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Starred by 8 users

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Status: Started
Owner:
Cc:
Components:
NextAction: ----
OS: ----
Pri: 2
Type: Bug

Blocked on:
issue 3521
issue 5367



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Implement new API for AudioCoding Module

Project Member Reported by henrik.lundin@webrtc.org, Jun 27 2014

Issue description

This is part of the ACM redesign work.
 
Project Member

Comment 1 by henrik.lundin@webrtc.org, Jun 27 2014

Blockedon: webrtc:3521
Project Member

Comment 2 by henrik.lundin@webrtc.org, Aug 27 2014

Labels: -Pri-2 Pri-1
Status: Started
Project Member

Comment 3 by bugdroid1@chromium.org, Sep 5 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7085

------------------------------------------------------------------
r7085 | henrik.lundin@webrtc.org | 2014-09-05T13:16:23.906230Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h&spec=svn7085&r_previous=7084&r=7085&format=side

Create a new interface for AudioCodingModule

This is a first draft of the interface, and is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17299004
-----------------------------------------------------------------
Project Member

Comment 4 by bugdroid1@chromium.org, Sep 8 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7103

------------------------------------------------------------------
r7103 | henrik.lundin@webrtc.org | 2014-09-08T13:13:19.412719Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc&spec=svn7103&r_previous=7102&r=7103&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc&spec=svn7103&r_previous=7102&r=7103&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h&spec=svn7103&r_previous=7102&r=7103&format=side

Starting to implement the new ACM API

The new implementation class is called AudioCodingImpl, and will in the
end replace AudioCodingModuleImpl.

This is work in progress.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/17359004
-----------------------------------------------------------------
Project Member

Comment 5 by bugdroid1@chromium.org, Sep 22 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7258

------------------------------------------------------------------
r7258 | henrik.lundin@webrtc.org | 2014-09-22T12:07:12.053469Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h&spec=svn7258&r_previous=7257&r=7258&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi&spec=svn7258&r_previous=7257&r=7258&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc&spec=svn7258&r_previous=7257&r=7258&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc&spec=svn7258&r_previous=7257&r=7258&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc&spec=svn7258&r_previous=7257&r=7258&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7258&r_previous=7257&r=7258&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc&spec=svn7258&r_previous=7257&r=7258&format=side

Converting five tests to use new AudioCoding interface

The converted tests are:
AcmIsacMtTest
AcmReceiverBitExactness
AcmSenderBitExactness
AudioCodingModuleMtTest
AudioCodingModuleTest

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old tests were copied and given the
suffix OldApi:
AcmIsacMtTestOldApi
AcmReceiverBitExactnessOldApi
AcmSenderBitExactnessOldApi
AudioCodingModuleMtTestOldApi
AudioCodingModuleTestOldApi

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23719004
-----------------------------------------------------------------
Project Member

Comment 6 by bugdroid1@chromium.org, Sep 22 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7259

------------------------------------------------------------------
r7259 | henrik.lundin@webrtc.org | 2014-09-22T12:10:44.272243Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7259&r_previous=7258&r=7259&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h&spec=svn7259&r_previous=7258&r=7259&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc&spec=svn7259&r_previous=7258&r=7259&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc&spec=svn7259&r_previous=7258&r=7259&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h&spec=svn7259&r_previous=7258&r=7259&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc&spec=svn7259&r_previous=7258&r=7259&format=side

Convert AcmReceiverTest to new AudioCoding interface

In order to maintain test coverage for the old API (AudioCodingModule)
during the transition period, the old test was copied to
AcmReceiverTestOldApi.

Modified and extended AudioCoding and the implementation to make the
test compile and run.

Created a converter method from new to old config struct

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/31409004
-----------------------------------------------------------------
Project Member

Comment 7 by bugdroid1@chromium.org, Sep 22 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7260

------------------------------------------------------------------
r7260 | andresp@webrtc.org | 2014-09-22T13:18:34.929151Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7260&r_previous=7259&r=7260&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h&spec=svn7260&r_previous=7259&r=7260&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc&spec=svn7260&r_previous=7259&r=7260&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc&spec=svn7260&r_previous=7259&r=7260&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h&spec=svn7260&r_previous=7259&r=7260&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc&spec=svn7260&r_previous=7259&r=7260&format=side

Revert "Convert AcmReceiverTest to new AudioCoding interface" (rev 7258).

