Replace the two versions with a single DownConvertToCodecFormat. As
mentioned in comments, this could be further consolidated with
RemixAndResample but we should write a full audio converter class in
that case.
Along the way:
- Fix the bug present in Channel::Demultiplex with mono input and a
stereo codec.
- Remove the 32 kHz max from the OnDataAvailable path. This avoids a
48 -> 32 -> 48 conversion when VoE is passed 48 kHz audio; instead we
get a straight pass-through to ACM. The 32 kHz conversion is still
needed in the RecordedDataIsAvailable path until APM natively supports
48 kHz.
- Merge resampler improvements from ACM1 to ACM2. This allows ACM to
handle 44.1 kHz audio passed to VoE and was originally done here:
https://webrtc-codereview.appspot.com/1590004
- Reuse the RemixAndResample unit tests for DownConvertToCodecFormat.
- Remove unused functions from utility.cc.
BUG=3155,3000,b/12867572
TESTED=voe_cmd_test using both the OnDataAvailable and
RecordedDataIsAvailable paths, with a captured audio format of all
combinations of {44.1,48} kHz and {1,2} channels, running through all
codecs, and finally using both ACM1 and ACM2.
R=henrika@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/11019005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5843 4adac7df-926f-26a2-2b94-8c16560cd09d
In the case where a network glitch causes a packet to arrive so late
that the jitter buffer has gone into expand mode, the playout timestamp
could have been increased to a value that is larger than the RTP
timestamp of the late packet when it finally arrives. This causes
the difference to be negative, and would make the value wrap (unsigned).
With this fix, the difference is set to zero when the playout
timestamp is ahead of the incoming RTP timestamp. Further down in the
method, a zero-value will lead to the averaging filter not being updated.
BUG=3080
R=henrika@webrtc.org, tina.legrand@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/10199004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5735 4adac7df-926f-26a2-2b94-8c16560cd09d
- Add ability to VoE to send Absolute Sender Time header extension.
- Refactor handling of RTP header extensions in VoE to work the same as in ViE.
- Add API to enable receiving Absolute Sender Time in VoE.
This is part of the work to include audio packets in bandwidth estimation, for
better accuracy in estimates.
BUG=
TBR=solenberg@webrtc.org,henrikg@webrtc.org,stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/9509004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5654 4adac7df-926f-26a2-2b94-8c16560cd09d
Reasons for removing:
- Feels like a complete hack IMHO.
- Not used by any client.
- Unclear functionality regarding time stamp, marker bit etc.
- Causes several issues in tests due to a bad design which mainly depends on the fact that this API "breaks" an ongoing data/packet flow and it complicates the threading model and creates risks for deadlock and memory corruption. Not worth trying to fix given the very unclear benefit of maintaining the API. Better to remove the API instead.
- We also see lots of TSan races and memcheck errors related to this API.
BUG=2296,2240
R=mflodman@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8819004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5574 4adac7df-926f-26a2-2b94-8c16560cd09d
Voice engine shouldn't really have to manage this. Instead, have AGC
keep track of the last input volume, so that it can avoid getting stuck
under coarsely quantized conditions.
Add a test to verify the behavior.
TESTED=unittests, and observed that AGC didn't get stuck on a MacBook
where this problem can actually occur.
R=bjornv@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8729004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5571 4adac7df-926f-26a2-2b94-8c16560cd09d
These high level tests were disabled over time. Since they depend on
real-time results and the filesystem, they tended to be flaky on the
bots. We now give it a very generous 1 second to start up all channels
before verification and a further relaxed file length check. If we
continue to see problems, I will up the startup delay.
The restored tests would have caught the AGC bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5454
Add a new "real audio" stress test to exercise more code paths. This
would have caught the refactor bug fixed here:
https://code.google.com/p/webrtc/source/detail?r=5437
BUG=2164,2844
TESTED=git try. Verified it would have caught the aforementioned bugs
by reintroducing them.
R=andresp@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/8009004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5522 4adac7df-926f-26a2-2b94-8c16560cd09d
This will allow an embedder to use it directly.
Adding inertia/hangover time between updates of the reported detection status to the algorithm, controlled by a parameter. That is usually desired and this way a consumer of
the class don't have to implement that. (VoiceEngine will let it be 1, which results in the same behavior as before, and keep controlling the hangover itself.)
R=andrew@webrtc.org, niklas.enbom@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6219004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5462 4adac7df-926f-26a2-2b94-8c16560cd09d
Add a flag --log_file which produces the existing behaviour of dumping
logs of all severities to a file. By default, warnings and errors will
now be output to stderr. This is generally more useful for the testing
done with voe_cmd_test.
TESTED=logs output to stderr by default and to the usual file when the
flag is specified.
R=tnakamura@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6849005
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5409 4adac7df-926f-26a2-2b94-8c16560cd09d
- Remove one unneeded lock in CaptureLevel(), as the call to this
method should always come on the same thread as PrepareDemux().
- Remove check on analog AGC before doing volume calculations. Saves a
bit of code. Instead check if the incoming volume is set to zero, which
is a potentially common occurrence as it indicates no volume is
available.
R=aluebs@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/6859004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5366 4adac7df-926f-26a2-2b94-8c16560cd09d
Instead have ProcessStream transparently handle changes to the stream
audio parameters (sample rate and channels). This removes two locks
per 10 ms ProcessStream call taken by VoiceEngine (four total with the
audio level indicator.)
Also, prepare future improvements by having the splitting filter take
a length parameter. This will allow it to work at different sample
rates. Remove the useless splitting_filter wrapper.
TESTED=voe_cmd_test with audio processing enabled and switching between
codecs; unit tests.
R=aluebs@webrtc.org, bjornv@webrtc.org, turaj@webrtc.org, xians@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/3949004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@5346 4adac7df-926f-26a2-2b94-8c16560cd09d
