It's has been timing out in a lot of my jobs recently, and for
others too, it appears.
Bug: chromium:749455
Change-Id: I7ee9b70328696d61a2bf41b798a92a55396bad70
Reviewed-on: https://webrtc-review.googlesource.com/38620
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21551}
The gradle git repo seems to be broken, so remove it from
DEPS until it is fixed.
Bug: webrtc:8724
Change-Id: I718d3faadf9c636df8e840b0f8d32c52a73d7da4
Reviewed-on: https://webrtc-review.googlesource.com/38600
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21549}
This means that by default, targets are visible to everything under
the WebRTC root, but not visible to anything else.
API targets are manually tagged with visibility "*", so that targets
outside the WebRTC tree can see them.
BUG=webrtc:8254
Change-Id: Icdbee6e0d22d93240ff2fb530c8f9dc48e351509
Reviewed-on: https://webrtc-review.googlesource.com/24140
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21548}
GetSimulcastConfig should never return an empty vector of VideoStreams, because lower layers in the code expect atleast one VideoStream. It should also never be given input of max_streams equal to 0.
Bug: webrtc:8648
Change-Id: I60f59b3b267a732f07001e4c8a7fa64963802887
Reviewed-on: https://webrtc-review.googlesource.com/38061
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21545}
Now that there aren't any orphans[1], this change makes it a bit
harder to get more of them by, for instance, unlisting them from
.gn files. The previous check only covered new .h files.
The check will not catch all changes that create orphans,
such as when a file is simply dropped from a gn file. It's hard
to implement this, I believe. It should cover the major cases,
such as when a header moves between dirs.
[1] Depends on https://webrtc-review.googlesource.com/c/src/+/38200.
Bug: None
Change-Id: I6b61ea119a9ca1df6ebf381c0f5f4d8897c18b96
Reviewed-on: https://webrtc-review.googlesource.com/38220
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21544}
Changes places where we explicitly construct an Optional to instead use
nullopt or the requisite value type only.
This CL was uploaded by git cl split.
Bug: None
Change-Id: Ie622c215e06956d8d5629733c76f531b7af45012
Reviewed-on: https://webrtc-review.googlesource.com/23568
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21535}
The AudioProcessingBuilder was recently introduced in https://webrtc-review.googlesource.com/c/src/+/34651 to make it easier to create APM instances. This CL replaces all calls to the old Create methods with the new AudioProcessingBuilder.
Bug: webrtc:8668
Change-Id: Ibb5f0fc0dbcc85fcf3355b01bec916f20fe0eb67
Reviewed-on: https://webrtc-review.googlesource.com/36082
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21534}
It shouldn't be empty. As it was before it printed
RESULT min_test_bitrate_no_allocation_strategy: = 80 kbps
Whereas now it prints
RESULT min_test_bitrate_no_allocation_strategy: min_bitrate= 80 kbps
Bug: webrtc:7156
Change-Id: Ie86e3912d296d6d7bd6936d1709df9d2dc7fc143
Reviewed-on: https://webrtc-review.googlesource.com/38040
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21529}
This code is originally written by marc@frankensteinmotorworks.com
Bug: webrtc:8275
Change-Id: I35e6d21b12e71199e0209ff91740d95c9df3bd10
Reviewed-on: https://webrtc-review.googlesource.com/36520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21528}
This is a reland of d2b912aed132c751919ed286439fb39bbd714dda
Original change's description:
> Wiring the RTCRtpEncodingParameters.priority down to the bitrate allocator.
>
> I followed the wiring path for the max bitrate.
