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Luis de Bethencourt [Tue, 24 Mar 2015 16:17:00 +0000 (16:17 +0000)]
Revert "audioaggregator: check sink caps are valid"
This reverts commit
6d4d0d1cdf7c7531fbf72ce39e1eab1260d20550.
Never put code with side effects into an assertion, it can be compiled out
Luis de Bethencourt [Tue, 24 Mar 2015 15:45:25 +0000 (15:45 +0000)]
audioaggregator: check sink caps are valid
CID #1291622
Olivier Crête [Fri, 14 Nov 2014 01:39:11 +0000 (20:39 -0500)]
audiointerleave: Add unit tests
Almost a copy of the "interleave" unit tests, improved to support
the thread on the src pad on GstAggregator.
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Fri, 6 Mar 2015 18:49:48 +0000 (13:49 -0500)]
audiointerleave: Set src caps in aggregate
This prevents races between the setcaps of the sink pads
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Thu, 13 Nov 2014 20:40:15 +0000 (15:40 -0500)]
audiointerleave: Add interleave element based on audioaggregator
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Fri, 6 Mar 2015 21:51:12 +0000 (16:51 -0500)]
audioaggregator: Print a message when a buffer is late
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Sat, 15 Nov 2014 22:54:51 +0000 (17:54 -0500)]
audioaggregator: Don't re-send the caps if they did not change
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Thu, 6 Nov 2014 22:15:17 +0000 (17:15 -0500)]
audioaggregator: Split base class from audiomixer
Also:
- Don't modify size on early buffer
The size is the size of the buffer, not of remaining part.
- Use the input caps when manipulating the input buffer
Also store in in the sink pad
- Reply to the position query in bytes too
- Put GAP flag on output if all inputs are GAP data
- Only try to clip buffer if the incoming segment is in time or samples
- Use incoming segment with incoming timestamp
Handle non-time segments and NONE timestamps
- Don't reset the position when pushing out new caps
- Make a number of member variables private
- Correctly handle case where no pad has a buffer
If none of the pads have buffers that can be handled, don't claim to be EOS.
- Ensure proper locking
- Only support time segments
https://bugzilla.gnome.org/show_bug.cgi?id=740236
Olivier Crête [Sun, 8 Mar 2015 03:08:40 +0000 (22:08 -0500)]
audiomixer: Release pad object lock before dropping buffer
Otherwise, the locking order is violated and deadlocks happen.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
Olivier Crête [Sat, 7 Mar 2015 01:22:13 +0000 (20:22 -0500)]
audiomixer: Only ignore pads with no buffers on timeout
When the timeout is reached, only ignore pads with no buffers, iterate
over the other pads until all buffers have been read. This is important
in the cases where the input buffers are smaller than the output buffer.
https://bugzilla.gnome.org/show_bug.cgi?id=745768
Olivier Crête [Sat, 7 Mar 2015 01:25:03 +0000 (20:25 -0500)]
audiomixer: Only advance by the buffer size when a buffer is late
https://bugzilla.gnome.org/show_bug.cgi?id=745768
Sebastian Dröge [Thu, 12 Mar 2015 17:11:31 +0000 (17:11 +0000)]
audiomixer: Fix discont detection and buffer alignment code
Actually accumulate the sample counter to check the accumulated error
between actual timestamps and expected ones instead of just resetting
the error back to 0 with every new buffer.
Also don't reset discont_time whenever we don't resync. The whole point of
discont_time is to remember when we first detected a discont until we actually
act on it a bit later if the discont stayed around for discont_wait time.
https://bugzilla.gnome.org/show_bug.cgi?id=746032
Nirbheek Chauhan [Wed, 11 Mar 2015 20:19:58 +0000 (01:49 +0530)]
audiomixer: Add locking to fill_buffer and fix mix_buffer
The audiomixer pad struct fields may be changed from other threads
Nirbheek Chauhan [Wed, 11 Mar 2015 20:26:44 +0000 (01:56 +0530)]
audiomixer: Mark a discont when we receive a new segment event
This allows us to handle new segment events correctly; either by dropping
buffers or inserting silence; for example if the offset is changed on an srcpad
connected to audiomixer.
Sebastian Dröge [Wed, 4 Mar 2015 12:16:03 +0000 (13:16 +0100)]
Revert "audiomixer: Latency is twice the output buffer duration, not only once"
This reverts commit
d387cf67df91b59540f32ee17b4c02f747969add.