Breaks android modules_unittests tests by crashing on AcmReceiverBitExactness.8kHzOutput
Was already visible on "git cl try" before submitting on https://webrtc-codereview.appspot.com/23719004/#

BUG=3520
R=kwiberg@webrtc.org, henrik.lundin@webrtc.org
TBR=kwiberg@webrtc.org, henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/25629004
-----------------------------------------------------------------
Project Member

Comment 8 by bugdroid1@chromium.org, Sep 22 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7264

------------------------------------------------------------------
r7264 | andresp@webrtc.org | 2014-09-22T15:49:56.352921Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7264&r_previous=7263&r=7264&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h&spec=svn7264&r_previous=7263&r=7264&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi&spec=svn7264&r_previous=7263&r=7264&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc&spec=svn7264&r_previous=7263&r=7264&format=side
   D http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h&spec=svn7264&r_previous=7263&r=7264&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h&spec=svn7264&r_previous=7263&r=7264&format=side

Revert "Converting five tests to use new AudioCoding interface" (rev 7258).

This time reverts the Cl that actually broke the tests. Got the wrong rev before. :/

BUG=3520
TESTED=Locally with CHECKOUT_SOURCE_ROOT=`pwd` build/android/test_runner.py gtest -s modules_unittests --gtest_filter=AcmReceiverBitExactness.8kHzOutput --verbose --isolate-file-path=webrtc/modules/modules_unittests.isolate
TBR=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/26579004
-----------------------------------------------------------------
Project Member

Comment 9 by bugdroid1@chromium.org, Sep 23 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7272

------------------------------------------------------------------
r7272 | andresp@webrtc.org | 2014-09-23T11:37:57.920581Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h&spec=svn7272&r_previous=7271&r=7272&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc&spec=svn7272&r_previous=7271&r=7272&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7272&r_previous=7271&r=7272&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/interface/audio_coding_module.h&spec=svn7272&r_previous=7271&r=7272&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc&spec=svn7272&r_previous=7271&r=7272&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc&spec=svn7272&r_previous=7271&r=7272&format=side

Reland (rev 7259) "Convert AcmReceiverTest to new AudioCoding interface"

Was reverted by mistake in 7260. Actual culprit was 7258.

BUG=3520
R=henrik.lundin@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/22719004
-----------------------------------------------------------------
Project Member

Comment 10 by bugdroid1@chromium.org, Sep 23 2014

The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7273

------------------------------------------------------------------
r7273 | henrik.lundin@webrtc.org | 2014-09-23T12:05:34.463963Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/modules.gyp&spec=svn7273&r_previous=7272&r=7273&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test.h&spec=svn7273&r_previous=7272&r=7273&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test_oldapi.h&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module.gypi&spec=svn7273&r_previous=7272&r=7273&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc&spec=svn7273&r_previous=7272&r=7273&format=side
   A http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h&spec=svn7273&r_previous=7272&r=7273&format=side

Reland "Converting five tests to use new AudioCoding interface" (r7258)

This CL reverts r7264. The problem was that iSAC-SWB and iSAC-FB are
not supported on android. These are now disabled.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23739004
-----------------------------------------------------------------
Project Member

Comment 11 by bugdroid1@chromium.org, Oct 7 2014

Project Member

Comment 12 by tina.legrand@webrtc.org, Oct 30 2014

Labels: EngTriaged IceBox
Project Member

Comment 15 by bugdroid1@chromium.org, Feb 25 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/af82f75690fbaceabaea9cd7222a14d05dc390cc

commit af82f75690fbaceabaea9cd7222a14d05dc390cc
Author: henrik.lundin@webrtc.org <henrik.lundin@webrtc.org>
Date: Wed Feb 25 10:33:10 2015

Let Add10MsData method do the encoding work as well

This change essentially makes the Process method a no-op. All it does
now is to return a stored value from the last encoding.

The purpose of this change is to forge the Add... and Process methods
into one and the same.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=minyue@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/38229004

Cr-Commit-Position: refs/heads/master@{#8499}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8499 4adac7df-926f-26a2-2b94-8c16560cd09d

[modify] http://crrev.com/af82f75690fbaceabaea9cd7222a14d05dc390cc/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/af82f75690fbaceabaea9cd7222a14d05dc390cc/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h

Project Member

Comment 17 by bugdroid1@chromium.org, Mar 2 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/f56c1623103e3f72198d6b0f50de9f0585b6f52f

commit f56c1623103e3f72198d6b0f50de9f0585b6f52f
Author: henrik.lundin@webrtc.org <henrik.lundin@webrtc.org>
Date: Mon Mar 02 12:29:30 2015

Remove AudioCodingModule::Process()

An earlier change moved the encoding work from Process to
Add10MsData; process was just a no-op.

