Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_drain), (gst_base_audio_sink_event),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
patch to make timestamp checking more tollerant to rounding
errors given that real discontinuities are to be marked on
buffers. Fixes some asf files and #338778.
Also avoid some crashers when we receive an event in the
NULL state.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_init),
(gst_base_audio_src_get_time), (gst_base_audio_src_create):
GstBaseAudioSrc must be live or it does not work.
* gst-libs/gst/audio/gstaudiosrc.c: (gst_audio_src_init):
Don't set live to TRUE as this is the default in the parentclass.
Original commit message from CVS:
2006-04-12 Philippe Kalaf <philippe.kalaf@collabora.co.uk>
* gst-libs/gst/rtp/gstrtpbuffer.h:
Added GST_RTP_PAYLOAD_DYNAMIC_STRING for use by children
* gst-libs/gst/rtp/gstbasertpaudiopayload.c:
* gst-libs/gst/rtp/gstbasertpaudiopayload.h:
New RTP audio base payloader class. Supports frame or sample based codecs
Original commit message from CVS:
Patch by: Antoine Tremblay <hexa00 at gmail dot com>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_finalize), (gst_base_rtp_depayload_push):
Fix some memory leaks: on finalize, free buffers left in the queue
before destroying the queue; in _push(), unref rtp_buf even if
the process vfunc returned a NULL buffer as output buffer (#337548);
demote some recuring debug messages to LOG level.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event):
Starting the ringbuffer when we did not acquire it can cause
a deadlock, is pointless and causes nasty things for
subclasses.
Fixes gst-launch audiotestsrc num-buffers=0 ! alsasink.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
Fix audio sources, forgot to make the ringbuffer
startable...
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_get_time), (gst_base_audio_src_create),
(gst_base_audio_src_change_state):
unparent instead of unref the ringbuffer.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_async_play),
(gst_base_audio_sink_do_play), (gst_base_audio_sink_change_state):
Implement new async_play vmethod to start slaving and allow
playback start in case of async PLAY state changes.
* gst-libs/gst/video/gstvideosink.c: (gst_video_sink_init):
Enable QoS with new method in base class.
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),
(gst_riff_create_video_template_caps):
* gst/ffmpegcolorspace/avcodec.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_pixfmt_to_caps), (gst_ffmpeg_caps_to_pixfmt),
(gst_ffmpegcsp_avpicture_fill):
* gst/ffmpegcolorspace/imgconvert.c:
Patch for support of YVU9 AVI files (#334822)
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_dispose):
Since we _parent the ringbuffer, we also need to
_unparent instead of a plain _unref.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstringbuffer.c: (wait_segment),
(gst_ring_buffer_may_start):
* gst-libs/gst/audio/gstringbuffer.h:
Only start playback if we are playing.
should fix#330748.
Original commit message from CVS:
* gst-libs/gst/tag/gstid3tag.c:
Add TXXX frame identifiers for replaygain stuff as used
by some taggers (see #323721).
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.
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c:
Fix prepare-xwindow-id code example in the docs - we need to
ignore all messages that aren't element messages as well.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Fix playback of non-synchronised streams by assuming a rate
of 1.0 instead of a random one.
Makes this work again:
gst-launch filesrc location=raw_audio.file ! 'audio/x-raw-int,
endianness=(int)4321, signed=(boolean)true, width=(int)16,
depth=(int)16, rate=(int)44100, channels=(int)2' ! audioconvert !
audioresample ! alsasink
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_handle_track_seek):
No need to post a tag message on the bus when seeking
within the same track, only post it when the current
track changes.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_parse_caps):
Set depth and width for alaw/mulaw (fixes#326601).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Name (private) union, makes Forte compiler happy (this time
for real) (#324900).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause), (wait_segment):
* gst-libs/gst/audio/gstringbuffer.h:
Sun's Forte compiler doesn't seem to like anonymous structs,
so use same setup as in GstBaseSrc (fixes#324900).
Original commit message from CVS:
* gst-libs/gst/cdda/gstcddabasesrc.c:
(gst_cdda_base_src_update_duration),
(gst_cdda_base_src_calculate_cddb_id):
An integer is not a string. Fix access to uninitialised variable.
* tests/check/Makefile.am:
Add cddabasesrc unit test; also actually enable the vorbis test.
* tests/check/generic/states.c:
Blacklist new cd audio elements as well.
* tests/check/libs/cddabasesrc.c:
Unit test for GstCddaBaseSrc (discid calculation mostly).
Original commit message from CVS:
* docs/libs/Makefile.am:
* docs/libs/gst-plugins-base-libs-docs.sgml:
* docs/libs/gst-plugins-base-libs-sections.txt:
* docs/libs/gst-plugins-base-libs.types:
Add docs for libgstcdda/GstCddaBaseSrc.
