Commit graph

22 commits

Author SHA1 Message Date
Tim-Philipp Müller
01f23d88f3 audiomixer: register function name for debugging just once
Not every time aggregate is called...
2015-11-24 15:17:30 +00:00
Olivier Crête
f0cbeb4140 audioaggregator: Improve log messages
Make the level of log messages saner and improve some.
2015-11-02 19:40:28 -05:00
Vineeth TM
3b89dd4768 audioaggregator: Fix build error
Build error due to wrong argument type in debug message
aagg->priv->offset and next_offset are of type int64, but uint64
formatter is being used in logs. Changing all those to int64

https://bugzilla.gnome.org/show_bug.cgi?id=756065
2015-10-07 11:20:35 +01:00
Sebastian Dröge
40a908b1d9 audioaggregator: Select the initial offset based on the start segment position
instead of always using 0. Otherwise we might output a lot of silence in the
beginning instead of outputting from the relevant position.

https://bugzilla.gnome.org/show_bug.cgi?id=755623
2015-10-01 17:40:59 +02:00
Tim-Philipp Müller
fccee018f3 audiomixer: fix deadlock when G_DISABLE_ASSERT is not defined
This makes the audiomixer unit test time out in master.
Broke with 587e7c4
2015-09-26 10:21:41 +01:00
Sebastian Dröge
bed2c6820f audioaggregator: Stop using deprecated gst_segment_to_position() 2015-09-26 00:17:55 +02:00
Sebastian Dröge
3c44d3eca4 audioaggregator: Only skip the remaining part of a GAP buffer
We might've queued up a GAP buffer that is only partially inside the current
output buffer (i.e. we received it too late!). In that case we should only
skip the part of the GAP buffer that is inside the current output buffer, not
also the remaining part. Otherwise we forward this pad too far into the future
and break synchronization.
2015-09-18 18:00:05 +02:00
Jan Schmidt
587e7c4a23 Don't throw compiler warnings with G_DISABLE_ASSERT
Disable code that warns about unused variables when G_DISABLE_ASSERT
is defined, as it is in tarballs and pre-releases.
2015-09-18 00:29:51 +10:00
Sebastian Dröge
637106e287 audioaggregator: Fix mixup of running times and segment positions
We have to queue buffers based on their running time, not based on
the segment position.

Also return running time from GstAggregator::get_next_time() instead of
a segment position, as required by the API.

Also only update the segment position after we pushed a buffer, otherwise
we're going to push down a segment event with the next position already.

https://bugzilla.gnome.org/show_bug.cgi?id=753196
2015-09-14 19:57:00 +02:00
Sebastian Dröge
97fe89f351 audioaggregator: Use stream time in the position query instead of segment position
https://bugzilla.gnome.org/show_bug.cgi?id=753196
2015-09-14 19:56:51 +02:00
Olivier Crête
3f2bc1e4b2 audioaggregator: On timeout, resync pads with not enough data
https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-07-30 14:00:05 -04:00
Olivier Crête
6efc106a67 aggregator: Queue "latency" buffers at each sink pad.
In the case where you have a source giving the GstAggregator smaller
buffers than it uses, when it reaches a timeout, it will consume the
first buffer, then try to read another buffer for the pad. If the
previous element is not fast enough, it may get the next buffer even
though it may be queued just before. To prevent that race, the easiest
solution is to move the queue inside the GstAggregatorPad itself. It
also means that there is no need for strange code cause by increasing
the min latency without increasing the max latency proportionally.

This also means queuing the synchronized events and possibly acting
on them on the src task.

https://bugzilla.gnome.org/show_bug.cgi?id=745768
2015-07-30 14:00:05 -04:00
Olivier Crête
86fb628d09 audioaggregator: Register function name
Otherwise, it sometimes segfaults with debugging enabled
2015-07-22 19:30:19 -04:00
Olivier Crête
034feb5bb9 audioaggregator: Use 1.0 style buffer allocation 2015-07-22 19:30:12 -04:00
Nirbheek Chauhan
ad8cb458ba audioaggregator: Sync pad values before aggregating
We need to sync the pad values before taking the aggregator and pad locks
otherwise the element will just deadlock if there's any property changes
scheduled using GstController since that involves taking the aggregator and pad
locks.

Also add a test for this.

https://bugzilla.gnome.org/show_bug.cgi?id=749574
2015-07-22 19:50:38 +01:00
Olivier Crête
f8f9c72cc5 audioaggregator: Read output buffer duration with lock held 2015-07-21 21:55:25 -04:00
Luis de Bethencourt
78ce8ff74f audioaggregator: check sink caps are valid 2015-03-24 16:18:22 +00:00
Luis de Bethencourt
cfdcb14730 Revert "audioaggregator: check sink caps are valid"
This reverts commit 6d4d0d1cdf.

Never put code with side effects into an assertion, it can be compiled out
2015-03-24 16:17:00 +00:00
Luis de Bethencourt
6d4d0d1cdf audioaggregator: check sink caps are valid
CID #1291622
2015-03-24 15:53:17 +00:00
Olivier Crête
c565877991 audioaggregator: Print a message when a buffer is late
https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:44:03 -04:00
Olivier Crête
01520c7e47 audioaggregator: Don't re-send the caps if they did not change
https://bugzilla.gnome.org/show_bug.cgi?id=740236
2015-03-16 16:41:45 -04:00
Olivier Crête
959f8e4a3e 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
2015-03-16 16:41:45 -04:00