Commit graph

2328 commits

Author SHA1 Message Date
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
Reynaldo H. Verdejo Pinochet
7559fb29a4 Add missing default include paths to androgenizer call
Fixes building tag/ with Android's NDK
2011-11-03 21:35:38 -03:00
Mersad Jelacic
d430eb65c5 audiosink: avoid deadlocking audioringbuffer thread
... when it goes into wait for ringbuffer starting just after such
having been signalled.

Fixes #661738.
2011-10-28 14:07:40 +02:00
Mark Nauwelaerts
981070eb44 audiodecoder: having gather queue contents implies some draining is in order
... which ensures e.g. processing and sending last fragment of reverse playback
downstream at EOS.
2011-10-19 16:51:09 +02:00
Tim-Philipp Müller
4e59e63ff7 baseaudiosink: fix unused variable compiler warning if debugging in core is disabled
https://bugzilla.gnome.org/show_bug.cgi?id=660150
2011-10-19 00:32:13 +01:00
Alessandro Decina
bc6f00becb audioencoder: fix compile warning 2011-10-09 16:48:18 +02:00
Mark Nauwelaerts
871b1584c9 audioencoder: only resync to upstream upon discont in perfect ts mode
... as documented, where discont is marked here if tolerance has been
exceeded.
2011-10-08 20:20:10 +02:00
Mark Nauwelaerts
a7ce550d04 audiodecoder: fix timestamp tolerance handling 2011-10-08 20:20:06 +02:00
Mark Nauwelaerts
d8312994aa audiodecoder: handle empty input by discarding 2011-10-08 20:20:03 +02:00
Mark Nauwelaerts
37c629fcc6 audioencoder: make upstream queries MT-safe 2011-10-07 14:52:50 +02:00
Mark Nauwelaerts
77069f01b1 audiodecoder: make upstream queries and events MT-safe 2011-10-07 14:52:48 +02:00
Vincent Penquerc'h
93900d47ed encoding-profile: add a function to create a profile from a discoverer info
Only A/V streams are added at the moment, there does not seem to be
a similar way to add other streams (eg, subtitles).

https://bugzilla.gnome.org/show_bug.cgi?id=642878
2011-10-03 11:51:23 +02:00
Mark Nauwelaerts
8633eb391d audiodecoder: really push pending events 2011-09-28 15:42:46 +02:00
Tim-Philipp Müller
e4e2e3c7b0 audioencoder: remove more tags from upstream tag events such as bitrate tags
We want to remove all codec specific tags.
2011-09-28 14:32:20 +01:00
Mark Nauwelaerts
01d27ee084 audioencoder: only got_data if we really got some
... which avoids going loopy with casual subclass.
2011-09-27 16:58:44 +02:00
Mark Nauwelaerts
24d71cf7a6 audioencoder: really push pending events 2011-09-27 16:58:41 +02:00
Mark Nauwelaerts
803b65613b audioencoder: send tag event after pending events
... which probably includes a pending newsegment event.
2011-09-27 16:21:55 +02:00
Mark Nauwelaerts
89f6720545 audioencoder: protect pending_events with proper lock 2011-09-27 16:21:45 +02:00
Mark Nauwelaerts
9a9541ff35 audioencoder: clean up some documentation 2011-09-27 16:21:41 +02:00
Sebastian Dröge
e4c895dfaf audioencoder: Improve set_frame_sample_{min,max} documentation 2011-09-26 16:35:55 +02:00
Sebastian Dröge
b767be2f68 audiodecoder: Fix thread safety issues if both pads have different streaming threads 2011-09-26 16:22:00 +02:00
Sebastian Dröge
d0bf465248 audiodecoder: Delay sending of serialized events to finish_frame() 2011-09-26 16:19:42 +02:00
Sebastian Dröge
f3f416004f Revert "audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code"
This reverts commit 11e375486e.

