Commit Graph

277 Commits

Author SHA1 Message Date
5e227abfe9 Move under enable_google_benchmarks targets that rely on the benchmarks
Some targets depends on targets under enable_google_benchmarks. But they
are not under such if statement themeself.

Bug: webrtc:12404
Change-Id: I7c0b9a75bd3fa18090ef6a44fda22ed5f33d79b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/204063
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33104}
2021-01-29 15:45:19 +00:00
76bbc98d72 Adding MockVoipEngine for downstream project's tests
Bug: webrtc:11989
Change-Id: Ie9cfe11a0c2b041457de66c3e3a6cdcd6179e4e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201900
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33093}
2021-01-28 21:06:16 +00:00
5eb43b4777 Prefix HAVE_SCTP macro with WEBRTC_.
Generated automatically with:

  git grep -l "\bHAVE_SCTP\b" | xargs \
    sed -i '' 's/HAVE_SCTP/WEBRTC_HAVE_SCTP/g'

Bug: webrtc:11142
Change-Id: I30e16a40ca7a7e388940191df22b705265b42cb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202251
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33042}
2021-01-20 10:51:07 +00:00
b985748f66 Add WebRTC code freshness version string.
This CL adds a string to the resulting WebRTC library (trying to make
sure the version string will be there no matter how WebRTC is packaged).

This CL should be followed by some process to regularly and
automatically update the version string.

No-Try: True
No-Presubmit: True
Bug: webrtc:12159
Change-Id: I9143aeae2cd54d0d4048c138772888100d7873cb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/191223
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32825}
2020-12-14 16:22:35 +00:00
3d25935127 Rename RoboCaller to CallbackList.
As discussed on a design review, the name RoboCaller is not clear
enough and switching to CallbackList will provide readability benefits.

Bug: webrtc:11943
Change-Id: I010cf0a91b5323e4e9c96b83703be7af1e67439c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190142
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32478}
2020-10-23 15:14:22 +00:00
52fa992ccc Rename CancerStickCastle to RoboCaller.
The name was chosen because just like a real-world robocaller
[https://en.wikipedia.org/wiki/Robocall], webrtc::RoboCaller will
call multiple recipients and give all of them the same message,
without giving them the chance to reply.

Change-Id: Ia95f4543b15b48fa6388a50706e489dfccc19f71
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184621
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32152}
2020-09-21 19:50:39 +00:00
9d77762023 Move SampleStatsCounter to public API
Bug: None
Change-Id: I8956f6febbb1caf71e951d212d57746fe1ec5eb2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184506
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32142}
2020-09-18 17:42:53 +00:00
70026f9d14 Put UntypedFunction in untyped_function.h, not function.h
This will make it easier to find stuff...

Bug: webrtc:11943
Change-Id: I4f1ae80b40b4966cb2d8db36701bbc02ac148df6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184512
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32137}
2020-09-18 09:43:07 +00:00
3e98280633 Add unit tests for cancer stick castle library
- Fix the minor issues with the initial library implementation.
- Add unit tests to cover basic scenarios.

Bug: none
Change-Id: Ibf28b4e20f74792fce2fe11d4780fd375a4ad3a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183343
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32122}
2020-09-16 19:04:29 +00:00
78e9acd967 UntypedFunction: Add unit tests and fix a few issues
Bug: webrtc:11943
Change-Id: I1f3c0495612148546ec399a800f97fe88b439c83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/184260
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32116}
2020-09-16 13:30:01 +00:00
f264e70a47 Expand is_linux to is_linux || is_chromeos.
Currently is_linux is set to true on Chrome OS build,
but it is planned to be set false. This CL is the preparation
to keep the compatibility.

Bug: chromium:1110266
Test: Build locally.
Change-Id: Ic79a202b0b3baeff157955cd03a07556bfb958a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/183860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Hidehiko Abe <hidehiko@chromium.org>
Cr-Commit-Position: refs/heads/master@{#32073}
2020-09-10 17:01:16 +00:00
d381eede92 Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
We can then finally delete the top-level common_types.h, and the
corresponding build target webrtc_common.

