audio_processing: DestroyHandle() now returns void

The return value was not used anyhow and there is no proper action to be taken if we would have received an error. Hence, in line with issue441 we should return void upon free.

BUG=441
TESTED=trybots,modules_unittest
R=andrew@webrtc.org, aluebs@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12269004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5949 4adac7df-926f-26a2-2b94-8c16560cd09d
Reland "Make VoiceEngine choose ACM2 by default""

This cl was originally committed as r5923, but was reverted in r5926
due to a blocking bug (issue 3143). The blocking bug was resolved in
r5936.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12279004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5950 4adac7df-926f-26a2-2b94-8c16560cd09d
Disable tests in common_video_unittests for Dr Memory.

These tests takes about 1000 seconds to execute under
Dr Memory Full, causing them to be killed if taking more
than 1200 seconds.
* TestScaler.BiLinearScaleTest
* TestScaler.BoxScaleTest

BUG=3247
TBR=zhaoqin@google.com

Review URL: https://webrtc-codereview.appspot.com/12289004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5951 4adac7df-926f-26a2-2b94-8c16560cd09d
Create ACM2 instance when calling AudioCodingModule::Create

BUG=2996
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12079005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5952 4adac7df-926f-26a2-2b94-8c16560cd09d
Let A/V sync test use default AudioCoding module

This test used to run with both ACM1 and ACM2, to verify sync with both
versions of the module. ACM1 (and NetEq3) is now being deprecated,
wherefore this test should now use the default one (i.e., ACM2).

BUG=2996
R=stefan@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12299004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5953 4adac7df-926f-26a2-2b94-8c16560cd09d
Stop using ACM factory in VoiceEngine

The factory injection was introduces in order to facilitate switching
between ACM1 and ACM2. Now, ACM1 is being deprecated, and this switching
mechanism is no longer needed.

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12259005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5954 4adac7df-926f-26a2-2b94-8c16560cd09d
Revert "Stop using ACM factory in VoiceEngine"

Some of the bots where breaking.

TBR=henrika@webrtc.org
BUG=2996

Review URL: https://webrtc-codereview.appspot.com/12319004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5955 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix a bug in AcmReceiver::NetworkStatistics

One of the variables were not copied between the structs.

BUG=2996
TBR=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12329004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5956 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove 44.1 kHz workaround from the iOS AudioDevice.

Long, long ago, webrtc didn't support audio at 44.1 kHz. As a result we
treated 44.1 kHz audio as 44 kHz. We now have an arbitrary rate
resampler and have no trouble supporting 44.1 (see 1395 for all the
details). I must have missed updating iOS at the time.

This shouldn't result in a visible change as 16 kHz is selected as the
preferred hardware rate.

BUG=1395
R=fischman@webrtc.org, henrikg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/10949004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5957 4adac7df-926f-26a2-2b94-8c16560cd09d
Reland "Stop using ACM factory in VoiceEngine"

This change was originally landed as r5954, but had to be reverted in
r5955 due to bots failing. The failures should be fixed in r5956,
so the original change is now relanded.

BUG=2996
TBR=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12339004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5958 4adac7df-926f-26a2-2b94-8c16560cd09d
Support arbitrary input/output rates and downmixing in AudioProcessing.

Select "processing" rates based on the input and output sampling rates.
Resample the input streams to those rates, and if necessary to the
output rate.

- Remove deprecated stream format APIs.
- Remove deprecated device sample rate APIs.
- Add a ChannelBuffer class to help manage deinterleaved channels.
- Clean up the splitting filter state.
- Add a unit test which verifies the output against known-working
native format output.

BUG=2894
R=aluebs@webrtc.org, bjornv@webrtc.org, xians@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/9919004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5959 4adac7df-926f-26a2-2b94-8c16560cd09d
Provide GetStats method in RTCPeerConnection

BUG=3144
R=fischman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12069006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5960 4adac7df-926f-26a2-2b94-8c16560cd09d
(Auto)update libjingle 65394435-> 65417850

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5961 4adac7df-926f-26a2-2b94-8c16560cd09d
Restore sample_rate_hz() until Chromium is updated to not use it.

TBR=bjornv
TESTED=Chromium builds against webrtc head.
BUG=2894

Review URL: https://webrtc-codereview.appspot.com/12349004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5962 4adac7df-926f-26a2-2b94-8c16560cd09d
Exclude the new AudioProcessingTest from some sanitizer bots.

It takes too long.

TBR=kjellander

Review URL: https://webrtc-codereview.appspot.com/12359004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5963 4adac7df-926f-26a2-2b94-8c16560cd09d
Remove ACM1/ACM2 switching from VoiceEngine tests

The option to run VoiceEngine tests with both ACM1 and ACM2 was
introduced while the two versions of AudioCoding module where both
in use. Now, ACM1 is being deprecated, and the tests should use the
defualt one (ACM2).

BUG=2996
R=henrika@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12309004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5964 4adac7df-926f-26a2-2b94-8c16560cd09d
AEC: Adds a reported_delay_enabled_ flag

Adds a feature to completely turn on or off buffer handling based on reported delay values. During startup, reported delays are controlled differently through, e.g., WEBRTC_UNTRUSTED_DELAY. By default, the feature is enabled giving the same output as before this change.

TESTED=trybots, modules_unittest
R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/12349005

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5965 4adac7df-926f-26a2-2b94-8c16560cd09d
Fix the return value of DtlsTransportChannelWrapper::SendPacket in the case of invalid RTP packet.

R=juberti@webrtc.org, mallinath@webrtc.org

BUG=3244

Review URL: https://webrtc-codereview.appspot.com/12299006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@5966 4adac7df-926f-26a2-2b94-8c16560cd09d