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Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_flush_unlocked):
Don't use new API.
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Original commit message from CVS:
* gst-libs/gst/app/gstappsink.c: (gst_app_sink_base_init),
(gst_app_sink_class_init), (gst_app_sink_dispose),
(gst_app_sink_flush_unlocked), (gst_app_sink_start),
(gst_app_sink_stop), (gst_app_sink_event), (gst_app_sink_preroll),
(gst_app_sink_render), (gst_app_sink_get_caps),
(gst_app_sink_set_caps), (gst_app_sink_end_of_stream),
(gst_app_sink_pull_preroll), (gst_app_sink_pull_buffer):
* gst-libs/gst/app/gstappsink.h:
Make love to appsink.
Make it support pulling of the preroll buffer.
Add docs and debug statements.
Fix some races wrt to EOS handling and stopping.
Implement getcaps.
Implement FLUSHING.
API: gst_app_sink_pull_preroll()
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doc as this sink use the mixer interface now.
Original commit message from CVS:
* docs/plugins/gst-plugins-bad-plugins.args:
Remove directsoundsink property doc as this sink use the mixer
interface now.
* docs/plugins/gst-plugins-bad-plugins.interfaces:
Add interfaces implemented by Windows sinks.
* sys/directsound/gstdirectsoundsink.c:
* sys/directsound/gstdirectsoundsink.h:
Remove directsoundsink property and implement the mixer interface.
* win32/vs6/gst_plugins_bad.dsw:
* win32/vs6/libgstdirectsound.dsp:
Update project files.
* gst-libs/gst/dshow/gstdshow.cpp:
* gst-libs/gst/dshow/gstdshow.h:
* gst-libs/gst/dshow/gstdshowfakesink.cpp:
* gst-libs/gst/dshow/gstdshowfakesink.h:
* gst-libs/gst/dshow/gstdshowfakesrc.cpp:
* gst-libs/gst/dshow/gstdshowfakesrc.h:
* gst-libs/gst/dshow/gstdshowinterface.cpp:
* gst-libs/gst/dshow/gstdshowinterface.h:
* win32/common/libgstdshow.def:
* win32/vs6/libgstdshow.dsp:
Add a new gst library which allow to create internal Direct Show
graph (pipelines) to wrap Windows sources, decoders or encoders.
It includes a DirectShow fake source and sink and utility functions.
* sys/dshowsrcwrapper/gstdshowaudiosrc.c:
* sys/dshowsrcwrapper/gstdshowaudiosrc.h:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.c:
* sys/dshowsrcwrapper/gstdshowsrcwrapper.h:
* sys/dshowsrcwrapper/gstdshowvideosrc.c:
* sys/dshowsrcwrapper/gstdshowvideosrc.h:
* win32/vs6/libdshowsrcwrapper.dsp:
Add a new plugin to wrap DirectShow sources on Windows.
It gets data from any webcam, dv cam, micro. We could add
tv tunner card later.
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but maybe David can confirm that was what h...
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Use GST_ALL_LDFLAGS, which actually exists, but maybe David
can confirm that was what he wanted.
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Original commit message from CVS:
* configure.ac:
* examples/Makefile.am:
* examples/app/Makefile.am:
* examples/app/appsrc_ex.c:
Add appsrc/appsink example.
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsink.c:
* gst-libs/gst/app/gstappsink.h:
* gst/app/gstapp.c:
Add appsink.
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Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
Install the headers.
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proper handling of data chunks.
Original commit message from CVS:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstappbuffer.c:
* gst-libs/gst/app/gstappbuffer.h:
* gst-libs/gst/app/gstappsrc.c:
Add GstAppBuffer that includes a callback and closure for
proper handling of data chunks.
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Original commit message from CVS:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
Hacking to address issues in 413418.
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Original commit message from CVS:
* Makefile.am:
* configure.ac:
* ext/Makefile.am:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/app/Makefile.am:
* gst-libs/gst/app/gstapp.c:
* gst-libs/gst/app/gstappsrc.c:
* gst-libs/gst/app/gstappsrc.h:
* gst/app/Makefile.am:
* gst/app/gstapp.c:
* gst/app/gstappsrc.c:
* gst/app/gstappsrc.h:
Move the app library to gst-libs/gst/app (duh!)
