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Author SHA1 Message Date
Karl Wiberg
12edf4ce34 Separate build target for rtc_base/numerics/safe_minmax.h 
So that we can avoid dependency cycles.

Bug: none
Change-Id: I821d9f1319dff01403d6e4e310cbb2d4b2b125e8
Reviewed-on: https://webrtc-review.googlesource.com/60500
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22328}
 2018-03-07 14:12:00 +00:00
Sebastian Jansson
1b2e90beb6 Replaced DestructAndGetRtpStateTask with lambda. 
Slight change in functionality: send_stream_ member is no longer moved
to the QueuedTask. This means that a possible race on access to
send_stream_ will not cause nullpointer dereferencing until the posted
task has been run. Most usages of send_stream_ are protected by
thread_checker_, but not DeliverRtcp and EnableEncodedFrameRecording.

This change in behavior should be be able to cause new failures, but it
could potentially make existing race conditions less likely to happen.

Bug: None
Change-Id: Ife42071a4aa2811fcaf2f3ef21ca1888e6640ca3
Reviewed-on: https://webrtc-review.googlesource.com/59800
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22327}
 2018-03-07 12:52:20 +00:00
Sebastian Jansson
e4d79c73cc Replaced EncoderReconfiguredTask with lambda. 
Bug: None
Change-Id: If5f72592926d60235c1306a34b9126e0074cb92b
Reviewed-on: https://webrtc-review.googlesource.com/59200
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22326}
 2018-03-07 12:47:05 +00:00
Artem Titov
0f03973365 Separate test/fake_audio_device on API and implementation. Step 1. 
Adding ability of injecting audio in end to end tests, that are using
WebRTC. It will be done in 3 steps:
1. Test/fake_audio_device will be moved to production part of WebRTC
source code and renamed to test_audio_device_module. Old header is
replaced with alias to the new one.
2. Internal usage of FakeAudioDevice will be switch to TestAudioDevice.
3. test/fake_audio_device will be removed.

This CL implements 1st step.

Bug: webrtc:8946
Change-Id: Ia8df5155d369d83b3c2818a1129f78dd0848b01f
Reviewed-on: https://webrtc-review.googlesource.com/59740
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22325}
 2018-03-07 12:46:00 +00:00
Harald Alvestrand
2e18061033 Count key protocol for all media sections 
This will give accurate stats for the number of calls
that use video that are using SDES.

Bug: chromium:804275
Change-Id: I35b045a2301fb5267b656b424b9b3482b1b72f9a
Reviewed-on: https://webrtc-review.googlesource.com/60481
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22324}
 2018-03-07 11:32:55 +00:00
philipel
db4fa4b944 Propagating total_bitrate_bps from BitrateAllocator to ProbeController, part 3. 
Trigger on total bitrate change.

Bug: webrtc:8955
Change-Id: I2373a1b7f139c7ea748a9641593e714d6895c8f6
Reviewed-on: https://webrtc-review.googlesource.com/59323
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22323}
 2018-03-07 10:31:35 +00:00
Anders Carlsson
7dbb701076 Fix crash when setting duplicate receive codecs. 
Instead of crashing, log a warning.

Bug: chromium:810173
Change-Id: I7e43889fdab429fcb231657f5770b0ff26f34a8f
Reviewed-on: https://webrtc-review.googlesource.com/59020
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22322}
 2018-03-07 09:57:16 +00:00
Autoroller
69d2dc94d3 Roll chromium_revision 92107f0efb..ec5ab775db (541279:541385) 
Change log: 92107f0efb..ec5ab775db
Full diff: 92107f0efb..ec5ab775db

Changed dependencies:
* src/base: a53e288e7e..5daf8f3473
* src/build: ca739dc873..8c04da5f28
* src/testing: dc30e2eb78..7fb4f2160c
* src/third_party: e34b3d3468..e9c2db0e52
* src/third_party/depot_tools: 462839ea99..c29602466d
* src/tools: 2ddd4aba2e..437d435a01
DEPS diff: 92107f0efb..ec5ab775db/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I1e0526dd1fce2e4a976ad1342e6f118bbaf8d5b3
Reviewed-on: https://webrtc-review.googlesource.com/60462
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22321}
 2018-03-07 09:16:42 +00:00
Anders Carlsson
6deb65c2af Move hardwarevideoencoderfactory jni for building w/o SW codecs. 
This file ended up in the wrong place and prevented building without
SW codecs.