Bug: webrtc:7660
Change-Id: I1c1096541477586d90774c7a3405b9d36edec14a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182800
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32044}
2020-09-07 08:37:14 +00:00
cccd55094d Delete unneeded dependencies on deprecated build target webrtc_common
Bug: webrtc:7660
Change-Id: Iad32aad8432fa2c6b3018d511b51943f869fbd11
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/182420
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31986}
2020-08-25 07:33:12 +00:00
72e4321f7f Reland "Support AVX2/FMA intrinsics in Audio Resampler module"
This is a reland of 1ca8d87239f1209031bbc77a6443bc7ac2dcee8c

Original change's description:
> Support AVX2/FMA intrinsics in Audio Resampler module
>
> From the test result, using AVX2/FMA is 1.60x faster than SSE on atlas.
>
> Bug: webrtc:11663
> Test: common_audio_unittests on atlas and octopus.
> Change-Id: Ibd45ea46aa97d5790a24e5116f741592b95f6416
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176382
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31810}

Bug: webrtc:11663
Change-Id: I92f5832a42c0314853c9fead46425c08e2040dc0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31945}
2020-08-17 10:40:44 +00:00
0c9204c183 Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
This reverts commit 1ca8d87239f1209031bbc77a6443bc7ac2dcee8c.

Reason for revert: breaks downstream project

Original change's description:
> Support AVX2/FMA intrinsics in Audio Resampler module
> 
> From the test result, using AVX2/FMA is 1.60x faster than SSE on atlas.
> 
> Bug: webrtc:11663
> Test: common_audio_unittests on atlas and octopus.
> Change-Id: Ibd45ea46aa97d5790a24e5116f741592b95f6416
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176382
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31810}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,henrik.lundin@webrtc.org,saza@webrtc.org,peah@webrtc.org,mflodman@webrtc.org,zhaoliang.ma@intel.com

Change-Id: I1dad31df446e336dacb29ff637bd66f809376458
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11663
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/180622
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31813}
2020-07-30 17:35:30 +00:00
1ca8d87239 Support AVX2/FMA intrinsics in Audio Resampler module
From the test result, using AVX2/FMA is 1.60x faster than SSE on atlas.

Bug: webrtc:11663
Test: common_audio_unittests on atlas and octopus.
Change-Id: Ibd45ea46aa97d5790a24e5116f741592b95f6416
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176382
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31810}
2020-07-30 11:39:38 +00:00
c7f0dff191 Convert GN libs lists to frameworks
GN recently added support for Apple frameworks to link, rather than
overloading the libs lists. This pulls .frameworks out of the libs
lists, so that GN can stop supporting .frameworks in libs in the
future.

Bug: chromium:1052560
Change-Id: I263230ddd3c468061584423bba9e1f887503bcaa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178601
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sylvain Defresne <sdefresne@chromium.org>
Cr-Commit-Position: refs/heads/master@{#31632}
2020-07-06 10:08:09 +00:00
8e75bd40e0 Mutex: remove Abseil static initializer.
The change adds conditional inclusion of mutex_abseil.h from mutex.h
and conditional referencing of
//third_party/abseil-cpp/absl/synchronization
which introduces a static initializer.

https://webrtc-review.googlesource.com/c/src/+/176230 introduced a
static initializer which broke the Chromium autoroll,
https://chromium-review.googlesource.com/c/chromium/src/+/2230887.
Example failure:
https://ci.chromium.org/p/chromium/builders/try/android-lollipop-arm-rel/34133

TBR=karl@webrtc.org

No-Try: True
Bug: webrtc:11567
Change-Id: Id78af798f34d5d6beaf9f6e0150e6b3ddd31ff4f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176513
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31451}
2020-06-05 10:26:28 +00:00
f70fbc8411 Introduces rtc_base/synchronization/mutex.h.
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.

The mutex types supportable by webrtc::Mutex are

- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)

In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.

The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.

Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.

Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
2020-06-04 09:55:12 +00:00
c0df5fc25b VoIP API implementation on top of AudioIngress/Egress
This is one last CL that includes the rest of VoIP API implementation.

Bug: webrtc:11251
Change-Id: I3f1b0bf2fd48be864ffc73482105f9514f75f9e0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173860
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31168}
2020-05-05 19:55:29 +00:00
cc73ed3e70 APM: Add build flag to allow building WebRTC without APM
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.

Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
2020-04-26 23:06:44 +00:00
11f92bc81b Audio ingress implementation for voip api.
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.

Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
2020-04-21 20:19:37 +00:00
3c9bcc1f7a Reland of the test portion of:
https://webrtc-review.googlesource.com/c/src/+/172847

------------ original description --------------

Preparation for ReceiveStatisticsProxy lock reduction.

Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: I491e13344b9fa714de0741dd927d907de7e39e83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173583
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31077}
2020-04-15 16:09:44 +00:00
16cc9efd54 Revert "Preparation for ReceiveStatisticsProxy lock reduction."
This reverts commit 24eed2735b2135227bcfefbabf34a89f9a5fec99.

Reason for revert: Speculative revert: breaks downstream project

Original change's description:
> Preparation for ReceiveStatisticsProxy lock reduction.
> 
> Update tests to call VideoReceiveStream::GetStats() in the same or at
> least similar way it gets called in production (construction thread,
> same TQ/thread).
> 
> Mapped out threads and context for ReceiveStatisticsProxy,
> VideoQualityObserver and VideoReceiveStream. Added
> follow-up TODOs for webrtc:11489.
> 
> One functional change in ReceiveStatisticsProxy is that when sender
> side RtcpPacketTypesCounterUpdated calls are made, the counter is
> updated asynchronously since the sender calls the method on a different
> thread than the receiver.
> 
> Make CallClient::SendTask public to allow tests to run tasks in the
> right context. CallClient already does this internally for GetStats.
> 
> Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.
> 
> Bug: webrtc:11489
> Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
> Commit-Queue: Tommi <tommi@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31008}

TBR=mbonadei@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org,tommi@webrtc.org,juberti@webrtc.org,mflodman@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:11489
Change-Id: I48b8359cdb791bf22b1a2c2c43d46263b01e0d65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173082
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31023}
2020-04-07 19:50:20 +00:00
24eed2735b Preparation for ReceiveStatisticsProxy lock reduction.
Update tests to call VideoReceiveStream::GetStats() in the same or at
least similar way it gets called in production (construction thread,
same TQ/thread).

Mapped out threads and context for ReceiveStatisticsProxy,
VideoQualityObserver and VideoReceiveStream. Added
follow-up TODOs for webrtc:11489.

One functional change in ReceiveStatisticsProxy is that when sender
side RtcpPacketTypesCounterUpdated calls are made, the counter is
updated asynchronously since the sender calls the method on a different
thread than the receiver.

Make CallClient::SendTask public to allow tests to run tasks in the
right context. CallClient already does this internally for GetStats.

Remove 10 sec sleep in StopSendingKeyframeRequestsForInactiveStream.

Bug: webrtc:11489
Change-Id: Ib45bfc59d8472e9c5ea556e6ecf38298b8f14921
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172847
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31008}
2020-04-06 14:34:38 +00:00
aa42ecde9a Make transient suppression optionally excludable via defines
This allows clients to exclude the transient suppression submodule from WebRTC builds, by defining WEBRTC_EXCLUDE_TRANSIENT_SUPPRESSOR.

The changes have been shown to be bitexact for a test dataset (when the flag is _not_ defined.)

No-Try: True
Bug: webrtc:11226, webrtc:11292
Change-Id: I6931c82a280a9b40a53ee1c2a9820ed9e674a9a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171421
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30978}
2020-04-02 11:44:07 +00:00
8ab3c77c01 Audio egress implementation for initial voip api in api/voip.
For simplicity and flexibility on audio only API, it deemed
to be better to trim off all audio unrelated logic to serve
the purpose.

Bug: webrtc:11251
Change-Id: I40e3eba2714c171f7c98b158303a7b3f744ceb78
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169462
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30922}
2020-03-27 18:45:43 +00:00
c69fd59270 Fix 'all' build on non Android platforms.
Example of failures when trying to build "all":
https://ci.chromium.org/p/webrtc/builders/ci/Mac%20Asan/23867
https://ci.chromium.org/p/webrtc/builders/try/linux_tsan2/32911
https://ci.chromium.org/p/webrtc/builders/try/linux_ubsan/32390

All related to missing //third_party/ced code.

Bug: webrtc:11411
Change-Id: Ie3d7e87192edfb48d13ab8b14aba05808411a3ea
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170112
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30752}
2020-03-11 08:24:27 +00:00
efbec9a304 [Overuse] Initial version of VideoStreamAdapter (Restrictor moved).
This CL simply moves the VideoSourceRestrictor from being an inner class
of OveruseFrameDetectorResourceAdaptationModule to a new class,
VideoStreamAdapter.

