If we are on API level 21 or higher, it is possible to use a special AudioTrack
constructor that uses AudioAttributes and AudioFormat as input. It allows us to
supersede the notion of stream types for defining the behavior of audio playback,
and to allow certain platforms or routing policies to use this information for more
refined volume or routing decisions.
BUG=b/30018586
Review-Url: https://codereview.webrtc.org/2804013002
Cr-Commit-Position: refs/heads/master@{#17737}
- BUILD file reorganized, unit tests now have dedicated targets.
- "fake_polqa" is a binary producing fake output in the same format of PolqaOem64; the binary is injected for unit tests instead of the actual POLQA tool.
- Minor refactoring to inject the path to the POLQA binary instead of its parent folder.
- Unit tests for the evaluation score workers.
- Unit tests for the ApmModuleSimulator class.
- Unit tests for the test data generators: ReverberationTestDataGenerator added.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2811953002
Cr-Commit-Position: refs/heads/master@{#17674}
This CL adds tuning to AEC3 for the purpose of reducing the impact of
gain changes in the analog microphone gain.
BUG=chromium:710818, webrtc:6018
Review-Url: https://codereview.webrtc.org/2811283003
Cr-Commit-Position: refs/heads/master@{#17673}
This CL adds further SSE2 optimizations for the AEC3
adaptive filter.
The changes are bitexact
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2810133002
Cr-Commit-Position: refs/heads/master@{#17667}
Most of the complex functionality in AEC3 is done using
vector maths. This CL adds a new functionality for
performing these using SIMD operations in a simple manner
whenever such are available.
The reason for putting the implementations in the header file
is to allow any possible inlining.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2813823002
Cr-Commit-Position: refs/heads/master@{#17663}
Reason for revert:
Broke downstream dependencies.
Original issue's description:
> Change NetEq::InsertPacket to take an RTPHeader
>
> It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
> a member. None of the other member in WebRtcRTPHeader where used in
> NetEq.
>
> This CL adapts the production code; tests and tools will be converted
> in a follow-up CL.
>
> BUG=webrtc:7467
>
> Review-Url: https://codereview.webrtc.org/2807273004
> Cr-Commit-Position: refs/heads/master@{#17652}
> Committed: 4d027576a6TBR=ivoc@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2812933002
Cr-Commit-Position: refs/heads/master@{#17657}
The SSE2 optimizations of the filter core in the matched
filter was only half-done. This CL finalizes those.
In particular:
-It adds finalization of updating of the filter.
-It removes the manual loop unrolling in order to reduce and
simplify the code.
Note that the changes pass the bitexactness tests in an
external AEC3 test suite, and the test
MatchedFilter.TestOptimizations succeed.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2813563003
Cr-Commit-Position: refs/heads/master@{#17655}
Reason for revert:
Breaks android buildbots.
Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with appropriate changes to API to not break depending projects.
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2812913002
> Cr-Commit-Position: refs/heads/master@{#17651}
> Committed: 774f6b4b96TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2809653004
Cr-Commit-Position: refs/heads/master@{#17653}
It used to take a WebRtcRTPHeader as input, which has an RTPHeader as
a member. None of the other member in WebRtcRTPHeader where used in
NetEq.
This CL adapts the production code; tests and tools will be converted
in a follow-up CL.
BUG=webrtc:7467
Review-Url: https://codereview.webrtc.org/2807273004
Cr-Commit-Position: refs/heads/master@{#17652}
Reason for revert:
Reland with appropriate changes to API to not break depending projects.
Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2812913002
Cr-Commit-Position: refs/heads/master@{#17651}
Reason for revert:
Relanded by mistake.
Original issue's description:
> Reland of Add content type information to encoded images and corresponding rtp extension header (patchset #1 id:1 of https://codereview.webrtc.org/2816463002/ )
>
> Reason for revert:
> Reland with fixes which break API
>
> Original issue's description:
> > Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
> >
> > Reason for revert:
> > Breaks dependent projects.
> >
> > Original issue's description:
> > > Add content type information to Encoded Images and add corresponding RTP extension header.
> > > Use it to separate UMA e2e delay metric between screenshare from video.
> > > Content type extension is set based on encoder settings and processed and decoders.
> > >
> > > Also,
> > > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> > >
> > > BUG=webrtc:7420
> > >
> > > Review-Url: https://codereview.webrtc.org/2772033002
> > > Cr-Commit-Position: refs/heads/master@{#17640}
> > > Committed: 64e739aeae
> >
> > TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> > # Skipping CQ checks because original CL landed less than 1 days ago.
