Commit Graph

724 Commits

Author SHA1 Message Date
ed754e71ae Enable GN check for webrtc/base
It's not possible to enable it for the rtc_base_approved
target but since a larger refactoring is ongoing for webrtc/base
this CL doesn't attempt to fix that.

Changes made:
* Move webrtc/system_wrappers/include/stringize_macros.h into
  webrtc/base:rtc_base_approved_unittests (and corresponding
  unit test to rtc_base_approved_unittests).
* Move md5digest.* from rtc_base_approved to rtc_base_test_utils target.
* Move webrtc/system_wrappers/include/stringize_macros.h (+test) into
  webrtc/base.
* Remove unused use include of webrtc/base/fileutils.h in
  webrtc/base/pathutils.cc

BUG=webrtc:6828, webrtc:3806, webrtc:7480
NOTRY=True

Review-Url: https://codereview.webrtc.org/2717083002
Cr-Commit-Position: refs/heads/master@{#17766}
2017-04-19 15:37:36 +00:00
92aef17cb2 Replace Clock with timeutils in AudioEncoder.
BUG=webrtc:7398

Review-Url: https://codereview.webrtc.org/2782563003
Cr-Commit-Position: refs/heads/master@{#17732}
2017-04-18 07:11:48 +00:00
b4fc73a3ab Removing unnecessary parameters from initializeAndroidGlobals.
The "initialize audio/video" parameters are no longer needed, but
at the same time were required to be true, causing a lot of confusion.
This CL removes them, but leaves the old method signature around,
marked "deprecated".

BUG=webrtc:3416
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2800353002
Cr-Commit-Position: refs/heads/master@{#17626}
2017-04-10 22:08:02 +00:00
8d609f6b6d Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: 292084c376
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: fbcc5cb386

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
2017-04-10 14:39:05 +00:00
fbcc5cb386 Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
Reason for revert:
Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see

https://bugs.chromium.org/p/webrtc/issues/detail?id=7465

Original issue's description:
> Added the GetSources() to the RtpReceiverInterface and implemented
> it for the AudioRtpReceiver.
>
> This method returns a vector of RtpSource(both CSRC source and SSRC
> source) which contains the ID of a source, the timestamp, the source
> type (SSRC or CSRC) and the audio level.
>
> The RtpSource objects are buffered and maintained by the
> RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> the info of the contributing source will be pulled along the object
> chain:
> AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> AudioReceiveStream -> voe::Channel -> RtpRtcp module
>
> Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
>
> BUG=chromium:703122
> TBR=stefan@webrtc.org, danilchap@webrtc.org
>
> Review-Url: https://codereview.webrtc.org/2770233003
> Cr-Commit-Position: refs/heads/master@{#17591}
> Committed: 292084c376

TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=chromium:703122

Review-Url: https://codereview.webrtc.org/2809613002
Cr-Commit-Position: refs/heads/master@{#17616}
2017-04-10 11:38:13 +00:00
292084c376 Added the GetSources() to the RtpReceiverInterface and implemented
it for the AudioRtpReceiver.

This method returns a vector of RtpSource(both CSRC source and SSRC
source) which contains the ID of a source, the timestamp, the source
type (SSRC or CSRC) and the audio level.

The RtpSource objects are buffered and maintained by the
RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
the info of the contributing source will be pulled along the object
chain:
AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
AudioReceiveStream -> voe::Channel -> RtpRtcp module

Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource

BUG=chromium:703122
TBR=stefan@webrtc.org, danilchap@webrtc.org

Review-Url: https://codereview.webrtc.org/2770233003
Cr-Commit-Position: refs/heads/master@{#17591}
2017-04-07 17:57:22 +00:00
1ffbd6c93c Injectable audio encoders: voice_engine/channel changes.
Adds a SetEncoder call to voe::Channel, so that we can move encoder setup outside of Voice Engine.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2703373006
Cr-Commit-Position: refs/heads/master@{#17572}
2017-04-06 19:05:04 +00:00
a1a040a4a4 Injectable audio encoders: BuiltinAudioEncoderFactory
This CL contains all the changes made to audio_coding while making
audio encoders injectable. Apart from some small changes to
webrtcvoiceengine, nothing here is hooked up to the outside
world. Those changes will be added to a follow-up CL.