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Original commit message from CVS:
* ext/xine/gstxine.h:
* gst-libs/gst/play/play.h:
* sys/v4l2/gstv4l2element.h:
* sys/ximagesrc/ximageutil.h:
Fix broken GObject macros
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Original commit message from CVS:
expand tabs
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Original commit message from CVS:
fix disting and spec file
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Original commit message from CVS:
remove stuff that's in -base
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Original commit message from CVS:
remove gst-libs from gst-plugins module as it is in gst-plugins-base now
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values for alaw and mulaw audio instead of ju...
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Do actually fix invalid RIFF fmt header values for alaw
and mulaw audio instead of just saying so.
* gst/wavparse/gstwavparse.c: (gst_wavparse_fmt):
Give gst_riff_create_audio_caps_with_data() a chance to
fix up broken format header fields before extracting any
parameters from the header. (fixes #167633)
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work when included from C++ code
Original commit message from CVS:
Add G_BEGIN_DECLS and G_END_DECLS around headers where missing, so that they work when included from C++ code
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(Fixes #165997)
Original commit message from CVS:
* configure.ac: Put DEFAULT_AUDIOSINK in config.h and use
whereever possible. (Fixes #165997)
* examples/capsfilter/capsfilter1.c: (main):
* examples/dynparams/filter.c: (create_ui):
* examples/seeking/cdparanoia.c: (get_track_info), (main):
* examples/seeking/chained.c: (main):
* examples/seeking/seek.c: (make_mod_pipeline), (make_dv_pipeline),
(make_wav_pipeline), (make_flac_pipeline), (make_sid_pipeline),
(make_vorbis_pipeline), (make_mp3_pipeline), (make_avi_pipeline),
(make_mpeg_pipeline), (make_mpegnt_pipeline):
* examples/seeking/spider_seek.c: (make_spider_pipeline):
* examples/switch/switcher.c: (main):
* ext/dv/demo-play.c: (main):
* ext/faad/gstfaad.c: (gst_faad_change_state):
* ext/mad/gstmad.c: (gst_mad_chain):
* ext/smoothwave/demo-osssrc.c: (main):
* gst-libs/gst/gconf/gconf.c: (gst_gconf_set_string),
(gst_gconf_render_bin_from_description),
(gst_gconf_get_default_audio_sink),
(gst_gconf_get_default_video_sink),
(gst_gconf_get_default_audio_src),
(gst_gconf_get_default_video_src),
(gst_gconf_get_default_visualization_element):
* gst/level/demo.c: (main):
* gst/level/plot.c: (main):
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element):
* gst/playback/test.c: (gen_video_element), (gen_audio_element):
* gst/playondemand/demo-mp3.c: (setup_pipeline):
* gst/sine/demo-dparams.c: (main):
* gst/spectrum/demo-osssrc.c: (main):
* gst/speed/demo-mp3.c: (main):
* gst/volume/demo.c: (main):
* testsuite/embed/embed.c: (main):
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add extradata to huffyuv (fixes #165013).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Fix extradata extraction if it is in the chunk size.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Audio can be <8000Hz.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_element_data),
(gst_riff_read_element_data):
* gst-libs/gst/riff/riff-read.h:
Add _peek version (req'ed in CDXA).
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_init),
(gst_cdxaparse_loop):
Fix parsing in playbin.
* gst/playback/gstdecodebin.c: (close_pad_link):
Ignore current_ pads, they cause major annoyance.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add intel-h263.
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Original commit message from CVS:
ignore more
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Original commit message from CVS:
ignore generated files
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Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't bail on unknown events.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_chain):
Don't crash on events before negotiation.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Send tags on pads, too.
* gst/playback/gststreamselector.c:
(gst_stream_selector_request_new_pad):
Forward events on first pad if no input was selected yet.
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Original commit message from CVS:
* configure.ac:
* examples/capsfilter/capsfilter1.c: (main):
* examples/seeking/spider_seek.c: (make_spider_pipeline):
* ext/dvdread/Makefile.am:
* ext/dvdread/demo-play:
* ext/dvdread/demo-play.c:
* gconf/gstreamer.schemas.in:
* gst-libs/gst/gconf/gconf.c:
* sys/v4l/TODO:
* testsuite/Makefile.am:
* testsuite/embed/Makefile.am:
* testsuite/embed/embed.c: (cb_expose), (main):
Remove all references to xvideosink, fix examples (#140845).
