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2007-11-23gst/audioresample/gstaudioresample.c: Implement latency query.Sebastian Dröge1-0/+78
Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_query), (audioresample_query_type), (gst_audioresample_set_property): Implement latency query.
2007-04-27ext/theora/theoradec.c: Calculate buffer duration correctly to generate a ↵Julien Moutte1-3/+1
perfect stream (#433888). Original commit message from CVS: 2007-04-27 Julien MOUTTE <julien@moutte.net> * ext/theora/theoradec.c: (_theora_granule_time), (theora_dec_push_forward), (theora_handle_data_packet), (theora_dec_decode_buffer): Calculate buffer duration correctly to generate a perfect stream (#433888). * gst/audioresample/gstaudioresample.c: (audioresample_check_discont): Glib provides ABS.
2007-04-21gst/audioresample/gstaudioresample.c: Make more functions static, just ↵Tim-Philipp Müller1-9/+9
because we can. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Make more functions static, just because we can.
2007-04-16ext/pango/gstclockoverlay.c: Fix unused variable warning if HAVE_LOCALTIME_R ↵Vincent Torri1-1/+1
is undefinied Original commit message from CVS: Patch by: Vincent Torri <vtorri at univ-evry dot fr> * ext/pango/gstclockoverlay.c: (gst_clock_overlay_render_time): Fix unused variable warning if HAVE_LOCALTIME_R is undefinied * gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps): * gst/audioresample/gstaudioresample.c: (audioresample_do_output): Use the correct format strings for integer formats.
2007-03-15gst/audioresample/gstaudioresample.c: Don't trigger discontinuities for very ↵Michael Smith1-16/+34
small imperfections; a filter flush will... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (audioresample_check_discont), (audioresample_transform): Don't trigger discontinuities for very small imperfections; a filter flush will sound bad, and many plugins have rounding errors leading to these.
2007-03-14gst/audioresample/gstaudioresample.c: Handle discontinuous streams.Julien Moutte1-9/+46
Original commit message from CVS: 2007-03-14 Julien MOUTTE <julien@moutte.net> * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (audioresample_transform_size), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough): Handle discontinuous streams. * gst/audioresample/gstaudioresample.h: * tests/check/elements/audioresample.c: (test_discont_stream_instance), (GST_START_TEST), (audioresample_suite): Add a test for discontinuous streams. * win32/common/config.h: Updated.
2007-03-14add debugging and reformat docsThomas Vander Stichele1-8/+21
Original commit message from CVS: add debugging and reformat docs
2006-10-28gst/audioresample/gstaudioresample.c: Another typo fix (#366212).Tim-Philipp Müller1-1/+1
Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init): Another typo fix (#366212).
2006-08-20gst/audioconvert/gstaudioconvert.c: Lower debug, use g_assert in _get_unit_sizeStefan Kost1-1/+1
Original commit message from CVS: * gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_get_unit_size), (set_structure_widths): Lower debug, use g_assert in _get_unit_size * gst/audioresample/gstaudioresample.c: (audioresample_get_unit_size): * gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_get_unit_size): * gst/videoscale/gstvideoscale.c: (gst_video_scale_get_unit_size): use g_assert in _get_unit_size
2006-07-28gst/audioresample/gstaudioresample.c: Don't leak references to the incoming ↵Jan Schmidt1-3/+3
caps. Clean them up when stopping. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (audioresample_stop), (audioresample_set_caps): Don't leak references to the incoming caps. Clean them up when stopping. * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init), (gst_video_scale_finalize): Don't leak our temporary pixel buffer. * tests/check/Makefile.am: * tests/check/pipelines/simple-launch-lines.c: (run_pipeline), (GST_START_TEST), (simple_launch_lines_suite): Fix leaks and re-enable the test for valgrind checking.
2006-06-22gst/: Avoid unnecessary class cast check in class_init functions (#337747).Cody Russell1-1/+1
Original commit message from CVS: Patch by: Cody Russell <bratsche at gnome org> * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init): * gst/playback/gststreamselector.c: (gst_stream_selector_class_init): * gst/subparse/gstsubparse.c: (gst_sub_parse_class_init): * gst/tcp/gstmultifdsink.c: (gst_multi_fd_sink_class_init): * gst/tcp/gsttcpclientsink.c: (gst_tcp_client_sink_class_init): * gst/tcp/gsttcpclientsrc.c: (gst_tcp_client_src_class_init): * gst/tcp/gsttcpserversink.c: (gst_tcp_server_sink_class_init): * gst/tcp/gsttcpserversrc.c: (gst_tcp_server_src_class_init): * gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init): * gst/videotestsrc/gstvideotestsrc.c: (gst_video_test_src_class_init): * gst/volume/gstvolume.c: (gst_volume_class_init): Avoid unnecessary class cast check in class_init functions (#337747).
