New issue
Advanced search Search tips
Note: Color blocks (like or ) mean that a user may not be available. Tooltip shows the reason.
Starred by 5 users
Status: Started
Owner:
Cc:
Components:
NextAction: ----
OS: ----
Pri: 2
Type: Bug

Blocked on:
issue 3521
issue 5367



Sign in to add a comment
Implement new API for AudioCoding Module
Project Member Reported by henrik.lundin@webrtc.org, Jun 27 2014 Back to list
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
The following revision refers to this bug:
  http://code.google.com/p/webrtc/source/detail?r=7381

------------------------------------------------------------------
r7381 | henrik.lundin@webrtc.org | 2014-10-07T06:37:39.011423Z

Changed paths:
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_unittest.cc&spec=svn7381&r_previous=7380&r=7381&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.cc&spec=svn7381&r_previous=7380&r=7381&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/interface/neteq.h&spec=svn7381&r_previous=7380&r=7381&format=side
   M http://code.google.com/p/webrtc/source/diff?path=/trunk/webrtc/modules/audio_coding/neteq/neteq_impl.h&spec=svn7381&r_previous=7380&r=7381&format=side

Set NetEq playout mode through the Config struct

This change opens up the possibility to set the playout mode when
creating the NetEq object. The old methods SetPlayoutMode and
PlayoutMode are still available, but are deprecated.

BUG=3520
R=turaj@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/23869004
-----------------------------------------------------------------
Project Member Comment 12 by tina.legrand@webrtc.org, Oct 30 2014
Labels: EngTriaged IceBox
Project Member Comment 13 by bugdroid1@chromium.org, Jan 14 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/8315d7de8551963c53162e320835c158088fcdd6

commit 8315d7de8551963c53162e320835c158088fcdd6
Author: henrik.lundin@webrtc.org <henrik.lundin@webrtc.org>
Date: Wed Jan 14 16:07:26 2015

Remove dual stream functionality in VoiceEngine

This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. The corresponding code in ACM will be deleted in a
follow-up CL.

BUG=3520
R=henrika@webrtc.org, pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/32999004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8060 4adac7df-926f-26a2-2b94-8c16560cd09d

[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/talk/media/webrtc/fakewebrtcvoiceengine.h
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/channel.cc
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/channel.h
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/include/voe_codec.h
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/voe_codec_impl.cc
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/voe_codec_impl.h
[modify] http://crrev.com/8315d7de8551963c53162e320835c158088fcdd6/webrtc/voice_engine/voe_codec_unittest.cc

Project Member Comment 14 by bugdroid1@chromium.org, Jan 15 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/1f67b53c888c6305b60fba327c0a422c8884ff53

commit 1f67b53c888c6305b60fba327c0a422c8884ff53
Author: henrik.lundin@webrtc.org <henrik.lundin@webrtc.org>
Date: Thu Jan 15 09:36:30 2015

Remove dual stream functionality in ACM

This is old code that is no longer in use. The clean-up is part of the
ACM redesign work. With this change, there is no longer need for the
ProcessDualStream method, which is removed. Consequently, the method
ProcessSingleStream is renamed to Process.

BUG=3520
R=kwiberg@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/39489004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@8074 4adac7df-926f-26a2-2b94-8c16560cd09d

[modify] http://crrev.com/1f67b53c888c6305b60fba327c0a422c8884ff53/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/1f67b53c888c6305b60fba327c0a422c8884ff53/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/1f67b53c888c6305b60fba327c0a422c8884ff53/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[delete] http://crrev.com/9ce01e641604315fe3b16c922e29c9831a687feb/webrtc/modules/audio_coding/main/test/dual_stream_unittest.cc
[modify] http://crrev.com/1f67b53c888c6305b60fba327c0a422c8884ff53/webrtc/modules/modules.gyp

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 16 by bugdroid1@chromium.org, Feb 25 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/c5558b702193582a731aeb2dad33af85b685b4cf

commit c5558b702193582a731aeb2dad33af85b685b4cf
Author: henrik.lundin@webrtc.org <henrik.lundin@webrtc.org>
Date: Wed Feb 25 10:37:20 2015