In follow-up CLs, the responsibility of determining what the next step
for adapting up or down should also be moved to the VideoStreamAdapter.

The end-goal is that the VideoStreamAdapter takes care of "can adapt?"
and "do adapt!" type of logic so that a multi-stream aware adaptation
module can decide which stream (adapter) to adapt, and the adapter can
take care of the nitty gritty details of doing so.

In this CL the "can?"/"do!" part is realized but not the logic for
determining what the next step up or down is, and the class interface
needs improvement.

This CL also sets up the video/adaptation/ subdirectory and moves the
AdaptationCounters class here. Other adaptation-related classes (e.g.
the module and its resources) should move into this directory as well
in the future.

Bug: webrtc:11393
Change-Id: I2c12c1281eca854c62791abb65f0aca47a119726
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169542
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30705}
2020-03-06 12:20:01 +00:00
d4c3c3a454 Move video_replay under rtc_tools/.
As pointed out in [1], RTC public tools should live in rtc_tools.

[1] - https://webrtc-review.googlesource.com/c/src/+/168320/2#message-1f40103105ecb077aeec153c5270575138349a50

Bug: chromium:942546
Change-Id: Ic827d9b31ade9a32bf4ef24d020ef8c81d2c9a5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168308
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30486}
2020-02-07 17:57:30 +00:00
83245bde3d Make the dashboard upload script read protos instead of JSON.
I had to pivot and make tests output protos instead of JSON.
I basically move the proto -> JSON conversion into this script instead
of doing it in the test binary.

This is a temporary state. Later it will be enough to just read up
the file and pass it straight to the Catapult implementation, once
it learns to de-serialize the proto directly.

Bug: chromium:1029452
Change-Id: I7ce992eeeb1a5ae0f20eed54174b08b496e74dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166920
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30419}
2020-01-30 10:25:47 +00:00
33aaa35d54 Fix video_replay to build and actually work
Add it to default build target, so it won't get broken accidentally
again. Fix configuration issue with field trials (new parameter was
added recently, but wasn't set by video_replay)

Bug: webrtc:11287
Change-Id: I9c18746d899acd7ac68c1b9b3a646b862c41897a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166900
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30345}
2020-01-22 13:16:28 +00:00
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00
e77f94c54c Remove android_junit_tests from the main BUILD.gn file.
This target has been migrated into two separate targets in
https://webrtc-review.googlesource.com/c/src/+/166603.

Bug: webrtc:11289
Change-Id: Ibdea7616b79695b2ffb67d2210b41db55c41f50b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166536
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30330}
2020-01-21 10:50:40 +00:00
73aa2de3d7 Split android_junit_tests and move targets in the right package.
This is the first step to move //:android_junit_tests to the righ
package (the target is triggering presubmit errors every time //BUILD.gn
gets updated).

Next steps:
* Update recipes
* Remove //:android_junit_tests

Issues with GN formatting, introduced by [1] will be addressed
separately in a "format all" CL.

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11289
No-Presubmit: True
Change-Id: I70c0927d722911f82dd971c30c7ffb581aed69c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166603
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30328}
2020-01-21 08:07:26 +00:00
3e3c551ac6 Suppress C5041 constexpr warning for MSVC 2019
Disable the C5041 warning which makes the build fail. This is a
C++17-only change and WebRTC doesn't support C++17 yet, so the code is
technically correct, but fails to build on MSVC 2019 and
warning-as-error active.

Also fix another warning-as-error build error with MSVC 2019 due to
ignoring the result of a [[nodiscard]] function.

No-Presubmit: True
Bug: webrtc:11275,webrtc:11276
Change-Id: I891a894ee87252f96e84fd8d282576f46907256f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30244}
2020-01-14 07:44:35 +00:00
65bbcabe2f [Android] Replace java_files with sources
Replace all usages of java_files with sources in gn files, and
automatically format.

This is in preparation for java_files being completely removed upstream
in favor of sources.