* gst/playback/gstplaybasebin.c: (group_destroy):
Apparently, disposal does not unlink - so do explicitely.
* ext/alsa/gstalsasink.c: (gst_alsa_sink_check_event):
Add debug.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-ids.h:
* gst/wavenc/riff.h:
Add AMR (VBR and CBR) ids to riff.h audio codec list
* gst/asfdemux/gstasfdemux.c:
(gst_asf_demux_process_ext_content_desc),
(gst_asf_demux_process_object):
Retrieve more tags from ASF files (Genre, AlbumTitle, Artist)
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Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_sinc_ft_s16):
Fix invalid memory access (#159211).
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Original commit message from CVS:
* examples/gstplay/player.c: (main):
Don't iterate.
* examples/seeking/seek.c: (fixate), (make_playerbin_pipeline):
Add visualizations.
* ext/a52dec/gsta52dec.c: (gst_a52dec_push),
(gst_a52dec_handle_frame):
Set duration.
* ext/dvdnav/gst-dvd:
Add audioconvert. Fixes #161325.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get):
Explicitely case to gint64. Possible valgrind error.
* gst-libs/gst/play/play.c: (caps_set), (setup_size),
(gst_play_tick_callback), (gst_play_change_state),
(gst_play_dispose), (gst_play_init), (gst_play_class_init),
(gst_play_set_location), (gst_play_get_location),
(gst_play_seek_to_time), (gst_play_set_data_src),
(gst_play_set_video_sink), (gst_play_set_audio_sink),
(gst_play_set_visualization), (gst_play_connect_visualization),
(gst_play_get_framerate), (gst_play_get_all_by_interface),
(gst_play_new):
Use playbin. Fixes #139749 and #147744.
* gst/apetag/apedemux.c: (gst_ape_demux_parse_tags):
Add genre tag.
* gst/audioscale/gstaudioscale.c: (gst_audioscale_method_get_type),
(audioscale_get_type), (gst_audioscale_base_init),
(gst_audioscale_class_init), (gst_audioscale_expand_caps),
(gst_audioscale_getcaps), (gst_audioscale_fixate),
(gst_audioscale_link), (gst_audioscale_get_buffer),
(gst_audioscale_decrease_rate), (gst_audioscale_increase_rate),
(gst_audioscale_init), (gst_audioscale_dispose),
(gst_audioscale_chain), (gst_audioscale_set_property),
(gst_audioscale_get_property), (plugin_init):
Indent properly.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_process_private):
Fix LPCM.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_udta),
(qtdemux_tag_add_str), (qtdemux_tag_add_num),
(qtdemux_tag_add_gnre), (qtdemux_video_caps):
Add more metadata (fixes #162656).
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
Add BLZ0 (Blizzard's version of DivX) fourcc.
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Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_auds_with_data):
Read extradata correctly (fixes #155879).
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add h264.
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Original commit message from CVS:
* gst-libs/gst/audio/Makefile.am:
Try to fix buildbot.
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Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/resample/resample.c: (gst_resample_close):
* gst-libs/gst/resample/resample.h:
* gst/audioscale/gstaudioscale.c:
Fix memleak (#159215).
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Original commit message from CVS:
* configure.ac: add audioresample and cairo plugins. Remove
HAVE_MMX stuff, because it's not used.
* ext/Makefile.am: same
* ext/audioresample/Makefile.am: You are not ready for an
audio resampling element based on audioresample.
* ext/audioresample/gstaudioresample.c:
* ext/audioresample/gstaudioresample.h:
* ext/cairo/Makefile.am: You are not ready for overlay elements
based on cairo. Don't look too closely, these elements kinda
suck right now.
* ext/cairo/gstcairo.c: new
* ext/cairo/gsttextoverlay.c: new
* ext/cairo/gsttextoverlay.h: new
* ext/cairo/gsttimeoverlay.c: new
* ext/cairo/gsttimeoverlay.h: new
* gst-libs/gst/media-info/media-info-priv.h: fix compile
problem with compilers that don't support variadic macros.