2006-06-16gst/audioresample/gstaudioresample.c: Implement GstBaseTransform::start and ↵Tim-Philipp Müller1-25/+28
::stop so that audioresample can clear it... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_class_init), (gst_audioresample_init), (audioresample_start), (audioresample_stop), (gst_audioresample_set_property), (gst_audioresample_get_property): Implement GstBaseTransform::start and ::stop so that audioresample can clear its internal state properly and be reused insted of causing non-negotiated errors with playbin under some circumstances (#342789). * tests/check/elements/audioresample.c: (setup_audioresample), (cleanup_audioresample): Need to set element state here so that ::start and ::stop are called.
2006-04-28make GstElementDetails constStefan Kost1-1/+1
Original commit message from CVS: * ext/alsa/gstalsamixerelement.c: * ext/alsa/gstalsasrc.c: * ext/cdparanoia/gstcdparanoiasrc.c: * ext/gnomevfs/gstgnomevfssink.c: * ext/gnomevfs/gstgnomevfssrc.c: * ext/ogg/gstoggdemux.c: * ext/ogg/gstoggmux.c: * ext/ogg/gstoggparse.c: * ext/ogg/gstogmparse.c: * ext/pango/gstclockoverlay.c: * ext/pango/gsttextoverlay.c: * ext/pango/gsttextrender.c: * ext/pango/gsttimeoverlay.c: * ext/theora/theoradec.c: * ext/theora/theoraenc.c: * ext/vorbis/vorbisdec.c: * ext/vorbis/vorbisenc.c: * gst-libs/gst/audio/gstaudiofilter.c: * gst-libs/gst/audio/gstaudiofiltertemplate.c: * gst/audioconvert/gstaudioconvert.c: * gst/audiorate/gstaudiorate.c: * gst/audioresample/gstaudioresample.c: * gst/audiotestsrc/gstaudiotestsrc.c: * gst/ffmpegcolorspace/gstffmpegcolorspace.c: * gst/playback/gstdecodebin.c: * gst/playback/gstplaybin.c: * gst/playback/gststreamselector.c: * gst/subparse/gstsubparse.c: * gst/tcp/gstmultifdsink.c: * gst/tcp/gsttcpclientsink.c: * gst/tcp/gsttcpclientsrc.c: * gst/tcp/gsttcpserversink.c: * gst/tcp/gsttcpserversrc.c: * gst/typefind/gsttypefindfunctions.c: (plugin_init): * gst/videorate/gstvideorate.c: * gst/videoscale/gstvideoscale.c: * gst/videotestsrc/gstvideotestsrc.c: * gst/volume/gstvolume.c: * sys/v4l/gstv4ljpegsrc.c: * sys/v4l/gstv4lmjpegsink.c: * sys/v4l/gstv4lmjpegsrc.c: * sys/v4l/gstv4lsrc.c: * sys/ximage/ximagesink.c: * sys/xvimage/xvimagesink.c: * tests/check/libs/cddabasesrc.c: make GstElementDetails const
2006-04-28gst/audioresample/gstaudioresample.c: Add support for other formats ↵Wim Taymans1-17/+75
audioresample can handle such as 32 bits in and f... Original commit message from CVS: * gst/audioresample/gstaudioresample.c: (gst_audioresample_init), (resample_set_state_from_caps): Add support for other formats audioresample can handle such as 32 bits in and float and 64 bits float. Fixes #301759
2006-03-02docs/plugins/: Add audioresample to docs.Wim Taymans1-64/+90
Original commit message from CVS: * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-base-plugins-docs.sgml: * docs/plugins/gst-plugins-base-plugins-sections.txt: Add audioresample to docs. * gst/audioconvert/gstaudioconvert.c: Add revision date. * gst/audioresample/gstaudioresample.c: (gst_audioresample_base_init), (gst_audioresample_class_init), (gst_audioresample_init), (gst_audioresample_dispose), (audioresample_get_unit_size), (audioresample_transform_caps), (resample_set_state_from_caps), (audioresample_transform_size), (audioresample_set_caps), (audioresample_event), (audioresample_do_output), (audioresample_transform), (audioresample_pushthrough), (gst_audioresample_set_property), (gst_audioresample_get_property), (plugin_init): * gst/audioresample/gstaudioresample.h: Added docs. Small code cleanups.
2005-12-15gst/audioresample/gstaudioresample.c: Don't leak all input buffers to ↵Michael Smith1-2/+3
audioresample. Original commit message from CVS: * gst/audioresample/gstaudioresample.c: Don't leak all input buffers to audioresample.
