Commit graph

700 commits

Author SHA1 Message Date
Matej Knopp
817f39608c Fix printf format compiler warnings for OSX / 64bit
https://bugzilla.gnome.org/show_bug.cgi?id=662607
2011-11-22 01:00:59 +00:00
Wim Taymans
8fc2a21775 update for activation changes 2011-11-21 13:35:34 +01:00
Wim Taymans
d0bd5f04c0 update for new scheduling query 2011-11-18 17:58:58 +01:00
Wim Taymans
1ad4d20607 add parent to activate functions 2011-11-18 13:56:04 +01:00
Wim Taymans
285702a1a6 fix for scheduling mode rename 2011-11-18 12:37:10 +01:00
Wim Taymans
7afdff3575 Merge branch 'master' into 0.11
Conflicts:
	gst-libs/gst/audio/gstaudiodecoder.c
2011-11-17 17:07:41 +01:00
Wim Taymans
e302833e65 add parent to pad functions 2011-11-17 12:48:25 +01:00
Mark Nauwelaerts
69c2c46472 audioencoder: invalidate format info when setup negotiation failed
... which ensures nothing subsequently tries to slip past _chain
and into a possibly improperly setup subclass.
2011-11-16 19:03:47 +01:00
Vincent Penquerc'h
f17f918b75 audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
2011-11-16 16:54:03 +00:00
Wim Taymans
2202511e77 add parent to query function 2011-11-16 17:25:17 +01:00
Wim Taymans
28157e6f21 _query_peer_*() -> _peer_query_*() 2011-11-15 18:04:17 +01:00
Wim Taymans
ab9ffa93f5 change getcaps to query
Add sink and src event functions in rtpbasepayload
Add query vmethod to rtpbasepayload.
2011-11-15 18:04:16 +01:00
Vincent Penquerc'h
3e095382a1 audiodecoder: accept dropped buffers before we know the format
This allows flacdec to not emit audio for headers, while allowing
the base audio decoder to keep its timestamps in sync.
2011-11-15 13:29:31 +00:00
Robert Swain
a23dff1fbb audio: Remove some unused variables 2011-11-14 12:49:50 +01:00
Mark Nauwelaerts
38615abdd8 audiodecoder: improve reverse playback
... by doing some more (reverse) timestamp interpolating and
refactoring downstream pushing.

Fixes #661983.
2011-11-14 12:00:06 +01:00
Tim-Philipp Müller
c76e5804b3 Update for GstURIHandler get_protocols() changes 2011-11-13 23:44:23 +00:00
Tim-Philipp Müller
455f337e3d gio, appsrc, appsink, cdaudiosrc: update for GstURIHandler API changes 2011-11-13 18:22:06 +00:00
Tim-Philipp Müller
4b0dce5148 Merge remote-tracking branch 'origin/master' into 0.11
Conflicts:
	gst-libs/gst/audio/Makefile.am
	gst-libs/gst/audio/audio.h
	tests/examples/seek/jsseek.c
	tests/examples/seek/seek.c
	tests/icles/test-colorkey.c
2011-11-13 13:36:29 +00:00
Tim-Philipp Müller
cd21e69913 audio: add GST_AUDIO_INFO_IS_VALID macro and use in audio decoder base class
API: GST_AUDIO_INFO_IS_VALID
2011-11-13 13:18:16 +00:00
Tim-Philipp Müller
394b1f8c3c audio: fix order in LIBADD
Local libs must come first.
2011-11-12 12:13:05 +00:00
Tim-Philipp Müller
756c9e2948 audio: fix order in LIBADD
Local libs must come first.
2011-11-12 11:58:59 +00:00
Tim-Philipp Müller
dfc13ec632 cdda: rename GstCddaBaseSrc to GstAudioCdSrc and move to libgstaudio
Another mini-lib down, to make space for new mini libs.