The analysis was wrong: The first 20ms of latency are introduced by the source
already and put into the latency query, making it only necessary to cover the
additional 20ms of audiomixer inside audiomixer.
Sebastian Dröge [Tue, 3 Mar 2015 19:03:55 +0000 (20:03 +0100)]
audiomixer: Latency is twice the output buffer duration, not only once
Let's assume a source that outputs outputs 20ms buffers, and audiomixer having
a 20ms output buffer duration. However timestamps don't align perfectly, the
source buffers are offsetted by 5ms.
For our ASCII art picture, each letter is 5ms, each pipe is the start of a
20ms buffer. So what happens is the following:
0 20 40 60
OOOOOOOOOOOOOOOO
| | | |
5 25 45 65
IIIIIIIIIIIIIIII
| | | |
This means that the second output buffer (20 to 40ms) only gets its last 5ms
at time 45ms (the timestamp of the next buffer is the time when the buffer
arrives). But if we only have a latency of 20ms, we would wait until 40ms
to generate the output buffer and miss the last 5ms of the input buffer.
Tim-Philipp Müller [Fri, 13 Feb 2015 16:06:34 +0000 (16:06 +0000)]
audiomixer: use new gst_aggregator_pad_drop_buffer()
Tim-Philipp Müller [Thu, 12 Feb 2015 15:48:07 +0000 (15:48 +0000)]
tests: remove GST_DISABLE_PARSE guards from two tests that don't require it
Tim-Philipp Müller [Thu, 12 Feb 2015 11:26:44 +0000 (11:26 +0000)]
audiomixer: calculate stream_time used to sync pad values correctly
Use pad (input) segment to calculate the stream time from the
input timestamp, not the aggregator (output) segment.
Sebastian Dröge [Wed, 11 Feb 2015 13:16:21 +0000 (14:16 +0100)]
Improve and fix LATENCY query handling
This now follows the design docs everywhere, especially the maximum latency
handling.
https://bugzilla.gnome.org/show_bug.cgi?id=744106
Tim-Philipp Müller [Thu, 5 Feb 2015 15:57:59 +0000 (15:57 +0000)]
audiomixer: remove now-unused base_time field in object structure
Tim-Philipp Müller [Thu, 5 Feb 2015 15:23:04 +0000 (15:23 +0000)]
tests: audiomixer: add unit test for proper segment.base handling
As adjusted by gst_pad_set_offset(), or when doing segment seeks
or looping for example. See previous audiomixer commit.
Sebastian Dröge [Thu, 5 Feb 2015 15:02:54 +0000 (16:02 +0100)]
audiomixer: Remove weird and wrong segment handling
There's no reason why audiomixer should override the segment
base of upstream with whatever value it got from a SEEK event,
or even worse... with 0 if there was no SEEK event yet. This
broke synchronization if upstream provided a segment base other
than 0, e.g. when using pad offsets.
Also that this code did things conditional on the element's state
should've been a big warning already that something is just wrong.
If this breaks anything else now, let's fix it properly :)
Also don't do fancy segment position trickery when receiving a
segment event. It's just not correct.
Thibault Saunier [Mon, 26 Jan 2015 10:29:08 +0000 (11:29 +0100)]
aggregator: Make the PAD_LOCK private
Instead of using the GST_OBJECT_LOCK we should have
a dedicated mutex for the pad as it is also associated
with the mutex on the EVENT_MUTEX on which we wait
in the _chain function of the pad.
The GstAggregatorPad.segment is still protected with the
GST_OBJECT_LOCK.
Remove the gst_aggregator_pad_peak_unlocked method as it does not make
sense anymore with a private lock.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Thibault Saunier [Mon, 26 Jan 2015 10:25:54 +0000 (11:25 +0100)]
aggregator: Hide GstAggregatorPad buffer and EOS fileds
And add a getter for the EOS.
The user should always use the various getters to access
those fields
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Thu, 22 Jan 2015 00:09:13 +0000 (19:09 -0500)]
audiomixer: Make flush start/stop test non-racy
The flush stop could have happened between the source trying
to push the segment event and the buffer, this would cause a warning.