> > NOPRESUBMIT=true
> > NOTREECHECKS=true
> > NOTRY=true
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2816463002
> > Cr-Commit-Position: refs/heads/master@{#17644}
> > Committed: 5721866808
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2811963002
> Cr-Commit-Position: refs/heads/master@{#17645}
> Committed: 4fa0c4f97fTBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2810923004
Cr-Commit-Position: refs/heads/master@{#17648}
Reason for revert:
Reland with fixes which break API
Original issue's description:
> Revert of Add content type information to encoded images and corresponding rtp extension header (patchset #31 id:600001 of https://codereview.webrtc.org/2772033002/ )
>
> Reason for revert:
> Breaks dependent projects.
>
> Original issue's description:
> > Add content type information to Encoded Images and add corresponding RTP extension header.
> > Use it to separate UMA e2e delay metric between screenshare from video.
> > Content type extension is set based on encoder settings and processed and decoders.
> >
> > Also,
> > Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
> >
> > BUG=webrtc:7420
> >
> > Review-Url: https://codereview.webrtc.org/2772033002
> > Cr-Commit-Position: refs/heads/master@{#17640}
> > Committed: 64e739aeae
>
> TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
> # Skipping CQ checks because original CL landed less than 1 days ago.
> NOPRESUBMIT=true
> NOTREECHECKS=true
> NOTRY=true
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2816463002
> Cr-Commit-Position: refs/heads/master@{#17644}
> Committed: 5721866808TBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2811963002
Cr-Commit-Position: refs/heads/master@{#17645}
Reason for revert:
Breaks dependent projects.
Original issue's description:
> Add content type information to Encoded Images and add corresponding RTP extension header.
> Use it to separate UMA e2e delay metric between screenshare from video.
> Content type extension is set based on encoder settings and processed and decoders.
>
> Also,
> Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
>
> BUG=webrtc:7420
>
> Review-Url: https://codereview.webrtc.org/2772033002
> Cr-Commit-Position: refs/heads/master@{#17640}
> Committed: 64e739aeaeTBR=tommi@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,nisse@webrtc.org,mflodman@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2816463002
Cr-Commit-Position: refs/heads/master@{#17644}
Use it to separate UMA e2e delay metric between screenshare from video.
Content type extension is set based on encoder settings and processed and decoders.
Also,
Fix full-stack-tests to calculate RTT correctly, so new metric could be tested.
BUG=webrtc:7420
Review-Url: https://codereview.webrtc.org/2772033002
Cr-Commit-Position: refs/heads/master@{#17640}
This CL changes the name of classes, methods and variables making using "noise generator".
This naming is replaced with "input-reference generator" which is more descriptive of the actual role.
Comments, CSS class and HTML item names have also been changed.
Consistency for variable names has been verified and the style checked with pylint.
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2805653002
Cr-Commit-Position: refs/heads/master@{#17639}
Changes in the microphone gain are effecting the AEC in the sense
that each change in the microphone gain is a change in the echo
path seen by the AEC. This CL utilizes the ability of AEC3 to
leverage information about known changes in the analog microphone
gain.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2808073002
Cr-Commit-Position: refs/heads/master@{#17625}
During the first few capture frames, there is no way for the AEC
to tell whether there is echo in the capture signal as the echo
removal functionality in the AEC has not yet seen any render
signal. To avoid initial echo bursts due to this, this CL adds
functionality for forcing the echo suppression gain to zero during
the first 50 blocks (200 ms) after call start and after a reported
echo path change.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2808733002
Cr-Commit-Position: refs/heads/master@{#17624}
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: fbcc5cb386TBR=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}
AudioDecoder and AudioDecoderFactory are in webrtc/api/ now, so move
their mocks to someplace central where tests from all over WebRTC are
allowed to #include them.
BUG=webrtc:5805
Review-Url: https://codereview.webrtc.org/2798063004
Cr-Commit-Position: refs/heads/master@{#17619}
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: 292084c376TBR=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}
fixing white spaces
updated authors file
Changed OLA window to use Q14 as Q5 dosnt work with 48khz. 1 ms @ 48 khz is > 2^5
BUG=webrtc:1361
Review-Url: https://codereview.webrtc.org/2763273003
Cr-Commit-Position: refs/heads/master@{#17611}
The unit test ConversationalSpeechTest.MultiEndCallWavReaderAdaptorSine uses CreateSineWavFile() and writes temporary wav files that are used for test (deleted only if the test passes).
