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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.
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clock instead of ist internal one.
Original commit message from CVS:
2004-11-28 Martin Soto <martinsoto@users.sourceforge.net>
* ext/alsa/gstalsasink.c (gst_alsa_sink_loop):
* ext/alsa/gstalsa.h:
* ext/alsa/gstalsa.c (gst_alsa_set_clock):
Make alsasink actually honor gst_element_set_clock and use that
clock instead of ist internal one.
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when going from READY to NULL so that we ...
Original commit message from CVS:
2004-11-27 Christophe Fergeau <teuf@gnome.org>
* gst/playback/gstplaybasebin.c: (setup_source): fixed a caps leak
(gst_play_base_bin_change_state): nullify source and decoder when
going from READY to NULL so that we don't try to do weird stuff with
them when going from NULL to READY
* gst/playback/gstplaybin.c: (gst_play_bin_init): use gst_object_unref
instead of g_object_unref
(gen_video_element), (gen_audio_element): more refcounting fixes, now
it should be correct
(gst_play_bin_change_state): don't call remove_sinks if we are
currently disposing the object
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is silent...
Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_loop):
Don't forget bass if it's there. Else left channel is silent...
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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.
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Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_channels),
(gst_a52dec_change_state):
Advertise that we can do surround sound.
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Original commit message from CVS:
* ext/a52dec/gsta52dec.c: (gst_a52dec_reneg):
Add buffer-frames=0.
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_get_type),
(dvdreadsrc_init), (dvdreadsrc_get_event_mask),
(dvdreadsrc_get_query_types), (dvdreadsrc_get_formats),
(dvdreadsrc_srcpad_event), (dvdreadsrc_srcpad_query),
(_seek_title), (_seek_chapter), (get_next_cell_for), (_read),
(seek_sector), (dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state), (dvdreadsrc_uri_get_type),
(dvdreadsrc_uri_get_protocols), (dvdreadsrc_uri_get_uri),
(dvdreadsrc_uri_set_uri), (dvdreadsrc_uri_handler_init):
* ext/dvdread/dvdreadsrc.h:
Add seeking, querying for bytes, sectors, title, angle and
chapter. Handle multiple chapters. Relicense to LGPL because
Billy agreed on that (thanks Billy!).
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Original commit message from CVS:
2004-11-27 Christophe Fergeau <teuf@gnome.org>
* gst/audioconvert/gstaudioconvert.c: (gst_audio_convert_dispose):
call parent dispose method
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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:
added TTA parser and decoder
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Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_class_init),
(probe_triggered), (check_queue), (buffer_underrun),
(buffer_running), (buffer_overrun), (gen_source_element),
(setup_source):
* gst/playback/gstplaybasebin.h:
Implement buffering. Needs some more work.
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Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_chain):
Fix ilog mask range overflow.
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) channel count. Don't set channels if <...
Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_get_caps):
Don't omit the last (which incase of dmix is the only :) )
channel count. Don't set channels if <= 2.
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Original commit message from CVS:
2004-11-26 Christophe Fergeau <teuf@gnome.org>
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element): Removed 2 obsolete comments
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Original commit message from CVS:
* ext/vorbis/oggvorbisenc.c
* ext/vorbis/vorbisenc.c :
change description fields of those plugins to differentiate them
(pitivi show Encoders by description, they had the same one)
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Original commit message from CVS:
Reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/playback/gstplaybin.c: (gst_play_bin_dispose),
(gst_play_bin_set_property), (gen_video_element),
(gen_audio_element):
Refcounting fixes for provided audio-/videosinks.
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and audiosinks.
Original commit message from CVS:
* gst/playback/gstplaybin.c: (gen_video_element),
(gen_audio_element), (setup_sinks), (gst_play_bin_change_state):
Don't reference all sinks, but only the video- and audiosinks.
The vis. element should be disposed when we're done with it.
We don't have any reason to keep it around. This fixes warnings
when reusing playbin for playing multiple audio files with
vis. enabled. Also release audio device on pause - idea stolen
from Rhythmbox.
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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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Nitpicking, but saves a few milliseconds of e...
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_push):
Fix position for discont if we're close as well. Nitpicking, but
saves a few milliseconds of extra waiting or skipping.
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those too.
Original commit message from CVS:
* gst/playback/gstdecodebin.c: (gst_decode_bin_factory_filter):
We sometimes need parsers for playback, so add those too.
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Original commit message from CVS:
* configure.ac:
* gst/apetag/Makefile.am:
* gst/apetag/apedemux.c: (gst_ape_demux_get_type),
(gst_ape_demux_base_init), (gst_ape_demux_class_init),
(gst_ape_demux_init), (gst_ape_demux_get_src_formats),
(gst_ape_demux_get_src_query_types),
(gst_ape_demux_handle_src_query), (gst_ape_demux_get_event_mask),
(gst_ape_demux_handle_src_event), (gst_ape_demux_handle_event),
(gst_ape_demux_typefind_peek), (gst_ape_demux_typefind_get_length),
(gst_ape_demux_typefind_suggest), (gst_ape_demux_typefind),
(gst_ape_demux_parse_tags), (gst_ape_demux_stream_init),
(gst_ape_demux_stream_data), (gst_ape_demux_loop),
(gst_ape_demux_change_state):
* gst/apetag/apedemux.h:
* gst/apetag/apetag.c: (plugin_init):
* gst/typefind/gsttypefindfunctions.c: (apetag_type_find),
(plugin_init):
APE v1/2 tag reader plus typefind function.
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Original commit message from CVS:
* configure.ac:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find):
Remove hacks for older core. Require newer core version
accordingly.
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Original commit message from CVS:
* gst/cdxaparse/Makefile.am:
* gst/cdxaparse/gstcdxaparse.c: (gst_cdxaparse_get_type),
(gst_cdxaparse_class_init), (gst_cdxaparse_init),
(gst_cdxaparse_loop), (gst_cdxaparse_change_state), (plugin_init):
* gst/cdxaparse/gstcdxaparse.h:
* gst/cdxaparse/gstcdxastrip.c: (gst_cdxastrip_get_type),
(gst_cdxastrip_base_init), (gst_cdxastrip_class_init),
(gst_cdxastrip_init), (gst_cdxastrip_get_src_formats),
(gst_cdxastrip_get_src_query_types),
(gst_cdxastrip_handle_src_query), (gst_cdxastrip_get_event_mask),
(gst_cdxastrip_handle_src_event), (gst_cdxastrip_strip),
(gst_cdxastrip_sync), (gst_cdxastrip_handle_event),
(gst_cdxastrip_chain), (gst_cdxastrip_change_state):
* gst/cdxaparse/gstcdxastrip.h:
SVCD/VCD header stripping separated from CDXA image parsing.
* gst/typefind/gsttypefindfunctions.c: (mp3_type_find),
(plugin_init):
Add VCD/SVCD header typefinding for VCD/SVCD.
* sys/vcd/vcdsrc.c: (gst_vcdsrc_get_type), (gst_vcdsrc_base_init),
(gst_vcdsrc_class_init), (gst_vcdsrc_init),
(gst_vcdsrc_set_property), (gst_vcdsrc_get_property),
(gst_vcdsrc_get_event_mask), (gst_vcdsrc_get_query_types),
(gst_vcdsrc_get_formats), (gst_vcdsrc_srcpad_event),
(gst_vcdsrc_srcpad_query), (gst_vcdsrc_get),
(gst_vcdsrc_open_file), (gst_vcdsrc_close_file),
(gst_vcdsrc_change_state), (gst_vcdsrc_msf),
(gst_vcdsrc_recalculate), (gst_vcdsrc_uri_get_type),
(gst_vcdsrc_uri_get_protocols), (gst_vcdsrc_uri_get_uri),
(gst_vcdsrc_uri_set_uri), (gst_vcdsrc_uri_handler_init):
* sys/vcd/vcdsrc.h:
Fix up, add seeking, querying, URI interface. Works in totem now.
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Original commit message from CVS:
back to cvs
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Original commit message from CVS:
releasing
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#158365
Original commit message from CVS:
patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/playback/gstplaybasebin.c:
Fix unplayable files error handling. Fixes #158365
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Original commit message from CVS:
patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* gst/typefind/gsttypefindfunctions.c:
Fix broken mp3 typefinding. Fixes #158375
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Original commit message from CVS:
patch by: Ronald Bultje <rbultje@ronald.bitfreak.net>
* ext/ogg/gstoggdemux.c:
Fix sync on broken files. Fixes #158976
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Original commit message from CVS:
patch by: Edward Hervey <bilboed@bilboed.com>
* ext/libpng/gstpngenc.c:
Copy over buffer properties. Fixes #158832
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Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c:
Fixes invalid reads (#158462)
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Original commit message from CVS:
* sys/v4l/gstv4lsrc.c:
* sys/v4l/gstv4lsrc.h:
* sys/v4l/v4lsrc_calls.c:
Probe less and cache it. Fixes #159187.
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Original commit message from CVS:
* gst/videorate/gstvideorate.c:
Handle all video formats. Fixes #159186.
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different samplerates/ buffer sizes.
Original commit message from CVS:
* gst/synaesthesia/gstsynaesthesia.c:
(gst_synaesthesia_class_init), (gst_synaesthesia_init),
(gst_synaesthesia_dispose), (gst_synaesthesia_finalize),
(gst_synaesthesia_sink_link), (gst_synaesthesia_src_getcaps),
(gst_synaesthesia_src_link), (gst_synaesthesia_chain),
(gst_synaesthesia_change_state), (plugin_init):
Fix up synaesthesia to work under different samplerates/ buffer sizes.
Force 320x200 output, as that's the only thing the underlying
synaesthesia implementation supports. Still needs to be made
re-entrant.
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Original commit message from CVS:
* configure.ac:
Fix mpeg2enc configure check (similar to mplex check below).
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Original commit message from CVS:
reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
Fix for gcc-2.95 (fixes #158221).
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core is released).
Original commit message from CVS:
* gst/playback/gstplaybasebin.c: (gst_play_base_bin_add_element):
Re-add clock distribution hack (until new core is released).
Fixes #158125.
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Original commit message from CVS:
fix mplex configure check (bug #140994)
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Original commit message from CVS:
* ext/alsa/gstalsa.c: (gst_alsa_pcm_wait):
add debugging
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
do a wait when we enter the loop func with no data available to
write instead of getting into an 100% CPU loop by just returning and
being called again by the scheduler
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Original commit message from CVS:
* configure.ac:
* ext/libvisual/visual.c: (gst_visual_get_type),
(libvisual_log_handler), (gst_visual_getcaps),
(gst_visual_srclink), (gst_visual_change_state), (make_valid_name),
(plugin_init):
Update libvisual to 0.1.7. Link in the debug handling to gstreamer
* ext/smoothwave/Makefile.am:
* ext/smoothwave/demo-osssrc.c: (main):
* ext/smoothwave/gstsmoothwave.c: (gst_smoothwave_class_init),
(gst_smoothwave_init), (gst_smoothwave_dispose), (gst_sw_sinklink),
(gst_sw_srclink), (gst_smoothwave_chain), (gst_sw_change_state),
(plugin_init):
* ext/smoothwave/gstsmoothwave.h:
Make gstsmoothwave a working element in the 20th century.
* gst/chart/gstchart.c: (gst_chart_init), (gst_chart_srcconnect):
Fix incorrect link function
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Original commit message from CVS:
* gst/volume/gstvolume.c:
Allow buffer-frames=0.
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Original commit message from CVS:
Added the polypaudio sink and hooked it into the build system.
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when the
Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsa_sink_loop):
Fix for negotiation order problem. This would show when the
ALSA loopfuction was called before any other function. ALSA
wouldn't do anything because we're not negotiated yet, leading
to an infinite loop. Showed in e.g. Rhythmbox. Fixes #158006.
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Original commit message from CVS:
reviewed by: Ronald S. Bultje <rbultje@ronald.bitfreak.net>
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query):
No warnings (#157986).
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file) over musepack.
Original commit message from CVS:
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
Prefer apev1/2 and id3v1 (at end of file) over musepack.
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Original commit message from CVS:
* gst/matroska/matroska-demux.c: (gst_matroska_demux_loop_stream):
Signal no-more-pads (so it works in playbin).
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Original commit message from CVS:
* ext/musepack/gstmusepackreader.cpp:
Workaround for older core.
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than testing whether a temporary variabl...
Original commit message from CVS:
* gst/ffmpegcolorspace/imgconvert.c: (yuv420p_to_yuv422):
Actually test for odd width/height rather than testing whether
a temporary variable that was 0 before we subtracted 1 is now
not equal to zero (which it always is).
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Original commit message from CVS:
2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/v4l2/gstv4l2element.c: (gst_v4l2_iface_supported):
Fix compilation if HAVE_XVIDEO is not defined
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Original commit message from CVS:
2004-11-11 Zaheer Abbas Merali <zaheerabbas at merali dot org>
* sys/v4l/gstv4lelement.c: (gst_v4l_iface_supported):
Fix compilation if HAVE_XVIDEO is not defined
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Original commit message from CVS:
Use bytestream in goom for input samples
Add a debug category
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