> Doc:
> https://docs.google.com/a/google.com/document/d/1sGT6y00prOIErFuGD44zWZacDpR6Rkjg_HXA_Z3Vw4Q/edit?usp=sharing
>
> Bug: webrtc:8630
> Change-Id: I6b861816670442656721c20f81d035ee5eb6218c
> Reviewed-on: https://webrtc-review.googlesource.com/30380
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Elad Alon <eladalon@webrtc.org>
> Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21397}
TBR=solenberg@webrtc.org,eladalon@webrtc.org,sprang@webrtc.org,stefan@webrtc.org
Bug: webrtc:8630
Change-Id: I7429d9e270c9ecb4dfaf6aef85d3055c47658631
Reviewed-on: https://webrtc-review.googlesource.com/35600
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21524}
When Chromium hooks up with the stereo codec, then it has difficulty
communicating with a google chrome without stereo codec. By design, we
do allow codec choice for the standalone codecs, but the problem is
that we do not handle the payload correctly, and thus the existence
of stereo codec will remove the payload registry of the standalone
version of its associated codec. (For example, stereo codec on top of
VP9 will remove the payload registry of standalone VP9 codec.)
This CL fixes the issue. When generating payload data, we should use
"stereo" as payload name, instead of its associated codecs.
Bug: webrtc:8657
Change-Id: I9e0b54de6bd41d370b9353f9553c998e4049789f
Reviewed-on: https://webrtc-review.googlesource.com/33122
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Qiang Chen <qiangchen@chromium.org>
Cr-Commit-Position: refs/heads/master@{#21523}
This adds a SetCapturer function to testing::FakeAudioDevice::PulsedNoiseCapturer
that can be used to update the volume of the generated audio mid-call. It also modifies
CreatePulsedNoiseCapturer to use PulsedNoiseCapturer's type directly so that its new
function is visible for the callers.
Bug: webrtc:8666
Change-Id: I47726e242ccf221f5511e2797b2954ce035ba371
Reviewed-on: https://webrtc-review.googlesource.com/34650
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Erik Varga <erikvarga@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21521}
Seems like there are still issues when uploading.
TBR=phoglund@webrtc.org
No-Try: true
Bug: webrtc:7156
Change-Id: I4ed1149afa1dc4f38ad7d48926f5b624743d1caa
Reviewed-on: https://webrtc-review.googlesource.com/37960
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21520}
This will require a manual roll to downstream projects, since
the //test:perf_test target was introduced.
This is a reland of 10a8e7a9b5261a7e3ce19900ba3511be3b5911f8
Original change's description:
> iOS: Save perf results under Documents/perf_result.json
>
> TBR=henrika@webrtc.org
>
> Bug: webrtc:7156
> Change-Id: Ib00992cce0007e0b5c9274340df1a892f810b0c5
> Reviewed-on: https://webrtc-review.googlesource.com/29202
> Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21244}
R=henrika@webrtc.org, phoglund@webrtc.org
Bug: webrtc:7156
Change-Id: I85fc7bc5fce0894af90017b71b9952b61b523424
Reviewed-on: https://webrtc-review.googlesource.com/37643
Reviewed-by: Patrik Höglund <phoglund@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21518}
This creates a new target for pure defines and interfaces. I think
that makes sense (though include/ makes it harder to see when .cc and
.h files should live together).
Bug: webrtc:7620
Change-Id: Ifb0f50faf99166202836c0446feed3443eb52c6e
Reviewed-on: https://webrtc-review.googlesource.com/34657
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21516}
This is a reland of 727b7d0470c0515397d21698ee089197c31cb5ff
Original change's description:
> Reland "Reland "Put internal video codec factories into separate target""
>
> This is a reland of 0efd1e8b7e69900a6a516a176f1ab69d0e6b8a26
> Original change's description:
> > Reland "Put internal video codec factories into separate target"
> >
> > This is a reland of 51698aefd4925f2dfa0310a321f836d433fa9258
> > Original change's description:
> > > Put internal video codec factories into separate target
> > >
> > > The purpose is to start splitting out the dependencies to the built-in
> > > SW video codecs, so that clients can decide to not depend on them and
> > > get a reduction in binary size.