GST_BOILERPLATE() can't define an abstract type and
G_DEFINE_ABSTRACT_TYPE() does not pass the class struct to
the instance_init function and there's no way to get the
class struct of the current type in instance_init().
2011-09-26 16:02:51 +02:00
Sebastian Dröge
4fa9749106 audioencoder: Add support for requesting a minimum and maximum number of samples per frame
This extends the special case of a fixed number of samples per frame
that was supported before already.
2011-09-26 15:59:22 +02:00
Sebastian Dröge
16c3d6b3d5 audioencoder: Fix thread safety issues if both pads have different streaming threads 2011-09-26 15:45:40 +02:00
Sebastian Dröge
61ffd7cb42 audioencoder: Delay sending of serialized events to finish_frame()
This makes sure that the caps are already set before any serialized
events are sent downstream.
2011-09-26 15:42:14 +02:00
Sebastian Dröge
11e375486e audioencoder: Use GST_BOILERPLATE instead of custom GObject boilerplate code 2011-09-26 15:34:54 +02:00
Mark Nauwelaerts
abafb030ac audioencoder: add some tag handling convenience help 2011-09-26 15:15:03 +02:00
Mark Nauwelaerts
a99b313c26 audioencoder: provide CODEC/AUDIO_CODEC handling 2011-09-26 15:10:08 +02:00
Mark Nauwelaerts
aae0312e10 audioencoder: filter AUDIO_CODEC/CODEC tags from passing tag events 2011-09-26 15:10:06 +02:00
Mark Nauwelaerts
001b4a0072 audioencoder: proxy some more optional downstream caps fields to upstream 2011-09-22 15:47:06 +02:00
Mark Nauwelaerts
2a362a95f7 audioencoder: changed is verily the opposite of equal 2011-09-22 15:47:06 +02:00
Mark Nauwelaerts
b420dd54ea audioencoder: prevent crashing when comparing to a freshly inited GstAudioInfo 2011-09-22 15:46:56 +02:00
Mark Nauwelaerts
7fa7de9221 audio: some more accessor macros for GstAudioInfo 2011-09-22 15:45:05 +02:00
Mark Nauwelaerts
b44978befe audiodecoder: fix documentation typo 2011-09-22 15:45:01 +02:00
Age Bosma
043ee22e25 discoverer: Don't use gtk-doc /* < ... > */ style comments for signals
The /*< ... >*/ style is only used for public|protected|private,
signal comments use /* signals */. This prevents the some code
parsers/binding generators to be confused by the comment.
2011-09-19 14:36:00 +02:00
Mark Nauwelaerts
e574f58e71 rtspdefs: add RTCP-Interval header 2011-09-19 11:32:23 +02:00
Tim-Philipp Müller
454c554b11 docs: minor addition to GST_TAG_ID3V2_HEADER_SIZE docs 2011-09-12 19:55:40 +01:00
Tim-Philipp Müller
55182ed841 baseaudiosrc: don't try to fixate "width" field for alaw/mulaw
Fixes warning when trying to fixate e.g. pulsesrc ! audio/x-alaw ! fakesink.
2011-09-10 18:30:55 +01:00
Tim-Philipp Müller
0f38f86182 colorbalance: add some guards to interface methods
https://bugzilla.gnome.org/show_bug.cgi?id=658584
2011-09-09 13:09:43 +01:00
Wim Taymans
dc28bd1b63 audio: rename IS_LE/BE to _IS_LITTLE_ENDIAN/BIG_ENDIAN 2011-09-06 16:27:27 +01:00
Tim-Philipp Müller
36a75bdb71 audio: update internal silent sample defines as well to match 0.11 2011-09-06 15:46:45 +01:00
Tim-Philipp Müller
91d1112360 audio: update audio format enums to match changes in 0.11
And add new audio format info stuff to docs.
2011-09-06 15:36:51 +01:00
Tim-Philipp Müller
9a8a989a22 docs: more docs clean-ups 2011-09-06 10:07:33 +01:00
Tim-Philipp Müller
5e61db25b5 audio: fix GST_AUDIO_FORMAT_INFO_IS_*() macros to return a boolean 2011-09-05 23:28:20 +01:00
Tim-Philipp Müller
ba05716485 docs: some docs love 2011-09-05 23:28:20 +01:00