* gst-libs/gst/interfaces/mixertrack.h:
Do one struct member per line with a semicolon at the end, that way
even gtk-doc might parse it without complaining.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_class_init):
update strings, values are in microseconds
change the default sink buffer time to something that is smaller
(to help software volume mixing have a slightly lower delay) but
still be acceptable on Wim's laptop
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_template_caps):
Made a quack, forgot to add DUCK to the riff video template.
Original commit message from CVS:
* ext/ogg/gstogmparse.c: (gst_ogm_text_parse_base_init),
(gst_ogm_parse_init), (gst_ogm_audio_parse_init),
(gst_ogm_video_parse_init), (gst_ogm_text_parse_init),
(gst_ogm_parse_chain):
Make sure pads are initialized correctly.
* gst-libs/gst/riff/riff-ids.h:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add a whole bunch of FOURCC <=> MimeType.
Extend the riff video pad template to support the newly added fourcc.
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init), (gst_base_rtp_depayload_handle_sink_event):
Handle downstream newsegment by sending our own newsegment before the
next buffer to be released. (#323900)
Original commit message from CVS:
2005-12-17 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_set_gst_timestamp):
add queue delay to new segment as well (as opposed to just the first
buffer). (bug #322347)
Original commit message from CVS:
* docs/libs/tmpl/gstcolorbalance.sgml:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertppayload.c:
* gst-libs/gst/rtp/gstrtpbuffer.c:
* gst-libs/gst/rtp/gstrtpbuffer.h:
Do burger's rename for rtp payloaders and depayloaders
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/net/README:
* gst-libs/gst/net/gstnetbuffer.c:
* gst-libs/gst/net/gstnetbuffer.h:
this was moved to netbuffer
Original commit message from CVS:
* gst-libs/gst/video/gstvideofilter.c: (gst_video_filter_get_type),
(gst_video_filter_class_init), (gst_video_filter_init):
* gst-libs/gst/video/gstvideofilter.h:
borgify name to bring in line with other classes
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
(libgstnetbuffer_@GST_MAJORMINOR@_la_SOURCES):
Ok, maybe not so quick next time.
Original commit message from CVS:
* gst-libs/gst/netbuffer/Makefile.am: (libgstnetbufferincludedir):
Let's not override libgstnet from core for no reason...
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
moved gst-libs/gst/net to netbuffer through CVS surgery
remove old directory
updating build to accomodate
(#322257)
Original commit message from CVS:
2005-11-29 Andy Wingo <wingo@pobox.com>
* pkgconfig/gstreamer-plugins-base.pc.in:
* pkgconfig/gstreamer-plugins-base-uninstalled.pc.in:
* gst-libs/gst/net/Makefile.am: Rename gstnet to gstnetbuffer
(#322257).
Original commit message from CVS:
* gst-libs/gst/video/Makefile.am:
compile in copied-over videofilter into the video library
* gst-libs/gst/video/videosink.h:
rename the header to gstvideosink.h since it's a base GstObject class
* sys/ximage/ximagesink.h:
* sys/xvimage/xvimagesink.h:
use the new header
Original commit message from CVS:
* gst-libs/gst/audio/TODO:
Updated TODO
* gst-libs/gst/audio/gstaudiosink.c:
(gst_audioringbuffer_open_device),
(gst_audioringbuffer_close_device), (gst_audioringbuffer_acquire),
(gst_audioringbuffer_release):
Small cleanups.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_render),
(gst_base_audio_sink_change_state):
Slave to the master clock when going to PLAYING and unslave when
going to PAUSED.
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_acquire), (gst_ring_buffer_release),
(gst_ring_buffer_samples_done), (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all), (wait_segment),
(gst_ring_buffer_commit), (gst_ring_buffer_read),
(gst_ring_buffer_advance):
* gst-libs/gst/audio/gstringbuffer.h:
Add some docs and cleanups.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_template_caps):
Add ATRAC3 to the list of riff-possible audio caps.
I know we still don't have a plugin for atrac3, but it's saner to output
that than a cryptic mimetype.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
remove silly include
* gst/tags/Makefile.am:
* gst/tags/gsttagediting.c:
* gst/tags/gsttageditingprivate.h:
* gst/tags/tagedit.vcproj:
remove directory, is as good as empty
Original commit message from CVS:
* configure.ac:
added GST_LIB_LDFLAGS and GST_ALL_LDFLAGS
* gst-libs/Makefile.am:
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/interfaces/Makefile.am:
* gst-libs/gst/net/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/rtp/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
and use them
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
If we are reading too slowly, jump forward in the ringbuffer
instead of blocking.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
Fix for calibration API change.