BUG=webrtc:5806

Review-Url: https://codereview.webrtc.org/2695243005
Cr-Commit-Position: refs/heads/master@{#17569}
2017-04-06 17:03:21 +00:00
cde46b7278 Resolve cyclic dependency between audio network adaptor and event log api
BUG=webrtc:7257

Review-Url: https://codereview.webrtc.org/2745473003
Cr-Commit-Position: refs/heads/master@{#17565}
2017-04-06 12:59:10 +00:00
368f5cf27e Replace use of system_wrappers/include/logging.h by base/logging.h.
BUG=webrtc:5118

Review-Url: https://codereview.webrtc.org/2781343002
Cr-Commit-Position: refs/heads/master@{#17539}
2017-04-05 12:00:33 +00:00
bc436ede07 Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )
Reason for revert:
broke internal project

Original issue's description:
> Supporting 48kHz PCM file.
>
> BUG=None
>
> Review-Url: https://codereview.webrtc.org/2790493004
> Cr-Commit-Position: refs/heads/master@{#17493}
> Committed: 5f93709e7c

TBR=niklas.enbom@webrtc.org,solenberg@webrtc.org,minyue@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=None

Review-Url: https://codereview.webrtc.org/2791453004
Cr-Commit-Position: refs/heads/master@{#17496}
2017-03-31 23:32:28 +00:00
5f93709e7c Supporting 48kHz PCM file.
BUG=None

Review-Url: https://codereview.webrtc.org/2790493004
Cr-Commit-Position: refs/heads/master@{#17493}
2017-03-31 19:27:09 +00:00
fdbfdc9786 Let PacketRouter separate send and receive modules.
This is in preparation for merging the ViERemb logic in packet_router,
to send REMB feedback as sender reports if possible, otherwise as
receiver reports.

BUG=webrtc:7135

Review-Url: https://codereview.webrtc.org/2774623006
Cr-Commit-Position: refs/heads/master@{#17489}
2017-03-31 12:44:52 +00:00
ec6fbd2776 Moves channel-dependent audio input processing to separate encoder task queue.
First approach to remove parts of the heavy load done for encoding, and
preparation for sending, from native audio thread to separate task queue.

With this change we will give the native input audio thread more time to
"relax" between successive audio captures.

Separate profiling done on Android has verified that the change works well;
the load is now redistributed and the load of the native AudioRecordThread
is reduced. Similar conclusions should be valid for all other OS:es as well.

BUG=NONE
CQ_INCLUDE_TRYBOTS=master.tryserver.chromium.android:android_compile_dbg,linux_android_rel_ng

Review-Url: https://codereview.webrtc.org/2665693002
Cr-Commit-Position: refs/heads/master@{#17488}
2017-03-31 12:43:36 +00:00
e6a8009417 Remove voe_auto_test cases for VoEFile.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2786083004
Cr-Commit-Position: refs/heads/master@{#17484}
2017-03-31 11:15:56 +00:00
0ffdcc51bc Delete unneeded includes of deprecated system_wrappers include files.
Deletes left-over includes of trace.h and critical_section_wrapper.h.

BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2784873002
Cr-Commit-Position: refs/heads/master@{#17460}
2017-03-30 07:31:15 +00:00
2877048afe Experiment-driven configuration of PLR/RPLR-based FecController
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2684773002
Cr-Commit-Position: refs/heads/master@{#17419}
2017-03-28 12:03:55 +00:00
4e7645118e Fix UT failure by temporarily uncommenting
BUG=webrtc:7322, webrtc:7405

Review-Url: https://codereview.webrtc.org/2780473002
Cr-Commit-Position: refs/heads/master@{#17393}
2017-03-27 15:53:11 +00:00
d701dfdeef remove more CriticalSectionWrappers.
BUG=webrtc:7035

Review-Url: https://codereview.webrtc.org/2779623002
Cr-Commit-Position: refs/heads/master@{#17392}
2017-03-27 14:24:57 +00:00
1c07c70d88 Reland "WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType"
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2774833003
Cr-Commit-Position: refs/heads/master@{#17391}
2017-03-27 14:15:49 +00:00
b8f9a32459 Define RtpTransportControllerSendInterface.
Implementation owned by call, and passed to VideoSendStream and
AudioSendStream.

BUG=webrtc:6847, webrtc:7135

Review-Url: https://codereview.webrtc.org/2685673003
Cr-Commit-Position: refs/heads/master@{#17389}
2017-03-27 12:36:15 +00:00
670a7f3611 Revert of WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType (patchset #13 id:260001 of https://codereview.webrtc.org/2686043006/ )
Reason for revert:
Makes perf and Chromium FYI bots unhappy.

Original issue's description:
> WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
>
> This removes one more place where we were unable to handle codecs not
> in the built-in set.
>
> BUG=webrtc:5805
>
> Review-Url: https://codereview.webrtc.org/2686043006
> Cr-Commit-Position: refs/heads/master@{#17370}
> Committed: 1724cfbdba

TBR=ossu@webrtc.org,solenberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2772043002
Cr-Commit-Position: refs/heads/master@{#17374}
2017-03-24 12:56:21 +00:00
8ed482e6d7 Remove voe_base_misc_test.cc.
Only one test case in it, testing an API which is deprecated.

BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2757213002
Cr-Commit-Position: refs/heads/master@{#17372}
2017-03-24 10:31:01 +00:00
1724cfbdba WebRtcVoiceMediaChannel::AddRecvStream: Don't call SetRecPayloadType
This removes one more place where we were unable to handle codecs not
in the built-in set.

BUG=webrtc:5805

Review-Url: https://codereview.webrtc.org/2686043006
Cr-Commit-Position: refs/heads/master@{#17370}
2017-03-24 10:16:04 +00:00
dadb4dc3c9 Allow ANA to receive RPLR (recoverable packet loss rate) indications
This is part of a series of CLs. Next CLs:
1. CL for RPLR-based FecController
2. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2661043003
Cr-Commit-Position: refs/heads/master@{#17368}
2017-03-23 22:29:50 +00:00
d12a8e1c8e Attach TransportFeedbackPacketLossTracker to ANA (PLR only)
This CL is one in a series. To finish the work, the following CLs will be added:
1. CL for connecting RPLR as well
2. CL for RPLR-based FecController
3. CL for allowing experiment-driven configuration of the above (through both field-trials and protobuf)

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2638083002
Cr-Commit-Position: refs/heads/master@{#17365}
2017-03-23 18:04:48 +00:00
92e448d529 TransportFeedbackPacketLossTracker to receive std::vector<PacketFeedback> in place of the entire feedback
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2754373002
Cr-Commit-Position: refs/heads/master@{#17322}
2017-03-21 14:31:35 +00:00
6a64bd96bf TransportFeedbackPacketLossTrackerTest cosmetic modification
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2751653006
Cr-Commit-Position: refs/heads/master@{#17320}
2017-03-21 12:58:04 +00:00
0a2391f74c Add thread check to ModuleProcessThread::DeRegisterModule and remove all unnecessary locking that was there due to one implementation calling from a different thread.
The only thing that was holding us back was the indeterministic teardown of voe::Channel(), but it turned out that fixing it wasn't that hard :)