Remove AudioCodingModule's dependency on the Module interface

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

Review URL: https://webrtc-codereview.appspot.com/42069004

Cr-Commit-Position: refs/heads/master@{#8500}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8500 4adac7df-926f-26a2-2b94-8c16560cd09d

[modify] http://crrev.com/c5558b702193582a731aeb2dad33af85b685b4cf/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/c5558b702193582a731aeb2dad33af85b685b4cf/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/c5558b702193582a731aeb2dad33af85b685b4cf/webrtc/modules/audio_coding/main/interface/audio_coding_module.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 19 by bugdroid1@chromium.org, Sep 18 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/061b79af6023b6caf9975be39fe53dd0ec3b7464

commit 061b79af6023b6caf9975be39fe53dd0ec3b7464
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Fri Sep 18 08:29:11 2015

ACM: Remove functions related to DTMF

The functions were essentially no-op. Also removing forward declaration
of ACMDTMFDetection, which was not used.

BUG=3520

Review URL: https://codereview.webrtc.org/1356543003

Cr-Commit-Position: refs/heads/master@{#9982}

[modify] http://crrev.com/061b79af6023b6caf9975be39fe53dd0ec3b7464/webrtc/modules/audio_coding/main/acm2/audio_coding_module.cc
[modify] http://crrev.com/061b79af6023b6caf9975be39fe53dd0ec3b7464/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.cc
[modify] http://crrev.com/061b79af6023b6caf9975be39fe53dd0ec3b7464/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.h
[modify] http://crrev.com/061b79af6023b6caf9975be39fe53dd0ec3b7464/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[modify] http://crrev.com/061b79af6023b6caf9975be39fe53dd0ec3b7464/webrtc/voice_engine/channel.cc

Project Member Comment 20 by bugdroid1@chromium.org, Sep 18 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/e510d7f100f716048a216e2786617d1bbd5bb815

commit e510d7f100f716048a216e2786617d1bbd5bb815
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Fri Sep 18 10:56:09 2015

Remove ACM AudioCodingFeedback callback object and derived classes

The callback object was not used anymore. Also removing the deprecated
WEBRTC_DTMF_DETECTION macro from engine_configurations.h.

BUG=3520

Review URL: https://codereview.webrtc.org/1353763002

Cr-Commit-Position: refs/heads/master@{#9988}

[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/engine_configurations.h
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/interface/audio_coding_module.h
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/test/APITest.cc
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/test/APITest.h
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/test/TwoWayCommunication.cc
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/test/utility.cc
[modify] http://crrev.com/e510d7f100f716048a216e2786617d1bbd5bb815/webrtc/modules/audio_coding/main/test/utility.h

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 24 by bugdroid1@chromium.org, Sep 28 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/8387c5f449c6cfcf6fe604489439efdb889e0c05

commit 8387c5f449c6cfcf6fe604489439efdb889e0c05
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Sep 28 16:24:51 2015

Remove AMR format parameter from AudioCoder in utility

The parameter was never used.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1365283002

Cr-Commit-Position: refs/heads/master@{#10095}

[modify] http://crrev.com/8387c5f449c6cfcf6fe604489439efdb889e0c05/webrtc/modules/utility/source/coder.cc
[modify] http://crrev.com/8387c5f449c6cfcf6fe604489439efdb889e0c05/webrtc/modules/utility/source/coder.h
[modify] http://crrev.com/8387c5f449c6cfcf6fe604489439efdb889e0c05/webrtc/modules/utility/source/file_player_impl.cc
[modify] http://crrev.com/8387c5f449c6cfcf6fe604489439efdb889e0c05/webrtc/modules/utility/source/file_recorder_impl.cc

Project Member Comment 25 by bugdroid1@chromium.org, Sep 29 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/fb9e76369d3270ba7f4272012a136adafe25745e

commit fb9e76369d3270ba7f4272012a136adafe25745e
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Tue Sep 29 06:30:22 2015

Remove last use of ACMAMRPackingFormat

It was no-op used in FileRecorder.