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Original commit message from CVS:
forgot to add H264 to avidemux template caps
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is quicktime specific
Original commit message from CVS:
add VSSH (VideoSoft h264) and remove s323 (h323) from riff-lib
because s323 is quicktime specific
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gst-libs/gst/riff/riff-media.c add new 4CC codes f...
Original commit message from CVS:
* gst/asfdemux/README
* gst/wavenc/riff.h
* gst-libs/gst/riff/riff-ids.h
* gst-libs/gst/riff/riff-media.c
add new 4CC codes for h263 related codecs
fixes partially bug #155163
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increasing timestamps.
Original commit message from CVS:
* ext/faad/gstfaad.c: (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_chain):
Set DURATION even if source buffer didn't. Also use increasing
timestamps.
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Block_align can have larger values than 8192.
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Original commit message from CVS:
* ext/esd/esdsink.c: (gst_esdsink_chain):
Make error actually say something useful (fixes #156798).
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_video_template_caps):
Add Intel Video 5.0 fourcc (IV50).
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Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_use_event):
Don't forward DISCONT events (fixes #159684).
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Original commit message from CVS:
2004-11-27 Martin Soto <martinsoto@users.sourceforge.net>
* gst-libs/gst/audio/audioclock.c (gst_audio_clock_set_active)
(gst_audio_clock_get_internal_time):
Fix active <-> inactive transitions: ensure time value always
grows and avoid abrupt value changes.
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Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels), (gst_a52dec_push),
(gst_a52dec_reneg), (gst_a52dec_loop), (plugin_init):
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
* ext/alsa/gstalsaplugin.c: (plugin_init):
* ext/dts/gstdtsdec.c: (gst_dtsdec_channels),
(gst_dtsdec_renegotiate), (gst_dtsdec_loop), (plugin_init):
* ext/faad/gstfaad.c: (gst_faad_init), (gst_faad_chanpos_from_gst),
(gst_faad_chanpos_to_gst), (gst_faad_sinkconnect),
(gst_faad_srcgetcaps), (gst_faad_srcconnect), (gst_faad_chain),
(gst_faad_change_state), (plugin_init):
* ext/faad/gstfaad.h:
* ext/vorbis/vorbis.c: (plugin_init):
* ext/vorbis/vorbisdec.c: (vorbis_dec_chain):
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/audio/audio.c: (plugin_init):
* gst-libs/gst/audio/multichannel.c:
(gst_audio_check_channel_positions),
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(add_list_to_struct), (gst_audio_set_caps_channel_positions_list),
(gst_audio_fixate_channel_positions):
* gst-libs/gst/audio/multichannel.h:
* gst-libs/gst/audio/testchannels.c: (main):
* gst/audioconvert/gstaudioconvert.c:
(gst_audio_convert_class_init), (gst_audio_convert_init),
(gst_audio_convert_dispose), (gst_audio_convert_getcaps),
(gst_audio_convert_parse_caps), (gst_audio_convert_link),
(gst_audio_convert_fixate), (gst_audio_convert_channels):
* gst/audioconvert/plugin.c: (plugin_init):
Surround sound support.