BUG=3520
COAUTHOR=kwiberg@webrtc.org
R=henrika@webrtc.org, minyue@webrtc.org, tina.legrand@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/43439004

Cr-Commit-Position: refs/heads/master@{#8553}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8553 4adac7df-926f-26a2-2b94-8c16560cd09d

[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/acm_send_test_oldapi.h
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/APITest.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/EncodeDecodeTest.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/SpatialAudio.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/TestAllCodecs.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/TestRedFec.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/TestStereo.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/TestVADDTX.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/delay_test.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/iSACTest.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/modules/utility/source/coder.cc
[modify] http://crrev.com/f56c1623103e3f72198d6b0f50de9f0585b6f52f/webrtc/voice_engine/channel.cc

Project Member

Comment 18 by henrik.lundin@webrtc.org, Mar 30 2015

Labels: -Pri-1 Pri-2
Project Member

Comment 21 by bugdroid1@chromium.org, Sep 28 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/82d6f2a3f7970665a98b1ee8f10252713d9a4f5e

commit 82d6f2a3f7970665a98b1ee8f10252713d9a4f5e
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Mon Sep 28 08:22:43 2015

ACM: Remove ACMVQMonCallback object

It was never used, and the underlying functionality was removed long
ago.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1365193003 .

Cr-Commit-Position: refs/heads/master@{#10083}

[modify] http://crrev.com/82d6f2a3f7970665a98b1ee8f10252713d9a4f5e/webrtc/modules/audio_coding/main/interface/audio_coding_module.h

Project Member

Comment 22 by bugdroid1@chromium.org, Sep 28 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/1bd0e03ce56ed5384c3377b6da171951ec654706

commit 1bd0e03ce56ed5384c3377b6da171951ec654706
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Sep 28 13:12:17 2015

ACM: Removing runtime APIs related to playout mode

The playout mode in NetEq can still be set through the constructor
configuration.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1362943004

Cr-Commit-Position: refs/heads/master@{#10089}

[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/test/APITest.cc
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/test/APITest.h
[modify] http://crrev.com/1bd0e03ce56ed5384c3377b6da171951ec654706/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc

Project Member

Comment 23 by bugdroid1@chromium.org, Sep 28 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/4b808eee856fd37dfc26aca24de4ee09fe3c2cae

commit 4b808eee856fd37dfc26aca24de4ee09fe3c2cae
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Mon Sep 28 13:45:10 2015

ACM: Remove unused and deprecated types

None of these were used.

BUG=webrtc:3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1364703007 .

Cr-Commit-Position: refs/heads/master@{#10090}

[modify] http://crrev.com/4b808eee856fd37dfc26aca24de4ee09fe3c2cae/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h

Project Member

Comment 26 by bugdroid1@chromium.org, Sep 30 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/6ba8e4a4f2f6c7715f33dca290f046967cad112f

commit 6ba8e4a4f2f6c7715f33dca290f046967cad112f
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Wed Sep 30 08:59:21 2015

ACM: Remove a few local enums that were no longer used

BUG=webrtc:3520
R=kwiberg@webrtc.org

Review URL: https://codereview.webrtc.org/1375863002 .

Cr-Commit-Position: refs/heads/master@{#10114}

[modify] http://crrev.com/6ba8e4a4f2f6c7715f33dca290f046967cad112f/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc

Project Member

Comment 27 by henrik.lundin@webrtc.org, Oct 21 2015

We have decided not to do a switch from old (AudioCodingModule) to new (AudioCoding) API. Instead, we will gradually evolve the old API to meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl classes will be removed (they were just wrapping the old interface at this point). Associated unit tests will be deleted. The unit tests using the old AudioCodingModule interface were tagged *oldapi*; this tag will be removed.
Project Member

Comment 28 by bugdroid1@chromium.org, Oct 26 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/e9eca8f5ae60116a2351634575faf4cd5e338e61

commit e9eca8f5ae60116a2351634575faf4cd5e338e61
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Oct 26 12:26:39 2015

Removing AudioCoding class, a.k.a the new ACM API

We have decided not to do a switch from old (AudioCodingModule) to new
(AudioCoding) API. Instead, we will gradually evolve the old API to
meet the new design goals.