> > >
> > > Replaces https://webrtc-review.googlesource.com/c/src/+/29101
> > >
> > > Bug: webrtc:7925
> > > Change-Id: I46b95aaf42ead70ba78776de60600b8a66a1fe0c
> > > Reviewed-on: https://webrtc-review.googlesource.com/33420
> > > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#21381}
> >
> > Bug: webrtc:7925
> > Change-Id: I105287fd41ec3ee5bd964b94efcc9c7b3ecdb842
> > Reviewed-on: https://webrtc-review.googlesource.com/35261
> > Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> > Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#21389}
>
> Bug: webrtc:7925
> Change-Id: Id1c7f270676e9e4ca57ca8aa1305cf5554290754
> Reviewed-on: https://webrtc-review.googlesource.com/35501
> Commit-Queue: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21464}
Bug: webrtc:7925
Change-Id: I0b3b5e03d29dadbcbe13cb7ce5369299bb6c0454
Reviewed-on: https://webrtc-review.googlesource.com/37000
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21513}
The removal of these headers has been announced in November with
https://groups.google.com/forum/#!topic/discuss-webrtc/0vWBzJs0yDU.
Bug: webrtc:5883
Change-Id: I6ead2e3bd619472db1a78c0ded5dc57bdb66b76c
Reviewed-on: https://webrtc-review.googlesource.com/34648
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21512}
Starting from Chromium Roll [1], WebRTC should start to use NDK r16
for Android builds. The roll cannot be completed because of three
compilation errors:
../../sdk/android/src/jni/pc/androidnetworkmonitor.cc:15:9: error: 'RTLD_NOLOAD' macro redefined [-Werror,-Wmacro-redefined]
^
../../third_party/android_tools/ndk/sysroot/usr/include/dlfcn.h:62:9: note: previous definition is here
../../modules/audio_device/android/audio_record_jni.cc:251:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
../../modules/audio_device/android/audio_track_jni.cc:229:41: error: format specifies type 'long long' but the argument has type 'jlong' (aka 'long') [-Werror,-Wformat]
ALOGD("direct buffer capacity: %lld", capacity);
This CL forward fixes these errors in order to fix the Chromium Roll
into WebRTC.
[1] - https://webrtc-review.googlesource.com/c/src/+/37540
Bug: webrtc:8710
Change-Id: I5bc64e73919eee7c9e965a442a386b5e1897b56a
Reviewed-on: https://webrtc-review.googlesource.com/37640
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21510}
AddTrack is just a legacy wrapper for the new AddTrack method, so
calling the new AddTrack method should do everything that the old one
does.
Bug: None
Change-Id: I272a9e9584c470d54243377c1307b786f41c660d
Reviewed-on: https://webrtc-review.googlesource.com/37546
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21508}
Bug: webrtc:8714
Change-Id: I107b7d5908cf6efe7f114522959da4503450d7fc
Reviewed-on: https://webrtc-review.googlesource.com/37724
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21506}
Expose RTCDtmfSender API for ObcC SDK via exising RTCRtpSender
to provide ability to use DTMF tones in ObjC apps which uses WebRTC.
Android SDK has already exposed DTMF API via Java's DtmfSender
object, there changes provide similar functionaly to ObjC SDK.
Bug: webrtc:8713
Change-Id: Id68fddbbc362211dc8032fa31b38812d1cff8ed9
Reviewed-on: https://webrtc-review.googlesource.com/35800
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21505}
Exposed setOptions API for iOS SDK via RTCPeerConnectionFactory method
to provide ability to disable encryption and control which network
adapters are ignored.
Only subset of webrtc::PeerConnectionFactoryInterface::Options options
are exposed via iOS SDK, additional options can be exposed as requested.
Android SDK has already exposed setOption API via Java's PeerConnection
constructor, there changes provide similar functionaly to iOS SDK.
Bug: webrtc:8712
Change-Id: Ia2de38cf382afc1bad9bbec6c6eac21ad29aee89
Reviewed-on: https://webrtc-review.googlesource.com/34900
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21504}