BUG=webrtc:4508

Review-Url: https://codereview.webrtc.org/2755273004
Cr-Commit-Position: refs/heads/master@{#17315}
2017-03-21 09:31:51 +00:00
afcf378459 Remove mixing_test.cc.
BUG=webrtc:4690, webrtc:7238, webrtc:7364

Review-Url: https://codereview.webrtc.org/2756263002
Cr-Commit-Position: refs/heads/master@{#17308}
2017-03-20 12:16:01 +00:00
5b3e49a29e Remove VoEHardware interface.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2753753004
Cr-Commit-Position: refs/heads/master@{#17255}
2017-03-15 15:08:07 +00:00
3f9b12d056 R/PLR calculation - time-based window
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2711473003
Cr-Commit-Position: refs/heads/master@{#17251}
2017-03-15 14:38:13 +00:00
c6192a9e32 Remove VoENetEqStats interface.
(TBR stefan@ for changes to webrtc/test/mock_voice_engine.h)

BUG=webrtc:4690
TBR=stefan@webrtc.org

Review-Url: https://codereview.webrtc.org/2744953003
Cr-Commit-Position: refs/heads/master@{#17201}
2017-03-13 09:36:19 +00:00
fe7dd6d9ff Remove VoEAudioProcessing interface.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2738543002
Cr-Commit-Position: refs/heads/master@{#17185}
2017-03-11 16:10:43 +00:00
67fdb80837 Reland of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #1 id:1 of https://codereview.webrtc.org/2739143002/ )
Reason for revert:
Can reland it if backwards compatible API is kept.

Original issue's description:
> Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
>
> Reason for revert:
> The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.
>
> Original issue's description:
> > Enable cpplint and fix cpplint errors in webrtc/*audio
> >
> > Change usage accordingly throughout the codebase
> >
> > BUG=webrtc:5268
> >
> > TESTED=Fixed issues reported by:
> > find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
> >
> > Review-Url: https://codereview.webrtc.org/2683033004
> > Cr-Commit-Position: refs/heads/master@{#17133}
> > Committed: aebe55ca6c
>
> TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:5268
>
> Review-Url: https://codereview.webrtc.org/2739143002
> Cr-Commit-Position: refs/heads/master@{#17138}
> Committed: e47c1d3ca1

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739073003
Cr-Commit-Position: refs/heads/master@{#17144}
2017-03-09 14:25:06 +00:00
e47c1d3ca1 Revert of Enable cpplint and fix cpplint errors in webrtc/*audio (patchset #4 id:180001 of https://codereview.webrtc.org/2683033004/ )
Reason for revert:
The API change in audio/utility/audio_frame_operations.h caused breakage. Need to keep backward-compatible API.

Original issue's description:
> Enable cpplint and fix cpplint errors in webrtc/*audio
>
> Change usage accordingly throughout the codebase
>
> BUG=webrtc:5268
>
> TESTED=Fixed issues reported by:
> find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py
>
> Review-Url: https://codereview.webrtc.org/2683033004
> Cr-Commit-Position: refs/heads/master@{#17133}
> Committed: aebe55ca6c

TBR=henrika@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:5268

Review-Url: https://codereview.webrtc.org/2739143002
Cr-Commit-Position: refs/heads/master@{#17138}
2017-03-09 10:43:31 +00:00
aebe55ca6c Enable cpplint and fix cpplint errors in webrtc/*audio
Change usage accordingly throughout the codebase

BUG=webrtc:5268

TESTED=Fixed issues reported by:
find webrtc/*audio -type f -name *.cc -o -name *.h | xargs cpplint.py

Review-Url: https://codereview.webrtc.org/2683033004
Cr-Commit-Position: refs/heads/master@{#17133}
2017-03-09 09:05:33 +00:00
8d73f8c6fa Remove VoEVolumeControl interface.
BUG=webrtc:4690, webrtc:6206

Review-Url: https://codereview.webrtc.org/2727063004
Cr-Commit-Position: refs/heads/master@{#17108}
2017-03-08 09:52:20 +00:00
92a7a1810c Update formatting of AudioLevel class
These changes are all no-op, only affecting the appearance of the code. The file names are changed to match the class name.