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1360173003

Cr-Commit-Position: refs/heads/master@{#10102}

[modify] http://crrev.com/fb9e76369d3270ba7f4272012a136adafe25745e/webrtc/modules/audio_coding/main/interface/audio_coding_module_typedefs.h
[modify] http://crrev.com/fb9e76369d3270ba7f4272012a136adafe25745e/webrtc/modules/utility/interface/file_recorder.h
[modify] http://crrev.com/fb9e76369d3270ba7f4272012a136adafe25745e/webrtc/modules/utility/source/file_recorder_impl.cc
[modify] http://crrev.com/fb9e76369d3270ba7f4272012a136adafe25745e/webrtc/modules/utility/source/file_recorder_impl.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 32 by bugdroid1@chromium.org, Nov 2 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/1d5c9bd800434bd8c86bc071a21cb025cee3d56a

commit 1d5c9bd800434bd8c86bc071a21cb025cee3d56a
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Nov 02 12:46:31 2015

Remove unused method AcmReceiver:RedPayloadType

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1415313007

Cr-Commit-Position: refs/heads/master@{#10481}

[modify] http://crrev.com/1d5c9bd800434bd8c86bc071a21cb025cee3d56a/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/1d5c9bd800434bd8c86bc071a21cb025cee3d56a/webrtc/modules/audio_coding/main/acm2/acm_receiver.h

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

Project Member Comment 35 by 76821325...@developer.gserviceaccount.com, Nov 23 2015
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add

commit 057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add
Author: henrik.lundin <henrik.lundin@webrtc.org>
Date: Mon Nov 23 16:19:52 2015

Add new method AcmReceiver::last_packet_sample_rate_hz()

This change allows us to delete AcmReceiver::last_audio_codec_id().

BUG=webrtc:3520

Review URL: https://codereview.webrtc.org/1467183002

Cr-Commit-Position: refs/heads/master@{#10756}

[modify] http://crrev.com/057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add/webrtc/modules/audio_coding/main/acm2/acm_receiver.cc
[modify] http://crrev.com/057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add/webrtc/modules/audio_coding/main/acm2/acm_receiver.h
[modify] http://crrev.com/057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add/webrtc/modules/audio_coding/main/acm2/acm_receiver_unittest_oldapi.cc
[modify] http://crrev.com/057fb89f83f4ba51d4f0151aa8f8cfa5d5bb0add/webrtc/modules/audio_coding/main/acm2/audio_coding_module_impl.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 37 by bugdroid1@chromium.org, Sep 27 2016
The following revision refers to this bug:
  https://chromium.googlesource.com/external/webrtc.git/+/46a8d18efadc790fbb9573fac4bd9a855bee70ee

commit 46a8d18efadc790fbb9573fac4bd9a855bee70ee
Author: ossu <ossu@webrtc.org>
Date: Tue Sep 27 12:43:33 2016

ACM: Removed the code for InitialDelayManager

It looks to have been unused since the landing of
https://codereview.webrtc.org/1419573013

BUG=webrtc:3520

Review-Url: https://codereview.webrtc.org/2363993002
Cr-Commit-Position: refs/heads/master@{#14397}

[modify] https://crrev.com/46a8d18efadc790fbb9573fac4bd9a855bee70ee/webrtc/modules/BUILD.gn
[modify] https://crrev.com/46a8d18efadc790fbb9573fac4bd9a855bee70ee/webrtc/modules/audio_coding/BUILD.gn
[modify] https://crrev.com/46a8d18efadc790fbb9573fac4bd9a855bee70ee/webrtc/modules/audio_coding/acm2/acm_receiver.h
[delete] https://crrev.com/29a44e351e1e35c29dd5ff650f62cbc4d10b3b1b/webrtc/modules/audio_coding/acm2/initial_delay_manager.cc
[delete] https://crrev.com/29a44e351e1e35c29dd5ff650f62cbc4d10b3b1b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
[delete] https://crrev.com/29a44e351e1e35c29dd5ff650f62cbc4d10b3b1b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
[modify] https://crrev.com/46a8d18efadc790fbb9573fac4bd9a855bee70ee/webrtc/modules/audio_coding/audio_coding.gypi

Project Member Comment 38 by bugdroid1@chromium.org, Sep 29
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

Sign in to add a comment