Original commit message from CVS:
* gst-libs/gst/audio/multichannel.c:
(gst_audio_get_channel_positions),
(gst_audio_set_channel_positions),
(gst_audio_set_structure_channel_positions_list),
(gst_audio_fixate_channel_positions):
Use gst_value_array_*() functions on value arrays, not
gst_value_list_*().
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_set_clock),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
And we provide a clock by default, of course...
Original commit message from CVS:
* gst-libs/gst/audio/gstaudioclock.c: (gst_audio_clock_init):
This clock can be slaved to a master clock now.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_class_init), (gst_base_audio_sink_init),
(gst_base_audio_sink_dispose), (gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_set_clock),
(gst_base_audio_sink_set_property),
(gst_base_audio_sink_get_property), (gst_base_audio_sink_preroll),
(gst_base_audio_sink_render), (gst_base_audio_sink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
Handle slaving the internal clock to the clock selected in the
pipeline.
Add property to make the basesink not provide a clock.
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_class_init), (gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_wait):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We can use the clock in GstElement, no need to store it ourselves.
Original commit message from CVS:
* gst-libs/gst/audio/audio.c: (gst_audio_duration_from_pad_buffer):
* gst-libs/gst/audio/audio.h:
fix prototype - wondering why the test worked regardless
Original commit message from CVS:
* gst-libs/gst/interfaces/xoverlay.c: (gst_x_overlay_base_init),
(gst_x_overlay_got_xwindow_id), (gst_x_overlay_prepare_xwindow_id):
* gst-libs/gst/interfaces/xoverlay.h:
Remove everything having to do with the desired size; add
gst_x_overlay_prepare_xwindow_id() function; remove the
'have-xwindow-id' signal and make gst_x_overlay_got_xwindow_id()
post a message on the bus instead (#321816).
* sys/ximage/ximagesink.c: (gst_ximagesink_xoverlay_init):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_setcaps),
(gst_xvimagesink_xoverlay_init):
Remove desired size stuff (#321816).
Original commit message from CVS:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/tag/gstvorbistag.c:
(gst_tag_list_to_vorbiscomment_buffer):
Remove obsolete vorbistag element and debug category.
* gst/playback/gstplaybasebin.c: (check_queue):
Don't divide by 0 when queue-threshold is 0.
* sys/ximage/ximagesink.c: (gst_ximagesink_set_property):
Don't modify an existing pixel-aspect-ratio if we fail to read
a new one.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Fix the audiosrc base class again, we did not unflush.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_provide_clock),
(gst_base_audio_sink_change_state):
Set ringbuffer to non-flushing when going to PAUSED, set to
flushing again when going to READY.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_init),
(gst_ring_buffer_stop):
Start in flushing mode by default.
Don't set flushing in the _stop method, let the app call
this explicitly.
Original commit message from CVS:
2005-11-16 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/gstvideosink.c:
(gst_video_sink_center_rect):
* gst-libs/gst/video/videosink.h: Add helper function needed
for video sinks.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_stop):
Set ringbuffer to flushing when stopping so that we don't
block on wait_segment anymore and livelock.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_init),
(gst_base_rtp_depayload_set_gst_timestamp):
* gst-libs/gst/rtp/gstbasertpdepayload.h:
We need to send a newsegment event for each instance, not
just for the first instance of this class (get rid of
static variable in function). (#321011).
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.h:
Don't break ABI.
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
(gst_ffmpeg_caps_to_pixfmt):
* gst/ffmpegcolorspace/gstffmpegcolorspace.c:
(gst_ffmpegcsp_set_caps):
Some more comments.
Handle missing required caps fields better.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_get_offset),
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ring_buffer_open_device), (gst_ring_buffer_close_device),
(gst_ring_buffer_set_flushing), (gst_ring_buffer_start),
(gst_ring_buffer_pause_unlocked), (gst_ring_buffer_pause),
(gst_ring_buffer_stop), (wait_segment), (gst_ring_buffer_commit),
(gst_ring_buffer_read):
* gst-libs/gst/audio/gstringbuffer.h:
Add flushing mode to the ringbuffer so that it in all cases does
not try to handle more audio. This makes sure it does not try to
block anymore when flushing and fixes a livelock.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Remove g_print
Use sync property from baseclass to disable sync.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_get_offset), (gst_base_audio_sink_render):
Buffers with no timestamps get aligned with previous buffers or
on underrun, played ASAP.
Original commit message from CVS:
2005-10-24 Julien MOUTTE <julien@moutte.net>
* gst-libs/gst/video/video.h:
* gst/ffmpegcolorspace/gstffmpegcodecmap.c:
* sys/ximage/ximagesink.c: (gst_ximagesink_xcontext_get):
* sys/xvimage/xvimagesink.c: (gst_xvimagesink_get_xv_support):
And
here comes my change on caps for framerate and geometry range.