As a consequence of this decision, the AudioCoding and AudioCodingImpl
classes are deleted. Also removing associated unit test sources. No
test coverage is lost with this operation, since the tests for the
"old" API are testing more than the deleted tests did.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415163002

Cr-Commit-Position: refs/heads/master@{#10406}

[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/acm_receive_test.cc
[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/acm_receive_test.h
[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest.cc
[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/acm_send_test.cc
[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/acm_send_test.h
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[delete] http://crrev.com/f054819e257a4f9cbb7fa82ba51dc2335f4359ec/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest.cc
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/audio_coding/main/audio_coding_module.gypi
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[modify] http://crrev.com/e9eca8f5ae60116a2351634575faf4cd5e338e61/webrtc/modules/modules.gyp

Project Member

Comment 29 by bugdroid1@chromium.org, Oct 29 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/48ed930975ef7e84023044ed584c4eff914e6c9a

commit 48ed930975ef7e84023044ed584c4eff914e6c9a
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Thu Oct 29 12:36:24 2015

ACM: Move NACK functionality inside NetEq

Negative acknowledgement (NACK) has up to now been implemented in
ACM. But, since NetEq is in charge of the actual packet buffer, it
makes more sense to have the NACK functionlaity in there.

This CL does the following:
- Move nack.{h,cc} and the unit tests from main/acm2 to neteq.
- Move the NACK related code in ACM into NetEq.
- NACK related functions in AcmReceiver are changed to simple
  forwarding APIs.
- Remove unused members in AcmReceiver.
- Remove unused API functions in NetEq.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1410073006

Cr-Commit-Position: refs/heads/master@{#10448}

[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/BUILD.gn
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/main/audio_coding_module.gypi
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/include/neteq.h
[rename] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/nack.cc
[rename] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/nack.h
[rename] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/nack_unittest.cc
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/neteq.gypi
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/neteq_impl.cc
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/audio_coding/neteq/neteq_impl.h
[modify] http://crrev.com/48ed930975ef7e84023044ed584c4eff914e6c9a/webrtc/modules/modules.gyp

Project Member

Comment 30 by bugdroid1@chromium.org, Nov 1 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/74f0f3551ecd596dc0f83146d218887082528fa8

commit 74f0f3551ecd596dc0f83146d218887082528fa8
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Sun Nov 01 19:43:30 2015

Delete a chain of methods in ViE, VoE and ACM

The end goal is to remove AcmReceiver::SetInitialDelay. This change is
in preparation for that goal. It turns out that
AcmReceiver::SetInitialDelay was only invoked through the following
call chain, where each method in the chain is never referenced from
anywhere else (except from tests in some cases):

ViEChannel::SetReceiverBufferingMode
-> ViESyncModule::SetTargetBufferingDelay
-> VoEVideoSync::SetInitialPlayoutDelay
-> Channel::SetInitialPlayoutDelay
-> AudioCodingModule::SetInitialPlayoutDelay
-> AcmReceiver::SetInitialDelay

The start of the chain, ViEChannel::SetReceiverBufferingMode was never
referenced.

This change deletes all the methods above except
AcmReceiver::SetInitialDelay itself, which will be handled in a
follow-up change.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1421013006

Cr-Commit-Position: refs/heads/master@{#10471}

[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/acm2/audio_coding_module_unittest_oldapi.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/include/audio_coding_module.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/test/delay_test.cc
[delete] http://crrev.com/e502bbe138599a10fe530cb789e3f715a56fd461/webrtc/modules/audio_coding/main/test/initial_delay_unittest.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/audio_coding/main/test/insert_packet_with_timing.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/modules/modules.gyp
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/test/fake_voice_engine.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/video_engine/vie_channel.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/video_engine/vie_channel.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/video_engine/vie_sync_module.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/video_engine/vie_sync_module.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/voice_engine/channel.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/voice_engine/channel.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/voice_engine/include/voe_video_sync.h
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/voice_engine/voe_video_sync_impl.cc
[modify] http://crrev.com/74f0f3551ecd596dc0f83146d218887082528fa8/webrtc/voice_engine/voe_video_sync_impl.h

Project Member

Comment 31 by bugdroid1@chromium.org, Nov 2 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/9bc2667fa6deee5d4162b13a878481640a58cce5

commit 9bc2667fa6deee5d4162b13a878481640a58cce5
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Nov 02 11:25:57 2015

ACM/NetEq: Restructure how post-decode VAD is enabled

This change avoids calling neteq_->EnableVad() and DisableVad from the
AcmReceiver constructor. Instead, the new member
enable_post_decode_vad is added to NetEq's config struct. It is
disabled by defualt, but ACM sets it to enabled. This preserves the
behavior both of NetEq stand-alone (i.e., in tests) and of ACM.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1425133002

Cr-Commit-Position: refs/heads/master@{#10476}

[modify] http://crrev.com/9bc2667fa6deee5d4162b13a878481640a58cce5/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/9bc2667fa6deee5d4162b13a878481640a58cce5/webrtc/modules/audio_coding/main/include/audio_coding_module.h
[modify] http://crrev.com/9bc2667fa6deee5d4162b13a878481640a58cce5/webrtc/modules/audio_coding/neteq/include/neteq.h
[modify] http://crrev.com/9bc2667fa6deee5d4162b13a878481640a58cce5/webrtc/modules/audio_coding/neteq/neteq.cc
[modify] http://crrev.com/9bc2667fa6deee5d4162b13a878481640a58cce5/webrtc/modules/audio_coding/neteq/neteq_impl.cc

Project Member

Comment 33 by bugdroid1@chromium.org, Nov 2 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/678c903eb2253163af15db1c0162ed10698646fc

commit 678c903eb2253163af15db1c0162ed10698646fc
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Nov 02 16:31:23 2015

Delete AcmReceiver::SetInitialDelay

This method is no longer called. With that gone, a number of other
methods and member variables are obsoleted, and removed.

Methods deleted:
AcmReceiver::InsertStreamOfSyncPackets
AcmReceiver::GetNumSyncPacketToInsert()
AcmReceiver::GetSilence, never called

Member variables deleted:
missing_packets_sync_stream_
late_packets_sync_stream_
av_sync_
initial_delay_manager_

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1419573013

Cr-Commit-Position: refs/heads/master@{#10484}

[modify] http://crrev.com/678c903eb2253163af15db1c0162ed10698646fc/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/678c903eb2253163af15db1c0162ed10698646fc/webrtc/modules/audio_coding/main/acm2/acm_receiver.h

Project Member

Comment 34 by 76821325...@developer.gserviceaccount.com, Nov 23 2015

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4

commit d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Nov 23 14:49:25 2015

NetEq: Add new method last_output_sample_rate_hz

This change moves the logics for keeping track of the last ouput
sample rate from AcmReceiver to NetEq, where it fits better. The
getter function AcmReceiver::current_sample_rate_hz() is renamed to
last_output_sample_rate_hz().

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1467163002

Cr-Commit-Position: refs/heads/master@{#10754}

[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/include/neteq.h
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/neteq_impl.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/neteq_impl.h
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
[modify] http://crrev.com/d89814bfd78b4c4ab6bf00e6ca4fd9e9ee6055c4/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc

The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/4cf61dd116288e9f119209c59e07f1d9439d8d05

commit 4cf61dd116288e9f119209c59e07f1d9439d8d05
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Wed Dec 09 14:20:58 2015

NetEq: Add codec name and RTP timestamp rate to DecoderInfo

The new fields are default-populated for built-in decoders, but for
external decoders, the name can now be given when registering the
decoder.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1484343003

Cr-Commit-Position: refs/heads/master@{#10952}

[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/acm_receiver.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/acm_receiver.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/include/audio_coding_module.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/decoder_database.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/decoder_database.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/decoder_database_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/include/neteq.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/mock/mock_decoder_database.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_external_decoder_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_impl.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_impl.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_impl_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_stereo_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/neteq_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/payload_splitter_unittest.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/tools/neteq_external_decoder_test.h
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.cc
[modify] http://crrev.com/4cf61dd116288e9f119209c59e07f1d9439d8d05/webrtc/modules/audio_coding/neteq/tools/neteq_rtpplay.cc

Project Member

Comment 38 by bugdroid1@chromium.org, Sep 29 2017

The following revision refers to this bug:
  https://webrtc.googlesource.com/src.git/+/d4a790fbeaa749f1f3e593661bec6a9481d05dda

commit d4a790fbeaa749f1f3e593661bec6a9481d05dda
Author: Henrik Lundin <henrik.lundin@webrtc.org>
Date: Fri Sep 29 14:23:27 2017

Remove AudioCodingModule::IncomingPayload

This method is no longer in use.

Bug: webrtc:3520
Change-Id: Ie1419fa95e6ef482e9adc1ed7af57af2c3510a65
Reviewed-on: https://webrtc-review.googlesource.com/4667
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20047}
[modify] https://crrev.com/d4a790fbeaa749f1f3e593661bec6a9481d05dda/modules/audio_coding/acm2/audio_coding_module.cc
[modify] https://crrev.com/d4a790fbeaa749f1f3e593661bec6a9481d05dda/modules/audio_coding/include/audio_coding_module.h

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