| Age | Commit message (Collapse) | Author | Files | Lines |
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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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Original commit message from CVS:
* gst/wavparse/gstwavparse.c:
add ATRAC3 to STATIC CAPS to fix a warning
* gst/matroska/ebml-read.c:
* gst-libs/gst/riff/riff-read.c:
fix typos
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Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
generate caps for ATRAC3 audio streams
* gst/realmedia/rmdemux.c:
generate caps for ATRAC3 audio streams
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