2005-12-06expand tabsThomas Vander Stichele1-1/+1
Original commit message from CVS: expand tabs
2005-12-02gst/audioresample/: Fix audioresample, seek torture, new segments, reverse ↵Wim Taymans1-49/+166
negotiation etc.. work fine. Original commit message from CVS: * gst/audioresample/buffer.c: (audioresample_buffer_queue_flush): * gst/audioresample/buffer.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: (resample_input_flush), (resample_input_pushthrough), (resample_input_eos), (resample_get_output_size_for_input), (resample_get_input_size_for_output), (resample_get_output_size), (resample_get_output_data): * gst/audioresample/resample.h: * gst/audioresample/resample_ref.c: (resample_scale_ref): Fix audioresample, seek torture, new segments, reverse negotiation etc.. work fine.
2005-11-21gst/: Segment update fix.Wim Taymans1-2/+2
Original commit message from CVS: * gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_class_init), (gst_base_audio_sink_init), (gst_base_audio_sink_provide_clock), (gst_base_audio_sink_set_clock), (gst_base_audio_sink_render), (gst_base_audio_sink_change_state): * gst/audioresample/gstaudioresample.c: Segment update fix.
2005-10-16restructure configure.ac, use correct libtool LDFLAGS, fix up definesThomas Vander Stichele1-1/+2
Original commit message from CVS: restructure configure.ac, use correct libtool LDFLAGS, fix up defines
2005-09-23gst/audioresample/: Convert to using gst debuggingDavid Schleef1-1/+1
Original commit message from CVS: * gst/audioresample/Makefile.am: * gst/audioresample/debug.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/resample.c: Convert to using gst debugging
2005-09-09check/: Add extra tests for basetransform based components.Jan Schmidt1-0/+2
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.
2005-08-28Updates for two-arg init from GST_BOILERPLATE_FULL.Andy Wingo1-4/+2
Original commit message from CVS: 2005-08-28 Andy Wingo <wingo@pobox.com> * Updates for two-arg init from GST_BOILERPLATE_FULL.
2005-08-26use base class' newsegment to properly timestampThomas Vander Stichele1-2/+5
Original commit message from CVS: use base class' newsegment to properly timestamp
2005-08-25check/: add a test for audioconvertThomas Vander Stichele1-3/+9
Original commit message from CVS: * check/Makefile.am: * check/elements/audioconvert.c: (setup_audioconvert), (cleanup_audioconvert), (get_int_caps), (verify_convert), (GST_START_TEST), (audioconvert_suite), (main): add a test for audioconvert * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: set DURATION so that TIMESTAMP(a) + DURATION(a) == TIMESTAMP(b); note that for buffers of 1/3 sec this means DURATION(c) is one nanosecond more than for a and b
2005-08-25add a check for audioresampleThomas Vander Stichele1-2/+10
Original commit message from CVS: add a check for audioresample
2005-08-25gst/audioresample/: add room for extra overlap samples when asked to ↵Thomas Vander Stichele1-25/+45
transform size protect against possible mem corr... Original commit message from CVS: * gst/audioresample/debug.c: * gst/audioresample/gstaudioresample.c: add room for extra overlap samples when asked to transform size protect against possible mem corruption and check for discrepancies between written size and outbuffer's size so we can warn for potential problems * gst/audioresample/resample.c: (resample_init), (resample_get_output_size_for_input), (resample_get_output_size), (resample_set_n_channels), (resample_set_format): set debug level based on RESAMPLE_DEBUG env var make sure that get_output_size* returns a whole number of sample_size set sample_size each time either channel or format is set * gst/audioresample/resample_chunk.c: (resample_scale_chunk): * gst/audioresample/resample_functable.c: (resample_scale_functable): * gst/audioresample/resample_ref.c: (resample_scale_ref): remove r->sample_size, it's done in resample.c now add some debugging to the ref implementation make sure we only give back bytes that are wholes of the sample size
2005-08-24port audioresample to basetransformThomas Vander Stichele1-202/+224
Original commit message from CVS: port audioresample to basetransform
2005-08-23gst/audioresample/Makefile.am: Leet audioresampling codeDavid Schleef1-0/+434
Original commit message from CVS: * gst/audioresample/Makefile.am: Leet audioresampling code * gst/audioresample/buffer.c: * gst/audioresample/buffer.h: * gst/audioresample/debug.c: * gst/audioresample/debug.h: * gst/audioresample/functable.c: * gst/audioresample/functable.h: * gst/audioresample/gstaudioresample.c: * gst/audioresample/gstaudioresample.h: * gst/audioresample/resample.c: * gst/audioresample/resample.h: * gst/audioresample/resample_chunk.c: * gst/audioresample/resample_functable.c: * gst/audioresample/resample_ref.c: