Original commit message from CVS:
* ext/alsa/gstalsasink.c: (gst_alsasink_finalise):
Chain up to the parent finalize method.
Add 32-bit sample size to the template caps.
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add the fourcc that the VMWare codec uses.
* gst/playback/gststreamselector.c:
(gst_stream_selector_set_property),
(gst_stream_selector_bufferalloc),
(gst_stream_selector_request_new_pad):
For the active pad, forward buffer-alloc requests, otherwise
return GST_FLOW_NOT_LINKED. This also prevents xvimagesink
having to memcpy every frame when used by playbin.
* gst/tcp/gstmultifdsink.c:
(gst_multi_fd_sink_handle_client_write):
Get negotiated caps from the sink pad, rather than the sink
pad's peer.
Original commit message from CVS:
Patch by: Fabrizio Gennari <fabrizio dot ge at tiscali dot it>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure the buffer we copy into is really always big
enough, this time for real (#333488).
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_init):
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c: (gst_ffmpegcsp_init):
* gst/videoscale/gstvideoscale.c: (gst_video_scale_class_init),
(gst_video_scale_init), (gst_video_scale_src_event):
Re-enable QoS after the release.
Rework videoscale to use the base class src_event handler.
Original commit message from CVS:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
* docs/plugins/gst-plugins-base-plugins.hierarchy:
Add cdparanoiasrc to docs.
* gst-libs/gst/cdda/gstcddabasesrc.c:
More GstCddaBaseSrc docs.
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-sections.txt:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_from_id3_user_tag):
* gst-libs/gst/tag/tag.h:
Add new API to libgsttag: gst_tag_from_id3_user_tag().
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Disable max-lateness by setting it to -1 for now, so that
we can bed QoS stuff in thoroughly between now and the next
release.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Make sure we don't read beyond the palette buffer in case of
broken or manipulated files (#333488, patch by: Fabrizio
Gennari)
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Allow palettes with less than 256 colours in AVI files
(#333488, patch by: Fabrizio Gennari).
Original commit message from CVS:
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init),
(gst_video_sink_class_init):
Throw away frames that are later than 20 ms.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set depth on WMA caps (#333545, patch by: Fabrizio Gennari).
Original commit message from CVS:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
* docs/plugins/Makefile.am:
* docs/plugins/gst-plugins-base-plugins-docs.sgml:
* docs/plugins/gst-plugins-base-plugins-sections.txt:
Added some more docs to libs and plugins.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_prepare_read), (gst_ring_buffer_clear):
* gst-libs/gst/audio/gstringbuffer.h:
Document ringbuffer some more.
* gst/videorate/gstvideorate.c: (gst_video_rate_class_init),
(gst_video_rate_setcaps), (gst_video_rate_reset),
(gst_video_rate_init), (gst_video_rate_flush_prev),
(gst_video_rate_swap_prev), (gst_video_rate_event),
(gst_video_rate_chain), (gst_video_rate_change_state):
* gst/videorate/gstvideorate.h:
Fix videorate to use segments.
Make it work with 0/1 framerates (closes#331903)
Handle EOS correctly.
Added docs.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock):
Don't try to provide a clock in the NULL state.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps):
Pick up palette for MS video v1 (#327028, patch by:
Fabrizio Gennari <fabrizio dot get at tiscali dot it>)
Original commit message from CVS:
* gst-libs/gst/audio/mixerutils.c:
(element_factory_rank_compare_func):
Make order in which elements are tried more determinable.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Update TODO
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset):
When trying to play samples ASAP and we don't have a
previous sample, try to play at position 0 instead of
an invalid position.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps):
Set codec_name for WAVEFORMATEX as well. When we have 'normal PCM'
with 4 or 6 channels, assume a default channel layout to make things
work (not sure there's anything else we can do in those cases).
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
Minor docs fix.
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_wavext_add_channel_layout), (gst_riff_create_audio_caps):
Add support for WAVEFORMATEX, eg. PCM audio with more than two
channels and a channel layout map.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions):
When we have more than 2 channels, but no channel layout is
specified in the caps, return some default channel layout
to the caller and warn about about a possibly buggy element
(could be buggy filtercaps as well of course) (#317038).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_delay),
(gst_ring_buffer_samples_done), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_clear):
Add some compiler G_(UN_)LIKELY help.
