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Author SHA1 Message Date
Tim-Philipp Müller
bcb8068e27 docs: remove outdated and pointless 'Last reviewed' lines from docs 
They are very confusing for people, and more often than not
also just not very accurate. Seeing 'last reviewed: 2005' in
your docs is not very confidence-inspiring. Let's just remove
those comments.
 2014-04-26 23:28:57 +01:00
Vincent Penquerc'h
7618699ffd audiobasesink: avoid possible sample count overflow 
At 48 kHz, 2<<31 samples is reached before 13 hours so it
sounds plausible this would be hit.

Coverity 1139800, 1139801
 2014-04-10 11:06:00 +01:00
Vincent Penquerc'h
169166d0a2 audiobasesink: clip start samples to match clipped start time 
Clock slaving can clip start time to zero, giving us a shorted
duration than we originally got. To keep in sync, we must then
discard the samples falling before that zero timestamp.

This possibly fixes random distortion caused by constant PA
underflows which are never resynced.
 2014-04-04 17:04:06 +01:00
Wim Taymans
6a88d6f8cd audiobasesink: make _get_time more threadsafe 
We call the _get_time function from the provided clock and we don't lock
the sink object for performance reasons. Make sure we only read and
check variables once so that they don't change while we are executing
the code.

Fixes https://bugzilla.gnome.org/show_bug.cgi?id=720661
 2014-01-21 11:25:18 +01:00
Jan Schmidt
c24a1254c9 audiodecoder: Choose a default initial caps before sending GAP 
If there are no caps from the audio decoder when handling a GAP
event - as when one is received right at the start on a DVD without
initial audio - then choose any default caps for downstream and
then send the GAP, so the audio sink has a configured format in
which to start the ringbuffer.

Also, make the audio sink reject a GAP without caps with a clearer
error message.

Fixes bug https://bugzilla.gnome.org/show_bug.cgi?id=603921
 2013-12-27 04:04:45 +11:00
Reynaldo H. Verdejo Pinochet
21190b9749 gstaudiobasesink: Always reset last_align 
Should be done for all the reset_sync() cases. Not
only for the READY to PAUSED one.
 2013-12-20 18:06:25 -03:00
Reynaldo H. Verdejo Pinochet
032779ff13 gstaudiobasesink: Reset last_align to 0, not -1 
This is the expected behavior in READY -> PAUSED
 2013-12-20 18:02:42 -03:00
Reynaldo H. Verdejo Pinochet
c1de7cdefb gstaudiobasesink: Always reset avg_skew on _reset 
Only case in which it wasn't (READY to PAUSED) should
have had this value reseted too.
 2013-12-20 17:58:43 -03:00
Reynaldo H. Verdejo Pinochet
adf800087c gstaudiobasesink: Retarget FIXME to 2.0 
Properly fixing this one would break API
 2013-12-20 17:48:22 -03:00
Reynaldo H. Verdejo Pinochet
d35db35258 gstaudiobasesink: Factor out reset sync routine 2013-12-20 17:47:38 -03:00
Reynaldo H. Verdejo Pinochet
b324d67586 gstaudiobasesink: Drop dead _sink_async_play() code 2013-12-20 13:58:34 -03:00
Reynaldo H. Verdejo Pinochet
2f04733a4b gstaudiobasesink: Break some too long lines 2013-12-20 13:58:33 -03:00
Reynaldo H. Verdejo Pinochet
187b106202 gstaudiobasesink: Cosmetics, grammar/spelling 
- Drop repeated 'yet' from debug msg
- Drop repeated 'to' from param desc
- Some spelling
 2013-12-20 13:58:33 -03:00
Reynaldo H. Verdejo Pinochet
86b0a0d6d0 gstaudiobasesink: Refactor alignment computation for clarity 2013-12-19 18:05:44 -03:00
Wim Taymans
df3718ea2b audiobasesink: handle the RESYNC flag 
Also resync when a buffer with the RESYNC flag is seen.
 2013-12-05 16:27:35 +01:00
Wim Taymans
c9ff3e4f98 audiobasesink: do big correction for large drift 
If we are using skew slaving and we drift more than twice the allowed amount, do
a big correction to get back on track more quickly.
 2013-09-25 16:03:07 +02:00
Sebastian Dröge
3f82e919dd libs: Use foo/foo.h as single-include header consistently everywhere 
https://bugzilla.gnome.org/show_bug.cgi?id=688785
 2012-12-12 17:13:10 +00:00
Tim-Philipp Müller
5f59b4f7ee Fix FSF address 
https://bugzilla.gnome.org/show_bug.cgi?id=687520
 2012-11-03 23:05:09 +00:00
Miguel Angel Cabrera Moya
244fdcc69a audiobasesink: use the same type as the internal type to return it 
https://bugzilla.gnome.org/show_bug.cgi?id=687466
 2012-11-02 19:52:38 +00:00
Sebastian Dröge
1813701ef2 audiobasesink: Add explanation to the GAP event handling code 2012-10-24 11:22:29 +02:00
Sebastian Dröge
b793d0bfae audiobasesink: Properly handle GAP events 
These are now converted into silence buffers if they have
a duration or cause the ringbuffer and clock to be started
if they don't have a duration.