We are now accepting 1 to MAXINT for width and height, and from
0.0 to MAXDOUBLE for framerate. That allows duration less png
frames
to be blended correctly in videomixer.
Original commit message from CVS:
* ext/theora/theoradec.c: (theora_dec_src_query),
(theora_dec_sink_event):
* ext/vorbis/vorbisdec.c: (vorbis_dec_src_query),
(vorbis_dec_sink_event), (vorbis_handle_identification_packet),
(vorbis_handle_data_packet):
* ext/vorbis/vorbisdec.h:
Fix old naming.
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Don't try to sync on buffers without a timestamp.
Original commit message from CVS:
Reviewed by: Tim-Philipp Müller <tim at centricular dot net>
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_video_template_caps):
Add support for Indeo-3 (IV32).
Original commit message from CVS:
* ext/vorbis/vorbisenc.c:
* gst-libs/gst/tag/gstid3tag.c: (gst_tag_list_new_from_id3v1):
* gst-libs/gst/tag/gstvorbistag.c: (gst_vorbis_tag_add),
(gst_tag_to_vorbis_comments):
Fix handling of GST_TAG_DATE, which is now of GST_TYPE_DATE.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read), (gst_ring_buffer_clear):
Don't assert on normal stuff.
* gst/playback/gstplaybin.c: (do_playbin_seek):
API fix.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit),
(gst_ring_buffer_read):
Cleanups.
Commit and read from ringbuffer in samples rather than bytes.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Respect segment rate and accum when scheduling samples.
Original commit message from CVS:
* ext/alsa/gstalsasink.c:
Also allow unsigned int.
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
Small cleanup
Original commit message from CVS:
2005-10-09 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/rtpbasedepayload.c:
Set timestamp and add queue delay to timestamp
* gst-libs/gst/rtp/rtpbuffer.h:
Set correct payload type for h263
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_stop):
Only actually wait for the thread to be stopped if it's
running.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_event), (gst_base_audio_sink_render):
If we receive EOS we can start playback of what we had.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c: (gst_base_audio_src_event),
(gst_base_audio_src_create), (gst_base_audio_src_change_state):
* gst-libs/gst/audio/gstbaseaudiosrc.h:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_read):
patch from Edgard Lima <edgard.lima@indt.org.br>
Fixed gstbaseaudiosrc adding ring buffer sync to it.
Original commit message from CVS:
* ext/ogg/gstoggdemux.c: (gst_ogg_demux_loop):
Report the FLOW_RETURN as string in the error message.
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_clear_all):
Don't assert when clearing an unnegotiated buffer.
Original commit message from CVS:
2005-10-02 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_clear)
(gst_ring_buffer_prepare_read):
* gst-libs/gst/audio/gstaudiosink.c (audioringbuffer_thread_func):
Demote to LOG.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertppayload.c:
(gst_basertppayload_class_init), (gst_basertppayload_init),
(gst_basertppayload_setcaps), (gst_basertppayload_chain),
(gst_basertppayload_set_options), (gst_basertppayload_set_outcaps),
(gst_basertppayload_is_filled), (gst_basertppayload_push),
(gst_basertppayload_set_property),
(gst_basertppayload_get_property),
(gst_basertppayload_change_state):
* gst-libs/gst/rtp/gstbasertppayload.h:
Added max-ptime to control amount of data in the rtp packets.
Original commit message from CVS:
* gst-libs/gst/rtp/gstbasertpdepayload.c:
(gst_base_rtp_depayload_add_to_queue),
(gst_base_rtp_depayload_push),
(gst_base_rtp_depayload_set_gst_timestamp),
(gst_base_rtp_depayload_queue_release):
This one was not supposed to go in.
Original commit message from CVS:
* check/generic/states.c:
* ext/ogg/gstoggdemux.c: (gst_ogg_pad_typefind):
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Fixes for changes in registry API.
* configure.ac: Only export gst_plugins_desc. Add -no-undefined
to GST_PLUGIN_LDFLAGS.
* ext/libvisual/visual.c: Make the library shut up.
* gst-libs/gst/audio/audio.c: Don't define a plugin in a library.
* gst-libs/gst/audio/gstaudiofilter.c: same
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_commit):
* gst/volume/gstvolume.c: (gst_volume_class_init),
(volume_transform):
fixing lost sync, some more debugging
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Resync if the buffer timestamps drift more than a 10th
of a second.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ring_buffer_set_sample),
(gst_ring_buffer_clear_all):
* gst-libs/gst/audio/gstringbuffer.h:
Added function to clear the ringbuffer.
Original commit message from CVS:
2005-08-23 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosrc.c
(gst_base_audio_src_fixate): Only fixate endianness if it is
present in the caps.