BUG=none

Review-Url: https://codereview.webrtc.org/2731993002
Cr-Commit-Position: refs/heads/master@{#17087}
2017-03-07 09:58:55 +00:00
759358c9db Remove unused VoiceChannelTransport.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2687843002
Cr-Commit-Position: refs/heads/master@{#17086}
2017-03-07 09:47:57 +00:00
7cb0e55823 Remove voe_cmd_test.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2681993004
Cr-Commit-Position: refs/heads/master@{#17085}
2017-03-07 09:43:21 +00:00
f949000834 Rename webrtc::PacketInfo to webrtc::PacketFeedback. This resolves ambiguity with a similarly named RTCPReceiver::PacketInformation and RtpPacketizerVp9::PacketInfo.
BUG=None

Review-Url: https://codereview.webrtc.org/2710093004
Cr-Commit-Position: refs/heads/master@{#17066}
2017-03-06 13:32:21 +00:00
02455b27a2 Modify TransportFeedbackPacketLossTrackerTest to use parameterization
BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2674733002
Cr-Commit-Position: refs/heads/master@{#17019}
2017-03-03 19:11:06 +00:00
7af9357580 Packet Loss Tracker - Stream Separation
1. Packet reception statistics (PLR and RPLR) calculated on each stream separately.
2. Algorithm changed from considering separate quadrants of the sequence-number-ring to simply looking at the oldest in-window SENT packet.

BUG=webrtc:7058

Review-Url: https://codereview.webrtc.org/2632203002
Cr-Commit-Position: refs/heads/master@{#17018}
2017-03-03 18:51:35 +00:00
dea489f33e Add support for Location (RTC_FROM_HERE) to ProcessThread::RegisterModule.
This makes a few things a lot clearer when looking at perf trace data:

* What module instances (where they were created) are called
* On what thread
* How frequently
* For how long

ProcessThread will be replaced by TaskQueue moving forward and this is a step towards understanding the behavior of the affected code.

BUG=webrtc:7219

Review-Url: https://codereview.webrtc.org/2729053002
Cr-Commit-Position: refs/heads/master@{#16998}
2017-03-03 11:20:24 +00:00
cfa95aa995 Enable GN check in voice_engine/
BUG=webrtc:6828
NOTRY=True

Review-Url: https://codereview.webrtc.org/2714353002
Cr-Commit-Position: refs/heads/master@{#16986}
2017-03-03 05:28:23 +00:00
fb4f8b6cb4 VoE Utility: Fix a naming nit in RemixAndResample
This was pointed out in https://codereview.webrtc.org/2712743004 after
committing.

BUG=webrtc:7220
NOTRY=True
TBR=solenberg@webrtc.org

Review-Url: https://codereview.webrtc.org/2721123005
Cr-Commit-Position: refs/heads/master@{#16965}
2017-03-02 12:10:57 +00:00
796b8f9d71 Remove usage of VoEVolumeControl from WVoE and Audio[Send|Receive]Stream.
BUG=webrtc:4690

Review-Url: https://codereview.webrtc.org/2721003002
Cr-Commit-Position: refs/heads/master@{#16956}
2017-03-02 01:02:23 +00:00
228c268065 Support 4 channel mic in Windows Core Audio
BUG=webrtc:7220

Review-Url: https://codereview.webrtc.org/2712743004
Cr-Commit-Position: refs/heads/master@{#16940}
2017-03-01 13:11:22 +00:00
09f090c40e Remove workaround for bug 6986
BUG=webrtc:6986

Review-Url: https://codereview.webrtc.org/2681733002
Cr-Commit-Position: refs/heads/master@{#16933}
2017-03-01 09:57:11 +00:00