SIGNAL the ringbuffer waiters when going to PAUSED as well to
make sure they can exit their functions. Should fix#330748
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Always sync on first sample we receive when starting.
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c: (gst_cdda_base_src_create):
No need to push an EOS event here, GstBaseSrc will do that for us
when we return FLOW_UNEXPECTED.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_time), (gst_base_audio_sink_setcaps),
(gst_base_audio_sink_drain), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Use scale functions when possible.
Fix error messages.
Free clockid when after waiting for EOS.
Use G_(UN_)LIKLY when it makes sense.
Fix sample clipping bug found by Arwed v. Merkatz fixes#330789.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Added FLV1 <==> 'video/x-flash-video,flvversion=1' conversion.
Also added the caps to the default set of riff video caps.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_setcaps), (gst_basertppayload_push):
update seqnum before setting it on the packet; this makes sure
that the timestamp and seqnum properties match after pushing
a buffer
Original commit message from CVS:
2006-02-09 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c
(gst_ring_buffer_samples_done): Cast to guint64, fixes an integer
overflow after 13.5 hours of recording. Kapow!
* ext/alsa/gstalsasrc.c (gst_alsasrc_delay): Clamp the delay to
the buffer size -- we don't care about underrun/overrun reporting
right now, just need to return a useful value.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_tag_from_vorbis_tag),
(gst_tag_to_vorbis_tag):
Make sure we called gst_tag_register_musicbrainz_tags()
before possibly mapping a vorbiscomment string from/to a
musicbrainz tag.
Original commit message from CVS:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
Add comment about LANGUAGE tag inconsistency (we want
ISO-639-1, but extract three-letter identifiers?)
* po/POTFILES.in:
Add two translatable files.
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add):
* gst-libs/gst/tag/tag.h:
* gst-libs/gst/tag/tags.c:
(gst_tag_register_musicbrainz_tags_internal),
(gst_tag_register_musicbrainz_tags):
Forward-port some tags stuff from the 0.8 branch. This is
mostly the addition of musicbrainz tags and their mapping
to vorbistags, and a vorbistag mapping of the language tag.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_render):
Don't try to provide a clock when we are not negotiated since
we might not be able to make it run.
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event):
On EOS, wait till the last sample is played before posting EOS.
Original commit message from CVS:
2006-02-01 Philippe Kalaf <burger at speedy dot org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
Patch by Kai Vehmanen : Adds ability to enable newsegment bypass by
setting queue_delay to zero. Also avoid thread being started if
queue_delay is zero.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_class_init), (gst_audioringbuffer_release),
(gst_audioringbuffer_pause):
Implement pause that does not wait for completion.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Don't drop buffers when going to PAUSED but perform preroll on
remaining samples now that core base class supports this.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_release),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_stop),
(gst_ring_buffer_commit):
Pause should not signal waiters.
Implement return value of _commit correctly.
Original commit message from CVS:
* gst-libs/gst/audio/audio.h: (GST_CLOCK_TIME_TO_FRAMES)
* gst-libs/gst/audio/gstbaseaudiosink.c: (gst_base_audio_sink_render)
use of gst_guint64_to_gdouble to be compliant with vs6
* gst/playback/gstdecodebin.c: (try_to_link_1)
* gst/videorate/videorate.c: (gst_video_rate_blank_data)
use of G_GINT64_CONSTANT for int64 constants
* win32/common/libgstinterfaces.def:
export some symbols (gst_mixer_get_type,gst_mixer_track_get_type)
* win32/vs6:
update and add new project files
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Undo previous commit, it breaks resume after pause.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_setcaps), (gst_base_audio_sink_event),
(gst_base_audio_sink_preroll), (gst_base_audio_sink_render):
Improve debugging.
Post error when caps cannot be parsed.
Resync on discontinuity in the stream.
Clip samples to segment boundaries.
return WRONG_STATE sooner when we are flushing.
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
Make audiosrc operate in TIME.
Set TIMESTAMP and DURATION on buffers.