Original commit message from CVS:
2005-08-22 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstaudiosrc.h:
* gst-libs/gst/audio/gstaudiosrc.c: Implement open_device and
close_device in the ring buffer, like gstaudiosink.
* ext/alsa/gstalsamixer.h:
* ext/alsa/gstalsamixer.c: Not a GObject any more. Include a nifty
macro to implement the interface without much code. Cleanups.
* ext/alsa/gstalsasrc.h:
* ext/alsa/gstalsasrc.c: Be a mixer. Open device and mixer in
READY.
* ext/alsa/Makefile.am: Add new files.
* ext/alsa/gstalsamixerelement.c:
* ext/alsa/gstalsamixerelement.c: Split element code out from
mixer code so that alsasrc can be a mixer too.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosrc.c:
(gst_base_audio_src_change_state):
Open and close device in READY<->NULL state change.
Original commit message from CVS:
2005-08-12 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Made a thread to release the queue.
Removed timestamp conversion for now.
Original commit message from CVS:
2005-08-10 Philippe Khalaf <burger@speedy.org>
* gst-libs/gst/rtp/gstbasertpdepayload.c:
* gst-libs/gst/rtp/gstbasertpdepayload.h:
Added rtp timestamp -> gst timestamp conversion.
Fixed several problems with queue.
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_chunk),
(gst_riff_parse_chunk):
Fix bug in debug message and add some more debug messages.
Original commit message from CVS:
2005-08-08 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/audio/gstbaseaudiosink.c
(gst_base_audio_sink_change_state): Open the device in NULL->READY
like good elements should. Close on READY->NULL too.
* gst-libs/gst/audio/gstaudiosink.c
(gst_audioringbuffer_open_device,
(gst_audioringbuffer_close_device, gst_audioringbuffer_acquire)
(gst_audioringbuffer_release): Updates for new ring buffer API,
hook into the new audio sink api.
* gst-libs/gst/audio/gstaudiosink.h (GstAudioSinkClass.open)
(GstAudioSinkClass.close): Just open and close the device -- no
resource allocation or configuration.
(GstAudioSinkClass.prepare, GstAudioSinkClass.unprepare): New
vmethods, handle device setup and resource allocation.
* ext/alsa/gstalsasink.c (gst_alsasink_open, gst_alsasink_close)
(gst_alsasink_prepare, gst_alsasink_unprepare): Update for new
base class API.
* gst-libs/gst/audio/gstringbuffer.h
(GstRingBufferClass.open_device, GstRingBufferClass.close_device):
New vmethods.
* gst-libs/gst/audio/gstringbuffer.c (gst_ring_buffer_open_device)
(gst_ring_buffer_close_device, gst_ring_buffer_device_is_open):
New API functions. The device should be opened before acquiring
and closed after releasing.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/gconf/.cvsignore:
* gst-libs/gst/gconf/Makefile.am:
* gst-libs/gst/gconf/test-gconf.c:
* pkgconfig/Makefile.am:
* pkgconfig/gstreamer-gconf-uninstalled.pc.in:
* pkgconfig/gstreamer-gconf.pc.in:
Remove gconf stuff, use gconf elements instead from now on.
Original commit message from CVS:
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_base_audio_sink_render):
Align samples even if we have roundoff errors in the
timestamp conversion.
Original commit message from CVS:
* gst-libs/gst/rtp
* gst-libs/gst/rtp/gstbasertpdepayload.c
* gst-libs/gst/rtp/gstbasertpdepayload.h
* gst-libs/gst/rtp/gstrtpbuffer.c
* gst-libs/gst/rtp/gstrtpbuffer.h
* gst-libs/gst/rtp/Makefile.am
* gst-libs/gst/rtp/README
Support libs for RTP. Basicaly this add a GstRTPBuffer (extended GstBuffer) and
a Depayloader Base class that shall be used by payload specific depayloaders.
Original commit message from CVS:
make GST_PLUGIN_LDFLAGS only be flags; GST_LIBS should be
added manually to each Makefile.am so we are sure it goes
*last* and doesn't add -L flags before linking in libs of our
own, like, say, internal .la libs, that then accidentally pick
up the installed copy.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_video_caps),
(gst_riff_create_audio_caps), (gst_riff_create_iavs_caps),
(gst_riff_create_video_template_caps),
(gst_riff_create_audio_template_caps),
(gst_riff_create_iavs_template_caps):
* gst-libs/gst/riff/riff-media.h:
* gst-libs/gst/riff/riff-read.h:
* gst-libs/gst/riff/riff.c: (gst_riff_init):
Add gst_riff_init() to initialize the debug category, instead
of plugin_init(). Port riff-media.[ch] from -THREADED to HEAD.