Remove bogus copyright line while at it.
2011-11-12 11:58:58 +00:00
Wim Taymans
c42e257751 audio: fix docs 2011-11-11 19:13:52 +01:00
Wim Taymans
b645287775 audio: fix headers
Add const to some methods.
Add padding.
Add GType for GstAudioInfo and GstAudioFormatInfo.
Add new/copy/free for GstAudioInfo.
2011-11-11 17:53:03 +01:00
Wim Taymans
a3416bc11f rename baseaudio* -> audiobase* 2011-11-11 12:00:52 +01:00
Wim Taymans
ee7072fe7e rename GstBaseAudio* ->GstAudioBase* 2011-11-11 11:52:47 +01:00
Wim Taymans
3d0ac3ded2 rename files to match contained objects 2011-11-11 11:33:15 +01:00
Wim Taymans
6511f36fdb audio: GstRingBuffer -> GstAudioRingBuffer 2011-11-11 11:21:41 +01:00
Wim Taymans
b81af23992 audio: rename internal audio ringbuffer 2011-11-11 10:54:39 +01:00
Wim Taymans
ad8f694ec6 remove bogus files
They got somehow commited in 7012e88090
2011-11-11 10:39:52 +01:00
Wim Taymans
e338792ab0 update for adapter api changes 2011-11-10 18:32:39 +01:00
Wim Taymans
f8ef57ca48 Merge branch 'master' into 0.11 2011-11-10 17:26:12 +01:00
Vincent Penquerc'h
0d47c615ad baseaudiosink: make unsigned properties unsigned, not signed 2011-11-10 15:55:31 +00:00
Wim Taymans
57eaf388e0 audio: fix base class vmethods 2011-11-10 16:24:12 +01:00
Wim Taymans
ea9bc40bf9 audiosrc: avoid deadlock 2011-11-10 16:05:19 +01:00
Wim Taymans
1f8fe283f6 audioclock: remove _full version 2011-11-10 13:51:23 +01:00
Wim Taymans
d77c8cafee Merge branch 'master' into 0.11
Conflicts:
	common
	ext/pango/gsttextoverlay.c
	gst-libs/gst/video/video.c
2011-11-09 12:11:59 +01:00
Wim Taymans
372b9329b9 remove query types 2011-11-09 11:47:54 +01:00
Tim-Philipp Müller
d7fc45f42e docs: fix up some Since: markers 2011-11-07 23:05:44 +00:00
Wim Taymans
7ac25e9b26 Merge branch 'master' into 0.11
Conflicts:
	common
	configure.ac
	gst-libs/gst/audio/gstbaseaudiosink.c
	gst/playback/gstdecodebin2.c
	gst/playback/gstplaysinkaudioconvert.c
	gst/playback/gstplaysinkaudioconvert.h
	gst/playback/gstplaysinkvideoconvert.c
	gst/playback/gstplaysinkvideoconvert.h
2011-11-07 12:23:15 +01:00
Felipe Contreras
3df415d4c7 baseaudiosink: make discont-wait configurable
Now we can configure how much time to wait before deciding that a
discont has happened.

Also, adds getter and setter to allow derived implementations to set
this value upon construction.

Suggestions and several improvements by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 11:58:46 +01:00
Felipe Contreras
0a111bf26e baseaudiosink: delay the resyncing of timestamp vs ringbuffertime
A common problem for audio-playback is that the timestamps might not
be completely linear. This is specially common when doing streaming over
a network, where you can have jittery and/or bursty packettransmission,
which again will often be reflected on the buffertimestamps.

Now, the current implementation have a threshold that says how far the
buffertimestamp is allowed o drift from the ideal aligned time in the
ringbuffer. This was an instant reaction, and ment that if one buffer
arrived with a timestamp that would breach the drift-tolerance, a resync
would take place, and the result would be an audible gap for the
listener.

The annoying thing would be that in the case of a "timestamp-outlier",
you would first resync one way, say +100ms, and then, if the next
timestamp was "back on track", you would end up resyncing the other way
(-100ms) So in fact, when you had only one buffer with slightly off
timestamping, you would end up with *two* audible gaps. This is the
problem this patch addresses.

The way to "fix" this problem with the previous implementation, would
have been to increase the "drift-tolerance" to a value that was greater
than the largest timestamp-outlier one would normally expect.  The big
problem with this approach, however, is that it will allow normal
operations with a huge offset timestamp vs running-time, which is
detrimental to lip-sync. If the drift-tolerance is set to 200ms, it
basically means that lip-sync can easily end up being off by that much.

This patch will basically start a timer when the first breach of
drift-tolerance is detected. If any following timestamp for the next n
nanoseconds gets "back on track" within the threshold, it has basically
eliminated the effect of an outlier, and the timer is stopped.  If,
however, all timestamps within this time-limit are breaching the
threshold, we are probably facing a more permanent offset in the
timestamps, and a resync is allowed to happen.

So basically this patch offers something as rare as both higher
accuracy, it terms of allowing smaller drift-tolerances, as well as much
smoother, less glitchy playback!

Commit message and improvments by Havard Graff.

Fixes bug #640859.
2011-11-07 11:33:32 +01:00
Felipe Contreras
3f1395afae baseaudiosink: rename some variables 2011-11-07 11:18:34 +01:00
Felipe Contreras
fbde258be6 baseaudiosink: use gst_util_uint64_scale_int when appropriate
It's probably safer this way.
2011-11-07 11:11:08 +01:00
Felipe Contreras
369cf3f14a baseaudiosink: split drift-tolerance into alignment-threshold
So that drift-tolerance is used for clock slaving resync, and
alignment-threshold is for timestamp drift.
2011-11-07 11:10:05 +01:00
Felipe Contreras
58b9818853 baseaudiosink: trivial comment fixes
Some found by Havard Graff.

Signed-off-by: Felipe Contreras <felipe.contreras@gmail.com>
2011-11-07 10:57:56 +01:00
Wim Taymans
2f8292b495 ringbuffer: store bpf in the right variable 2011-11-04 13:21:24 +01:00
Wim Taymans
a5fa136c0b update for tag API removal 2011-11-02 12:11:16 +01:00
Wim Taymans
5bdfd6d899 structure: fix for api update 2011-11-02 09:04:27 +01:00
Tim-Philipp Müller
b52c5819fb Update for pad API changes
GstProbeType, GstProbeReturn and GstActivateMode -> GstPad*
2011-11-01 00:34:28 +00:00