuffer.c: Avoid expensive type checks we already did as part of the _validate() function that ... Original commit message from CVS: * gst-libs/gst/rtp/gstrtpbuffer.c: (gst_rtp_buffer_validate_data), (gst_rtp_buffer_set_packet_len), (gst_rtp_buffer_get_packet_len), (gst_rtp_buffer_get_header_len), (gst_rtp_buffer_get_version), (gst_rtp_buffer_set_version), (gst_rtp_buffer_get_padding), (gst_rtp_buffer_set_padding), (gst_rtp_buffer_pad_to), (gst_rtp_buffer_get_extension), (gst_rtp_buffer_set_extension), (gst_rtp_buffer_get_extension_data), (gst_rtp_buffer_set_extension_data), (gst_rtp_buffer_get_ssrc), (gst_rtp_buffer_set_ssrc), (gst_rtp_buffer_get_csrc_count), (gst_rtp_buffer_get_csrc), (gst_rtp_buffer_set_csrc), (gst_rtp_buffer_get_marker), (gst_rtp_buffer_set_marker), (gst_rtp_buffer_get_payload_type), (gst_rtp_buffer_set_payload_type), (gst_rtp_buffer_get_seq), (gst_rtp_buffer_set_seq), (gst_rtp_buffer_get_timestamp), (gst_rtp_buffer_set_timestamp), (gst_rtp_buffer_get_payload_subbuffer), (gst_rtp_buffer_get_payload_len), (gst_rtp_buffer_get_payload): Avoid expensive type checks we already did as part of the _validate() function that should be called first.
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- 11 Nov, 2008 2 commits
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Wim Taymans authored
Original commit message from CVS: * gst-libs/gst/rtp/gstbasertpdepayload.c: (create_segment_event), (gst_base_rtp_depayload_push_full), (gst_base_rtp_depayload_set_gst_timestamp): Fix some cases where a newsegment event was not sent.
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Wim Taymans authored
gst/playback/gstplaybin2.c: Catch state change errors and stop from the uridecodebin elements instead of trying to co... Original commit message from CVS: * gst/playback/gstplaybin2.c: (activate_group): Catch state change errors and stop from the uridecodebin elements instead of trying to continue in vain.
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- 10 Nov, 2008 4 commits
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Edward Hervey authored
Original commit message from CVS: * gst-libs/gst/app/gstappsink.c: * gst-libs/gst/app/gstappsrc.c: * gst/h264parse/gsth264parse.c: Wim, you're a bad boy. You don't want people to contact you or what?
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Wim Taymans authored
gst-libs/gst/audio/gstbaseaudiosink.c: Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the ... Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_sync_latency), (gst_base_audio_sink_render), (gst_base_audio_sink_callback): Use gst_base_sink_do_preroll() to wait for PLAYING and before waiting for the latency to expire, fixes #559567.
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Thomas Vander Stichele authored
Original commit message from CVS: * gst/adder/gstadder.c: Change author string after seeing output of gst-inspector.
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Wim Taymans authored
gst/playback/gstplaysink.c: Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559... Original commit message from CVS: * gst/playback/gstplaysink.c: (gst_play_sink_reconfigure): Don't try to do crazy things when we only have a text pad without a video pad. Fixes #559478.
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- 07 Nov, 2008 1 commit
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Wim Taymans authored
Original commit message from CVS: * gst-libs/gst/app/gstappsrc.c: (gst_app_src_class_init), (gst_app_src_init), (gst_app_src_set_property), (gst_app_src_get_property), (gst_app_src_push_buffer): * gst-libs/gst/app/gstappsrc.h: Add is-live property. Add some more docs.
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- 06 Nov, 2008 1 commit
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Wim Taymans authored
gst-libs/gst/riff/riff-media.c: Fix case where we don't have a range for the rates or channels as is the case with tr... Original commit message from CVS: * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps): Fix case where we don't have a range for the rates or channels as is the case with truespeech.
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- 05 Nov, 2008 2 commits
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Wim Taymans authored
Original commit message from CVS: * gst/volume/gstvolume.c: (volume_update_real_volume), (gst_volume_set_volume), (gst_volume_get_volume), (gst_volume_set_mute), (gst_volume_init), (volume_setup), (volume_transform_ip), (volume_update_mute), (volume_update_volume), (volume_get_property): * gst/volume/gstvolume.h: Keep negotiated state in a separate variable. Protect the volume and mute properties with the object lock. Protect modifying the transform with the transform lock.
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Wim Taymans authored
Original commit message from CVS: * gst/ffmpegcolorspace/gstffmpegcodecmap.c: (gst_ffmpeg_pixfmt_to_caps): Only convert caps to string when debug is enabled.
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