Original commit message from CVS:
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_init):
* gst-libs/gst/audio/gstbaseaudiosink.c:
(gst_baseaudiosink_class_init), (gst_baseaudiosink_dispose),
(gst_baseaudiosink_change_state):
* gst-libs/gst/audio/gstbaseaudiosink.h:
* gst-libs/gst/audio/gstringbuffer.c:
(gst_ringbuffer_set_callback):
Fix compilation error.
Ringbuffer starts out as not running.
Free our clock in dispose.
When releasing the ringbuffer we need to renegotiate so
clear the pad caps.
Original commit message from CVS:
2005-06-25 Jan Schmidt <thaytan@mad.scientist.com>
* gst-libs/gst/audio/gstaudiosink.c: (gst_audioringbuffer_acquire):
Set the worker thread's running flag to TRUE before starting the
thread.
* gst/playback/gstdecodebin.c: (gst_decode_bin_init):
Catch a failure to add typefind to the bin.
Original commit message from CVS:
2005-06-09 Andy Wingo <wingo@pobox.com>
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/net/Makefile.am:
Add gstnet to build.
Original commit message from CVS:
* gst-libs/gst/net/Makefile.am:
* pkgconfig/gstreamer-libs-uninstalled.pc.in:
* pkgconfig/gstreamer-libs.pc.in:
Added net stuff, version net lib.
Original commit message from CVS:
* gst-libs/gst/audio/gstringbuffer.c: (gst_ringbuffer_delay):
Don't try to call the delay method when the device is not
opened.
Original commit message from CVS:
Make ringbuffer faster and more simple by removing the locks
in the playback thread.
Add sample accurate playback based on buffer sample offsets.
Make the baseaudiosink provide a clock.
Parse caps in the base class.
Correctly handle seeking, flushing and state changes.
Original commit message from CVS:
* configure.ac:
* gst-libs/gst/Makefile.am:
* gst-libs/gst/xwindowlistener/Makefile.am:
* gst-libs/gst/xwindowlistener/xwindowlistener.c:
* gst-libs/gst/xwindowlistener/xwindowlistener.h:
Remove deprecated xwindowlistener (I've moved xwindowlistening
in the v4l/v4l2 plugins over to serverside).
Original commit message from CVS:
Don't use GST_PLUGIN_LDFLAGS, because these aren't plugins.
* gst-libs/gst/audio/Makefile.am:
* gst-libs/gst/riff/Makefile.am:
* gst-libs/gst/tag/Makefile.am:
* gst-libs/gst/video/Makefile.am:
* gst-libs/gst/xwindowlistener/Makefile.am:
Convert to 0.9 API, seems to work:
* sys/ximage/Makefile.am:
* sys/ximage/ximagesink.c:
Original commit message from CVS:
* gst-libs/gst/Makefile.am: Remove idct. It hasn't been used
in gst-plugins in a long time, and properly belongs in liboil.
* gst-libs/gst/idct/Makefile.am:
* gst-libs/gst/idct/README:
* gst-libs/gst/idct/dct.h:
* gst-libs/gst/idct/doieee:
* gst-libs/gst/idct/fastintidct.c:
* gst-libs/gst/idct/floatidct.c:
* gst-libs/gst/idct/idct.c:
* gst-libs/gst/idct/idct.h:
* gst-libs/gst/idct/idtc.vcproj:
* gst-libs/gst/idct/ieeetest.c:
* gst-libs/gst/idct/intidct.c:
Original commit message from CVS:
2005-04-06 Andy Wingo <wingo@pobox.com>
* gst-libs/gst/video/Makefile.am (libgstvideo_la_LDFLAGS): Use
GST_BASE_LIBS.
Original commit message from CVS:
Plugin port to 0.9, ogg/theora playback should work in the seek
example now.
Removed old examples.
Removed old oggvorbisenc, renamed rawvorbisenc to vorbisenc as
explained in 0.9 TODO doc.
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)
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.
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.
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.
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)
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).
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.
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.
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
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.
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).
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).
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.
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
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.
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
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
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
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 wing commander format mimetype/fourccs.
* gst/avi/gstavidemux.c: (gst_avi_demux_massage_index):
Don't crash if some value is 0.
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 DIB fourcc (raw, palettized 8-bit RGB).
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data):
Oops, fix strf_data reading bug.
* gst/avi/gstavidemux.c: (gst_avi_demux_add_stream):
Use a non-NULL tag.
* gst/qtdemux/qtdemux.c: (qtdemux_parse_trak):
Time for hacks. Sorry Dave. At least one quicktime movie (a
trailer) that I've encountered contains multiple video tracks.
One of those is the actual video track, the other are one-frame
tracks (images). Unfortunately, the number of frames according
to the trak header is 1 for each, so that doesn't help. So
instead, I look at the duration and discard tracks with a
duration shorter than 20% of the length of the stream. Better
than nothing.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data),
(gst_riff_create_audio_caps_with_data):
Add codec_data handling (like asfdemux used to do).