Fixes bug #685273.
 2012-10-24 11:19:05 +02:00
Wim Taymans
a57198a0ba audio: improve property description 
Improve the description of the latency-time and buffer-time properties in the
audio sink and source.
 2012-09-14 16:08:50 +02:00
Wim Taymans
668ce33384 update for basesink change 2012-09-04 12:18:11 +02:00
Edward Hervey
def07410ef audiobasesink: Avoid resetting ringbuffer when not needed 
If the ringbuffer was configured to the same caps as previously, we
don't need to reconfigure it.
 2012-08-14 18:56:00 +02:00
Edward Hervey
2817bdadc9 libs: Remove "Since" markers and minor doc fixups 2012-07-13 12:11:06 +02:00
Edward Hervey
c9428c96b1 baseaudiosink: Resync when ringbuffer resets 
When the ringbuffer gets restarted (like in setcaps), we *will* have
to resync against the new values.

Without this we end up blindly assuming the new samples align to the
old ones.
 2012-07-12 09:51:35 +02:00
Wim Taymans
c003efcc63 audiobasesink: fix for basesink API change 2012-06-18 11:40:36 +02:00
Wim Taymans
dfb8e7cb2c don't pass random pointers to pull_range 2012-03-16 21:46:47 +01:00
Wim Taymans
25137962ad fix for caps API changes 2012-03-11 19:04:41 +01:00
Wim Taymans
7296ef7c63 audiobasesink: add some G_LIKELY 2012-03-09 17:15:38 +01:00
Wim Taymans
a75e9102c5 GST_FLOW_WRONG_STATE -> GST_FLOW_FLUSHING 2012-02-08 15:17:49 +01:00
Wim Taymans
fcdc385aa1 port to new map API 2012-01-25 12:30:53 +01:00
Sebastian Dröge
68c0790817 Merge branch 'master' into 0.11 
Conflicts:
	gst-libs/gst/interfaces/propertyprobe.c
	sys/xvimage/xvimagesink.c
 2012-01-25 11:50:54 +01:00
Wim Taymans
3d42f0f6ed port to new glib thread API 2012-01-19 11:36:17 +01:00
Sebastian Dröge
5cb3d75dbf audiobasesink: Fix infinite recursion by chaining up to the correct parent class vfunc 2012-01-09 14:19:54 +01:00
Edward Hervey
f562a29284 Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/theora/gsttheoraenc.c
	gst-libs/gst/tag/gstexiftag.c
	gst/adder/gstadder.c
	gst/adder/gstadder.h
	gst/playback/gstdecodebin2.c
	gst/playback/gstsubtitleoverlay.c
	tests/check/libs/tag.c
 2011-12-30 13:21:35 +01:00
Tim-Philipp Müller
fb6d09055a Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	ext/alsa/gstalsadeviceprobe.c
	ext/alsa/gstalsamixer.c
	ext/pango/gsttextoverlay.c
	ext/pango/gsttextoverlay.h
	gst-libs/gst/audio/gstaudiobasesink.c
	gst-libs/gst/audio/gstaudioringbuffer.c
	gst-libs/gst/audio/gstaudiosrc.c
	gst-libs/gst/video/Makefile.am
	gst-libs/gst/video/video.c
	gst/encoding/gststreamcombiner.c
	gst/encoding/gststreamsplitter.c
	gst/playback/gstplaybasebin.c
	gst/playback/gststreamsynchronizer.c
	gst/playback/gstsubtitleoverlay.c
	gst/playback/gsturidecodebin.c
	sys/xvimage/xvimagesink.c
	tests/examples/Makefile.am
	win32/common/libgstvideo.def

Video overlay composition disabled for now, needs
porting to buffer meta.
 2011-12-08 01:19:03 +00:00
Wim Taymans
f096b8a8d8 ringbuffer: remove old _full version 2011-12-06 15:06:12 +01:00
Wim Taymans
1225aa9a78 update for basesink event handler changes 2011-12-02 22:24:43 +01:00
Tim-Philipp Müller
177525f89f Merge remote-tracking branch 'origin/master' into 0.11 
Conflicts:
	gst-libs/gst/netbuffer/gstnetbuffer.c
	gst/ffmpegcolorspace/avcodec.h
	gst/ffmpegcolorspace/gstffmpegcodecmap.c
	gst/ffmpegcolorspace/imgconvert.c
	gst/ffmpegcolorspace/imgconvert_template.h
	gst/ffmpegcolorspace/mem.c
	gst/playback/README
	gst/playback/gstplaybasebin.c
	gst/playback/gstplaybasebin.h
	gst/playback/gstplaybin.c
	sys/v4l/v4lmjpegsrc_calls.c
	sys/v4l/videodev_mjpeg.h
	tests/check/elements/gnomevfssink.c
 2011-12-02 11:10:17 +00:00
Wim Taymans
59113af604 Use the new GstSample for snapshots 
Make appsink return a GstSample. Remove the pull_buffer_list method because it
is not very useful anymore.
Pass GstSample to the conversion function.
Update playbin2 and examples
 2011-12-01 16:53:11 +01:00
Wim Taymans
468d1dde89 audio: update for clock provider API change 2011-11-28 17:51:41 +01:00
Wim Taymans
285702a1a6 fix for scheduling mode rename 2011-11-18 12:37:10 +01:00
Wim Taymans
a3416bc11f rename baseaudio* -> audiobase* 2011-11-11 12:00:52 +01:00
Renamed from gst-libs/gst/audio/gstbaseaudiosink.c (Browse further)