Prevent that by taking the source's stream lock while flushing.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Thu, 22 Jan 2015 22:41:24 +0000 (17:41 -0500)]
audiomixer: Replace racy timeout based tested with drain query
Using the drain query, we can be certain that the buffer has done going
through the aggregator by taking the stream locks.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Thu, 22 Jan 2015 00:32:34 +0000 (19:32 -0500)]
audiomixer: Avoid race in caps negotiation
With the current audiomixer, the input caps need to be the same,
otherwise there is an unavoidable race in the caps negotiation. So
enforce that using capsfilters
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Wed, 21 Jan 2015 23:47:40 +0000 (18:47 -0500)]
audiomixer: Clear GstAudioInfo the the caps
When clearing the caps, also clear the matching GstAudioInfo
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Wed, 21 Jan 2015 23:39:24 +0000 (18:39 -0500)]
audiomixer: Don't reset caps on flush
A flush event doesn't invalidate the previous caps event.
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Olivier Crête [Wed, 14 Jan 2015 19:35:15 +0000 (14:35 -0500)]
aggregator: Replace event lock with pad's object lock
Reduce the number of locks simplify code, what is protects
is exposed, but the lock was not.
Also means adding an _unlocked version of gst_aggregator_pad_steal_buffer().
https://bugzilla.gnome.org/show_bug.cgi?id=742684
Tim-Philipp Müller [Tue, 30 Dec 2014 18:01:34 +0000 (18:01 +0000)]
audiomixer: update for aggregator start/stop vfunc change
Tim-Philipp Müller [Tue, 30 Dec 2014 15:29:36 +0000 (15:29 +0000)]
audiomixer: fix output-block-size property description
Nirbheek Chauhan [Sat, 27 Dec 2014 10:19:39 +0000 (15:49 +0530)]
audiomixer: Document the pad properties
Sebastian Dröge [Tue, 23 Dec 2014 11:24:48 +0000 (12:24 +0100)]
audiomixer: If getting a timeout before having caps, just advance our position
This can happen if this is a live pipeline and no source produced any buffer
and sent no caps until the an output buffer should've been produced according
to the latency.
Sebastian Dröge [Tue, 23 Dec 2014 11:15:50 +0000 (12:15 +0100)]
audiomixer: Make sure to release the current buffer in reset()
If we didn't output the last one in aggregate because we were shutting down
earlier we might otherwise leak it.
Sebastian Dröge [Tue, 23 Dec 2014 10:45:50 +0000 (11:45 +0100)]
audiomixer: Change blocksize property to output-buffer-duration in time format
This makes the interface of audiomixer independent of the actual caps.
Sebastian Dröge [Mon, 22 Dec 2014 21:11:13 +0000 (22:11 +0100)]
audiomixer: Use the src query implementation of aggregator as the default case
Stefan Sauer [Sun, 21 Dec 2014 12:47:25 +0000 (07:47 -0500)]
audiomixer: fix build flag order
Have the libraries/inlcudes from plugins-bad first to avoid picking up the installed version.
Fixes the build when the local api changed.
Sebastian Dröge [Fri, 19 Dec 2014 12:04:58 +0000 (13:04 +0100)]
audiomixer: Track discont-time per pad instead of globally
We do discont handling per pad, not per element!
Sebastian Dröge [Thu, 18 Dec 2014 22:33:58 +0000 (23:33 +0100)]
audiomixer: We're only EOS if all our pads are actually EOS
Having a buffer or not on the pad is irrelevant.
Sebastian Dröge [Thu, 18 Dec 2014 21:42:14 +0000 (22:42 +0100)]
audiomixer: The pad's size is always supposed to be the whole buffer size
And the offset the offset into that buffer. Changing the size will
cause all kinds of assumptions to fail and cause crashes.
Sebastian Dröge [Wed, 17 Dec 2014 18:51:32 +0000 (19:51 +0100)]
aggregator: Add function to allow subclasses to set their own latency
For audiomixer this is one blocksize, for videoaggregator this should
be the duration of one output frame.
Sebastian Dröge [Wed, 17 Dec 2014 18:37:22 +0000 (19:37 +0100)]
audiomixer: Make sure to not have pads being behind the current offset
We would break sync between the different streams then.
Sebastian Dröge [Wed, 17 Dec 2014 16:54:09 +0000 (17:54 +0100)]
aggregator: Add a timeout parameter to ::aggregate()
When this is TRUE, we really have to produce output. This happens
in live mixing mode when we have to output something for the current
time, no matter if we have enough input or not.