* gst/asfdemux/gstasf.c: (plugin_init):
* gst/asfdemux/gstasfdemux.c: (gst_asf_demux_base_init),
(gst_asf_demux_add_audio_stream), (gst_asf_demux_add_video_stream):
Use riff-media for caps creation instead of our own (mostly
broken) copy of its functions.
Original commit message from CVS:
* ext/dvdread/dvdreadsrc.c: (dvdreadsrc_class_init),
(dvdreadsrc_init), (dvdreadsrc_dispose), (dvdreadsrc_set_property),
(dvdreadsrc_get_property), (_open), (_seek), (_read),
(dvdreadsrc_get), (dvdreadsrc_open_file),
(dvdreadsrc_change_state):
Fix. Don't do one big huge loop around the whole DVD, that will
cache all data and thus eat sizeof(dvd) (several GB) before we
see something.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Actually NULL'ify event after using it.
* gst/matroska/ebml-read.c: (gst_ebml_read_use_event),
(gst_ebml_read_handle_event), (gst_ebml_read_element_id),
(gst_ebml_read_element_length), (gst_ebml_read_element_data),
(gst_ebml_read_seek), (gst_ebml_read_skip):
Handle events.
* gst/mpegstream/gstdvddemux.c: (gst_dvd_demux_base_init),
(gst_dvd_demux_init), (gst_dvd_demux_get_audio_stream),
(gst_dvd_demux_get_subpicture_stream), (gst_dvd_demux_plugin_init):
Fix timing (this will probably break if I seek using menus, but
I didn't get there yet). VOBs and normal DVDs should now work.
Add a mpeg2-only pad with high rank so this get autoplugged for
MPEG-2 movies.
* gst/mpegstream/gstmpegdemux.c: (gst_mpeg_demux_base_init),
(gst_mpeg_demux_class_init), (gst_mpeg_demux_init),
(gst_mpeg_demux_new_output_pad), (gst_mpeg_demux_get_video_stream),
(gst_mpeg_demux_get_audio_stream),
(gst_mpeg_demux_get_private_stream), (gst_mpeg_demux_parse_packet),
(gst_mpeg_demux_parse_pes), (gst_mpeg_demux_plugin_init):
Use this as second rank for MPEG-1 and MPEG-2. Still use this for
MPEG-1 but use dvddemux for MPEG-2.
* gst/mpegstream/gstmpegparse.c: (gst_mpeg_parse_class_init),
(gst_mpeg_parse_init), (gst_mpeg_parse_new_pad),
(gst_mpeg_parse_parse_packhead):
Timing. Only add pad template if it exists. Add sink template from
class and not from ourselves. This means we will always use the
correct sink template even if it is not the one defined in this
file.
Original commit message from CVS:
* ext/flac/gstflacdec.c: (gst_flacdec_src_query):
Only return true if we actually filled something in. Prevents
player applications from showing a random length for flac files.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_class_init),
(gst_riff_read_use_event), (gst_riff_read_handle_event),
(gst_riff_read_seek), (gst_riff_read_skip), (gst_riff_read_strh),
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data), (gst_riff_read_strf_iavs):
OK, ok, so I implemented event handling. Apparently it's normal
that we receive random events at random points without asking
for it.
* gst/avi/gstavidemux.c: (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_handle_src_event), (gst_avi_demux_stream_index),
(gst_avi_demux_sync), (gst_avi_demux_stream_scan),
(gst_avi_demux_massage_index), (gst_avi_demux_stream_header),
(gst_avi_demux_handle_seek), (gst_avi_demux_process_next_entry),
(gst_avi_demux_stream_data), (gst_avi_demux_loop):
* gst/avi/gstavidemux.h:
Implement non-lineair chunk handling and subchunk processing.
The first solves playback of AVI files where the audio and video
data of individual buffers that we read are not synchronized.
This should not happen according to the wonderful AVI specs, but
of course it does happen in reality. It is also a prerequisite for
the second. Subchunk processing allows us to cut chunks in small
pieces and process each of these pieces separately. This is
required because I've seen several AVI files with incredibly large
audio chunks, even some files with only one audio chunk for the
whole file. This allows for proper playback including seeking.
This patch is supposed to fix all AVI A/V sync issues.
* gst/flx/gstflxdec.c: (gst_flxdec_class_init),
(flx_decode_chunks), (flx_decode_color), (gst_flxdec_loop):
Work.
* gst/modplug/gstmodplug.cc:
Proper return value setting for the query() function.
* gst/playback/gstplaybasebin.c: (setup_source):
Being in non-playing state (after, e.g., EOS) is not necessarily
a bad thing. Allow for that. This fixes playback of short files.