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Original commit message from CVS:
Fix another typo in doc string :)
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Original commit message from CVS:
Fix typo in doc string
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Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps_internal):
buffer-frames property was missing
* ext/arts/gst_arts.c:
rate missing from sinkcaps
* ext/audiofile/gstafparse.c:
* ext/audiofile/gstafsink.c:
* ext/audiofile/gstafsrc.c:
* ext/swfdec/gstswfdec.c:
int audio doesn't know buffer-frames
* ext/cdparanoia/gstcdparanoia.c:
int audio doesn't know chunksize either
* ext/nas/nassink.c:
it's endianness, not endianess
* gst-libs/gst/audio/audio.h:
make float standard pad template caps really describe float
* gst/law/mulaw.c: (linear_factory):
signed only, please
* gst/mpegstream/gstdvddemux.c:
widths of 20 are not valid
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assumptions that dispose is only called once, o...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_class_init), (gst_alsa_dispose),
(gst_alsa_finalize):
* ext/cdaudio/gstcdaudio.c: (gst_cdaudio_class_init),
(gst_cdaudio_finalize):
* ext/cdparanoia/gstcdparanoia.c: (cdparanoia_class_init),
(cdparanoia_finalize):
* ext/divx/gstdivxdec.c: (gst_divxdec_dispose):
* ext/divx/gstdivxenc.c: (gst_divxenc_dispose):
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_finalize):
* ext/flac/gstflacdec.c: (gst_flacdec_class_init),
(gst_flacdec_finalize):
* ext/flac/gstflacenc.c: (gst_flacenc_class_init),
(gst_flacenc_finalize):
* ext/gnomevfs/gstgnomevfssink.c: (gst_gnomevfssink_class_init),
(gst_gnomevfssink_finalize):
* ext/gnomevfs/gstgnomevfssrc.c: (gst_gnomevfssrc_class_init),
(gst_gnomevfssrc_finalize):
* ext/libfame/gstlibfame.c: (gst_fameenc_class_init),
(gst_fameenc_finalize):
* ext/nas/nassink.c: (gst_nassink_class_init),
(gst_nassink_finalize):
* ext/sdl/sdlvideosink.c: (gst_sdlvideosink_finalize),
(gst_sdlvideosink_class_init):
* ext/sndfile/gstsf.c: (gst_sf_dispose):
* gst-libs/gst/mixer/mixertrack.c: (gst_mixer_track_dispose):
* gst-libs/gst/tuner/tunerchannel.c: (gst_tuner_channel_dispose):
* gst-libs/gst/tuner/tunernorm.c: (gst_tuner_norm_dispose):
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
(gst_x_window_listener_dispose):
* gst/audioscale/gstaudioscale.c:
* gst/playondemand/gstplayondemand.c: (play_on_demand_class_init),
(play_on_demand_finalize):
* gst/videofilter/gstvideobalance.c: (gst_videobalance_dispose):
* gst/videoscale/gstvideoscale.c: (gst_videoscale_chain):
* sys/cdrom/gstcdplayer.c: (cdplayer_class_init),
(cdplayer_finalize):
* sys/glsink/glimagesink.c: (gst_glimagesink_finalize),
(gst_glimagesink_class_init):
* sys/oss/gstosselement.c: (gst_osselement_class_init),
(gst_osselement_finalize):
* sys/oss/gstosssink.c: (gst_osssink_dispose):
* sys/oss/gstosssrc.c: (gst_osssrc_dispose):
* sys/v4l/gstv4lelement.c: (gst_v4lelement_dispose):
Fixes a bunch of problems with finalize and dispose functions,
either assumptions that dispose is only called once, or not calling
the parent class dispose/finalize function
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Original commit message from CVS:
* ext/dv/gstdvdec.c: (gst_dvdec_handle_sink_event):
Set EOS on the element when processing an EOS event.
* ext/speex/gstspeexdec.h:
* ext/speex/gstspeexenc.h:
Only keep a const ptr to the mode
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data),
(gst_riff_create_audio_template_caps):
Allow WMAV3, with up to 6 channels.
* gst/asfdemux/gstasfmux.c: (gst_asfmux_request_new_pad):
Don't call gst_pad_set_event_function on a sink pad.
* gst/mpegstream/gstdvddemux.c:
(gst_dvd_demux_get_subpicture_stream),
(gst_dvd_demux_set_cur_audio), (gst_dvd_demux_set_cur_subpicture):
Copy the explicit caps that were set across to the cur_* pads,
instead of trying to use a possibly non-existent negotiated caps.
Reset the type of subpicture pads to UNKNOWN after calling init_stream,
so that the caps get set.
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Original commit message from CVS:
2004-10-28 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix build
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channels and query width for floats
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiofilter.c: (gst_audiofilter_link):
fix link function to always query channels and query width for
floats
* configure.ac:
add equalizer dir
* gst/equalizer/Makefile.am:
* gst/equalizer/gstiirequalizer.c: (gst_iir_equalizer_get_type),
(gst_iir_equalizer_base_init), (gst_iir_equalizer_class_init),
(gst_iir_equalizer_init), (gst_iir_equalizer_finalize),
(arg_to_scale), (setup_filter),
(gst_iir_equalizer_compute_frequencies),
(gst_iir_equalizer_set_property), (gst_iir_equalizer_get_property),
(gst_iir_equalizer_filter_inplace), (gst_iir_equalizer_setup),
(plugin_init):
add an equalizer
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_template_caps):
WMV3 missing in template caps.
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