Sebastian Dröge [Wed, 17 Dec 2014 17:20:15 +0000 (18:20 +0100)]
audiomixer: Add queues after the (live) sources in the unit test
Sebastian Dröge [Tue, 16 Dec 2014 16:37:12 +0000 (17:37 +0100)]
audiomixer: Implement get_next_time()
Thibault Saunier [Thu, 27 Nov 2014 15:43:39 +0000 (16:43 +0100)]
audiomixer: Do not try to resize a buffer to a negative size on EOS
Tim-Philipp Müller [Sat, 25 Oct 2014 17:50:42 +0000 (18:50 +0100)]
tests: fix audiomixer test on big endian systems
Vineeth T M [Mon, 20 Oct 2014 12:55:08 +0000 (18:25 +0530)]
audiomixer: critical error for blocksize, timeout min/max values
Audiomixer blocksize, cant be 0, hence adjusting the minimum value to 1
timeout value of aggregator is defined with MAX of MAXINT64,
but it cannot cross G_MAXLONG * GST_SECOND - 1
Hence changed the max value of the same
https://bugzilla.gnome.org/show_bug.cgi?id=738845
Aurélien Zanelli [Thu, 9 Oct 2014 12:51:56 +0000 (14:51 +0200)]
test: use G_GSIZE_FORMAT in audiomixer test
https://bugzilla.gnome.org/show_bug.cgi?id=738227
Matthew Waters [Mon, 6 Oct 2014 10:46:24 +0000 (21:46 +1100)]
aggregator: add latency query handling
Thibault Saunier [Mon, 6 Oct 2014 11:09:00 +0000 (13:09 +0200)]
audiomixer: Handle seek event in READY state
Thibault Saunier [Mon, 21 Jul 2014 09:07:19 +0000 (11:07 +0200)]
audiomixer: Set the sinkpad segments basetime after seeking
Otherwise stream offset and running time comparison will not be
correct, leading to segfaults after seeks
Thibault Saunier [Wed, 28 May 2014 14:29:37 +0000 (16:29 +0200)]
audiomixer: Port to GstAggregator
https://bugzilla.gnome.org/show_bug.cgi?id=737183
Co-Authored by: Mathieu Duponchelle <mathieu.duponchelle@opencreed.com>
Sebastian Dröge [Sat, 3 May 2014 16:16:21 +0000 (18:16 +0200)]
Release 1.3.1
Stefan Sauer [Tue, 11 Mar 2014 21:36:01 +0000 (22:36 +0100)]
audiomixer: set a group-id on the stream-start event
Set a default group-id to fix a warning printed by the sink.
Sebastian Dröge [Thu, 16 Jan 2014 11:18:45 +0000 (12:18 +0100)]
audiomixer: Remove some useless assignments and skip the current part of the buffer if the pad is muted
Sebastian Dröge [Wed, 15 Jan 2014 22:30:28 +0000 (23:30 +0100)]
audiomixer: Fix and simplify overlap calculation
Sebastian Dröge [Thu, 19 Dec 2013 20:59:09 +0000 (21:59 +0100)]
audiomixer: Also resync timestamps on the RESYNC flag
Sebastian Dröge [Thu, 7 Nov 2013 13:12:17 +0000 (14:12 +0100)]
audiomixer: Add unit test for handling unaligned buffers
That is, buffers that start or end in the middle of an output block.
Sebastian Dröge [Thu, 7 Nov 2013 13:12:00 +0000 (14:12 +0100)]
audiomixer: Fix EOS handling if we have some pending data
Sebastian Dröge [Thu, 7 Nov 2013 12:57:35 +0000 (13:57 +0100)]
audiomixer: Add test for the discont handling
Sebastian Dröge [Thu, 7 Nov 2013 12:55:32 +0000 (13:55 +0100)]
audiomixer: Refactor sync test to be more reusable
Sebastian Dröge [Thu, 7 Nov 2013 12:48:06 +0000 (13:48 +0100)]
audiomixer: Remove some racy-ness from the unit test
We might already be EOS when we start the mainloop, so only
set the pipeline to PLAYING from the main loop.
Sebastian Dröge [Wed, 6 Nov 2013 14:50:08 +0000 (15:50 +0100)]
audiomixer: Don't take channel mask in consideration in mono or stereo
This could cause negotiation to fail.
https://bugzilla.gnome.org/show_bug.cgi?id=708633
Sebastian Dröge [Wed, 6 Nov 2013 14:18:58 +0000 (15:18 +0100)]
audiomixer: Add simply synchronization test
Sebastian Dröge [Wed, 6 Nov 2013 14:18:50 +0000 (15:18 +0100)]
audiomixer: Add new element based on adder that does synchronized audio mixing