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2774423005
Cr-Commit-Position: refs/heads/master@{#17608}
This CL includes extensive tests to match accept or reject decisions on several different timing setups. The setups are simulated using mocks (by far more light-weight than using actual timing and audio track files).
The client code, the unit tests in this case, passes information about the fake audio tracks to MockWavReaderFactory. MockWavReader instances are then created using the parameters defined in the client code. To improve the readability of the tests, generator_unittest.cc includes a docstring explaining how each MultiEndCallSetup* test is documented.
Run tests as follows:
$ out/Default/modules_unittests --gtest_filter=ConversationalSpeechTest.*
BUG=webrtc:7218
Review-Url: https://codereview.webrtc.org/2781573002
Cr-Commit-Position: refs/heads/master@{#17592}
This CL adds support for handling highly reverberant echoes in
AEC3. The functionality is hardcoded to be have no effect (via
a decay factor of 0), but this CL will be followed by other CLs
for which nonzero decay factors will be allowed.
Apart from this change, this CL also refactors the residual
echo estimation code to make it shorter and more readable.
The changes introduced herein are bitexact (for a decay factor
of 0).
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2804223002
Cr-Commit-Position: refs/heads/master@{#17589}
This CL ensures that the number of bands
for the render side matches that for the capture side
when AEC3 is active. Without this, there was problems
when the render rate is different from the capture rate.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2800033003
Cr-Commit-Position: refs/heads/master@{#17586}
Reason for revert:
Trying to re-land after solving some related issues.
There are no changes compared to the original CL.
Original issue's description:
> Revert of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #16 id:300001 of https://codereview.webrtc.org/2747863003/ )
>
> Reason for revert:
> I will try to reland next week because it is causing some problems.
>
> Original issue's description:
> > To accommodate some downstream WebRTC users we need to loosen
> > the coupling between our code and the //third_party/protobuf.
> >
> > This includes using typedefs to define strings instead of
> > assuming std::string.
> >
> > After this refactoring it will be possible to link with other
> > protobuf implementations than the current one.
> >
> > We moved the PRESUBMIT check to another CL [1]. The goal of this
> > presubmit is to avoid the direct usage of google::protobuf outside
> > of the webrtc/base/protobuf_utils.h header file.
> >
> > [1] - https://codereview.webrtc.org/2753823003/
> >
> > BUG=webrtc:7340
> > NOTRY=True
> >
> > Review-Url: https://codereview.webrtc.org/2747863003
> > Cr-Commit-Position: refs/heads/master@{#17466}
> > Committed: 16ab93b952
>
> TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=webrtc:7340
>
> Review-Url: https://codereview.webrtc.org/2786363002
> Cr-Commit-Position: refs/heads/master@{#17483}
> Committed: d00aad5eb2TBR=kjellander@webrtc.org,henrik.lundin@webrtc.org,kwiberg@webrtc.org,michaelt@webrtc.org,peah@webrtc.org
# Not skipping CQ checks because original CL landed more than 1 days ago.
BUG=webrtc:7340
NOTRY=True
Review-Url: https://codereview.webrtc.org/2791963003
Cr-Commit-Position: refs/heads/master@{#17584}
This CL corrects the behavior during buffer underruns.
Furthermore, it increases the tolerance for API call jitter, and
removes the minimum value for the comfort noise.
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2787123003
Cr-Commit-Position: refs/heads/master@{#17576}
This CL adds fairly significant changes to the echo removal
functionality, the main ones being.
-More centralized control over the echo removal.
-Updated echo suppression gain behavior.
-Significantly increased usage of the linear adaptive filter.
-New echo removal functionality when the linear filter is not usable.
This CL is chained to the CL https://codereview.webrtc.org/2784023002/
BUG=webrtc:6018
Review-Url: https://codereview.webrtc.org/2782423003
Cr-Commit-Position: refs/heads/master@{#17575}
Added new callback class that enables routing both the locally recorded
and playout audio data to external module.
A factory method that creates an AudioDeviceModule instance that also
registers the AudioTransportCapture with the device is also added.
BUG=webrtc:7337
R=solenberg@webrtc.org
Review-Url: https://codereview.webrtc.org/2753453002 .
Cr-Commit-Position: refs/heads/master@{#17570}
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}
Because on gcc, cast to void doesn't silence the warning. See
https://gcc.gnu.org/bugzilla/show_bug.cgi?id=66425
Also add an RTC_ prefix to the macro instead of only defining it if it
wasn't already defined, to ensure that we always get our own version.
BUG=none
Review-Url: https://codereview.webrtc.org/2797983003
Cr-Commit-Position: refs/heads/master@{#17563}