NOPRESUBMIT=true

Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
2020-01-02 20:08:20 +00:00
b04b2a1719 Initial version of ResourceAdaptationProcessor and friends.
This CL adds Resource, ResourceConsumer, ResourceConsumerConfiguration
and ResourceAdaptationProcessor and implements the algorithm outlined
in
https://docs.google.com/presentation/d/13jyqCWNpIa873iKT6yDuB5Q5ma-c0CvxBpX--0tCclY/edit?usp=sharing.

Simply put, if any resource (such as "CPU") is overusing, the most
expensive consumer (e.g. encoded stream) is adapted one step down.
If all resources are underusing, the least expensive consumer is
adapted one step up.

The current resources, consumers and configurations are all fakes;
this CL has no effect on the current adaptation algorithms used in
practise, but it lays down the foundation for future work in this
area.

Bug: webrtc:11167, webrtc:11168, webrtc:11169
Change-Id: I4054ec7728a52a49e137eee6fa67fa27debd9254
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161237
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30053}
2019-12-10 15:31:43 +00:00
ef3998ffd1 Add directive to make webrtc metrics optional.
Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
2019-12-09 13:55:50 +00:00
cee54179a3 Stop setting -Wextra (the toolchain already does that).
The comment was stale and setting -Wextra actually turns on diagnostics
that are turned off by Chromium.

For example:
"-Wextra -Wno-deprecated-copy -Wextra" will turn on -Wdeprecated-copy
because starting from https://reviews.llvm.org/D70342
-Wdeprecated-copy is part of -Wextra.

Bug: webrtc:11180
Change-Id: Ia5d1e22bfe42d67cd892ae07620e7fd2daf9a7a4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161332
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30019}
2019-12-05 15:14:32 +00:00
c7a3b08f07 Prefix ENABLE_RTC_EVENT_LOG with WEBRTC_.
Since this macro can be considered public, it makes sense to prefix it
with WEBRTC_ (also to avoid potential conflicts with client code).

This CL also removes some definitions of this macro in order to define
it only where it is strictly needed (it is only used in a .cc file).

Bug: webrtc:11142
Change-Id: Idce7389301e71d8434e238b3cf4ceaa9cf97cd87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161008
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29957}
2019-11-29 09:45:50 +00:00
9dc209a23a Add ability to disable detailed error message in RTC_CHECKs
Bug: webrtc:11133
Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:51:00 +00:00
2dec496f80 Add directive to make TRACE_EVENT macros optional.
Bug: webrtc:11132
Change-Id: I801994ad262e1acff73e4c20afd7a7343b56268c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160654
Commit-Queue: Doudou Kisabaka <doudouk@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29949}
2019-11-28 15:58:24 +00:00
6a4a14635e Add ability to strip out logging messages from the binary
Bug: webrtc:11125
Change-Id: I6e1e96536502c6ae94b6061ea09951cdc2fd87ac
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160410
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29919}
2019-11-26 16:29:50 +00:00
56d945233d Move stun.h to api/.
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).

I checked what our downstream users are actually using, and it's

cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage

I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.

There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.

Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.

Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
2019-11-18 16:11:27 +00:00
02fac7d86e Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

Chromium CL that introduces the component build support:
https://chromium-review.googlesource.com/c/chromium/src/+/1874722

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Tree-Checks: true
No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Iff8e35c6f9a53a0d08979bc873b6488dd7164ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29801}
2019-11-14 19:51:04 +00:00
f11b46bc4b Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit f181137b05e4b899fa3f15afafc4f27e683d83cc.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> No-Try: True
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I8582242910bb3633b7a4675ff261b5a6a0b6954e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159712
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29795}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I6bbe0288d07e49c8a4c808c758ecb6e2ddfa2aa8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159713
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29796}
2019-11-13 16:49:52 +00:00
f181137b05 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I8582242910bb3633b7a4675ff261b5a6a0b6954e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159712
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29795}
2019-11-13 16:45:40 +00:00
06a394e036 Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit 57d53cf1944952ac67df2f6a3522f38cdc01d0c1.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> No-Try: True
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I4719f5b5607ea491689429ca327a3521729e4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159700
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29787}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I1775bdac3ab9888d36f1552dd2eaaa000c43c9b9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29788}
2019-11-13 11:21:52 +00:00
57d53cf194 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I4719f5b5607ea491689429ca327a3521729e4ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29787}
2019-11-13 11:20:40 +00:00