They don't actually playback fully now, because the clock already
runs. This means that small files (<500kB) with a small length
(<2sec) will still not or barely play. Other files, such as mod
or flx, will work correctly, however.
Original commit message from CVS:
* ext/dirac/Makefile.am:
* ext/dirac/gstdirac.cc:
* ext/dirac/gstdiracdec.cc:
* ext/dirac/gstdiracdec.h:
Do something. Don't actually know if this works because I don't
have a demuxer yet.
* ext/gsm/gstgsmdec.c: (gst_gsmdec_getcaps):
Add channels=1 to caps returned from _getcaps().
* ext/ogg/gstogmparse.c: (gst_ogm_audio_parse_get_type),
(gst_ogm_video_parse_get_type), (gst_ogm_audio_parse_base_init),
(gst_ogm_video_parse_base_init), (gst_ogm_parse_init),
(gst_ogm_audio_parse_init), (gst_ogm_video_parse_init),
(gst_ogm_parse_sink_convert), (gst_ogm_parse_chain),
(gst_ogm_parse_change_state):
Separate between audio/video so ogmaudioparse actually uses the
audio pad templates. Both audio and video work now, including
autoplugging. Also use sometimes-srcpad hack.
* gst-libs/gst/riff/riff-read.c: (gst_riff_read_seek):
Handle events better. Don't hang on infinite loops.
* gst/avi/gstavidemux.c: (gst_avi_demux_class_init),
(gst_avi_demux_init), (gst_avi_demux_reset),
(gst_avi_demux_src_convert), (gst_avi_demux_handle_src_query),
(gst_avi_demux_stream_header), (gst_avi_demux_stream_data),
(gst_avi_demux_change_state):
* gst/avi/gstavidemux.h:
Improve A/V sync. Still not perfect.
* gst/matroska/ebml-read.c: (gst_ebml_read_seek),
(gst_ebml_read_skip):
Handle events better.
* gst/qtdemux/qtdemux.c: (gst_qtdemux_handle_sink_event),
(gst_qtdemux_loop_header), (qtdemux_parse_trak),
(qtdemux_audio_caps):
Add IMA4. Improve event handling. Save offset after a seek when
the headers are at the end of the file so that we don't end up in
an infinite loop.
* gst/typefind/gsttypefindfunctions.c: (qt_type_find):
Add low-priority typefind support for files with no length.
Original commit message from CVS:
2004-08-24 Sebastien Cote <sc5@hermes.usherb.ca>
* gst-libs/gst/riff/riff-read.c: (gst_riff_peek_head),
(gst_riff_read_element_data), (gst_riff_read_seek),
(gst_riff_read_skip): fix infinite loop in wavparse, fixes bug
#144616, patch reviewed by Ronald and committed by Christophe Fergeau
<teuf@gnome.org>
Original commit message from CVS:
Remove GPL'ed mmx32idct.c code and supporting code, since logic in gst-plugins
is not supposed to be GPL'ed. This code provided MMX optimisations, but was
never compiled in since configure never set HAVE_LIBMMX anyway.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_audio_caps_with_data):
Fix double end-to-native symbol conversion (#148021).
Original commit message from CVS:
* gst-libs/gst/riff/riff-read.c:
(gst_riff_read_strf_vids_with_data),
(gst_riff_read_strf_auds_with_data):
* gst/avi/gstavidemux.c: (gst_avi_demux_handle_src_query),
(gst_avi_demux_add_stream), (gst_avi_demux_stream_header):
Make sure we don't create 0 sized subbufers in riff-read.
Signal the no more pads signal after reading the avi header.
Original commit message from CVS:
* gst-libs/gst/video/video.h:
Added 32 bits RGBA. Not sure if we should use another mime-type
for alpha rgb. Currently the presence of the alpha_mask property
signals an alpha channel. Ronald?
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c:
(gst_riff_create_video_caps_with_data):
mp42/mp43 (no caps) exist too.
* gst/matroska/matroska-demux.c: (gst_matroska_demux_video_caps):
Set pixel_width/height; we've got them in-caps.
* gst/typefind/gsttypefindfunctions.c: (plugin_init):
* gst/wavparse/gstwavparse.c: (plugin_init):
Both are valid primary.
* sys/oss/gstossmixer.c:
Remove i18n hack and enable translations.
Original commit message from CVS:
* gst-libs/gst/riff/riff-media.c: (gst_riff_create_audio_caps),
(gst_riff_create_video_template_caps):
Fix the template caps to include some more media types.
Original commit message from CVS:
* gst-libs/gst/play/play.c: (gst_play_pipeline_setup),
(gst_play_get_length_callback), (gst_play_set_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_sink_element):
- add debugging info
- fix looking up sink elements by iterating over complete caps
- put everything except for source and autoplugger in a complete bin