Bug: webrtc:7925
Change-Id: I7909561d96051d5653821130c666ac66938e3edc
Reviewed-on: https://webrtc-review.googlesource.com/59640
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22320}
 2018-03-07 09:13:42 +00:00
Autoroller
b46c552dcd Roll chromium_revision c005394adc..92107f0efb (541159:541279) 
Change log: c005394adc..92107f0efb
Full diff: c005394adc..92107f0efb

Changed dependencies:
* src/base: a5a1a585d8..a53e288e7e
* src/build: 02dbfaae98..ca739dc873
* src/ios: 7c93f49550..9e39e1dc9a
* src/testing: 9d8f30c6ce..dc30e2eb78
* src/third_party: e515f9e2f1..e34b3d3468
* src/third_party/ffmpeg: ef99a5d252..4468d4967f
* src/third_party/googletest/src: fe1144246e..703b4a85a2
* src/third_party/libvpx/source/libvpx: edc9a46876..c6fcb9bb94
* src/tools: f267240af1..2ddd4aba2e
DEPS diff: c005394adc..92107f0efb/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I8f06baec60003ef294bfe32785e6b82e70c720d3
Reviewed-on: https://webrtc-review.googlesource.com/60443
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22319}
 2018-03-07 02:13:12 +00:00
Seth Hampson
9a58cc00e0 Update the android AppRTC to use PeerConnection Unified Plan API. 
This updates AppRTC to use addTrack instead of addStream, and removes
the use of onAddStream, because we no longer have to wait for this to be
fired to set the remote track's video renderers.

Bug: webrtc:8869
Change-Id: I1ecae684a9bc4b30512e8c5d717e72b52c589831
Reviewed-on: https://webrtc-review.googlesource.com/57840
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22318}
 2018-03-07 01:56:01 +00:00
Seth Hampson
c384e14707 Add new PeerConnection APIs to the Java SDK. 
This adds wrappers to the following native APIs:
- SdpSemantics enum added to the RTCConfiguration
- RtpTransceiver
- PeerConnection.addTrack
- PeerConnection.removeTrack
- PeerConnection.addTransceiver
- PeerConnection.getTransceivers
These APIs are used with the new Unified Plan semantics.

Bug: webrtc:8869
Change-Id: I19443f3ff7ffc91a139ad8276331f09e57cec554
Reviewed-on: https://webrtc-review.googlesource.com/57800
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22317}
 2018-03-07 01:17:41 +00:00
Seth Hampson
513449eab9 Changes name of RtpTransceiverInit's stream_labels to stream_ids. 
The naming convention according to the spec is stream id, not stream
labels.Changing things now to be spec compliant, before it is widely
used. This also includes changes to objective C wrapper code to be in
sync with the change.

Bug: webrtc:7932
Change-Id: I5705e6d8a647aaeed860316466a7320132f24b00
Reviewed-on: https://webrtc-review.googlesource.com/59301
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22316}
 2018-03-06 23:42:01 +00:00
Steve Anton
8e20f17d1d Report UMA metrics for received SDP format 
This change adds UMA stats that record the format of the remote offered
SDP. There are three classifications for the SDP format:
- Simple: No more than one audio and one video. Should be compatible
    with both Plan B and Unified Plan endpoints.
- ComplexPlanB: More than one audio or more than one video in the Plan B
    format (e.g., one audio mline with multiple streams).
- ComplexUnifiedPlan: More than one audio or more than one video in the
    Unified Plan format (e.g., two audio mlines).

Bug: chromium:811683
Change-Id: If46394edfa6a812ef313d632e27ec27516c18867
Reviewed-on: https://webrtc-review.googlesource.com/57220
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22315}
 2018-03-06 21:22:51 +00:00
Autoroller
cd7d86b7fe Roll chromium_revision a76818d7bf..c005394adc (541053:541159) 
Change log: a76818d7bf..c005394adc
Full diff: a76818d7bf..c005394adc

Changed dependencies:
* src/base: a84984c792..a5a1a585d8
* src/build: 06a6e63084..02dbfaae98
* src/ios: 5b99a0954d..7c93f49550
* src/testing: d127d05715..9d8f30c6ce
* src/third_party: 2d7c1c2b96..e515f9e2f1
* src/tools: 394bf90f9c..f267240af1
DEPS diff: a76818d7bf..c005394adc/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I3fa203063d26a13eb53d01ea84d2105fba6abc51
Reviewed-on: https://webrtc-review.googlesource.com/60400
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22314}
 2018-03-06 19:41:41 +00:00
Taylor Brandstetter
87d5a7494a Fix crash that occurs if GetStats is called from within OnStatsDelivered 
This was resulting in the currently executing stats callback to be
invoked *again*, possibly ad infinitum, resulting in stack overflow.

Bug: webrtc:8973
Change-Id: Ib3bcc8e6bdd991728214fa242dda2010efc919a7
Reviewed-on: https://webrtc-review.googlesource.com/59464
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22313}
 2018-03-06 18:41:52 +00:00
Per Åhgren
12eb85881c Separating the AEC3 suppressor gain rampup behavior for call startup and in-call resets 
This CL introduces a different rampup behavir for the call startup and after resets
that may occur due to delay changes, clock-drift and audio path glitches.

Bug: chromium:819111, webrtc:8979
Change-Id: Ied1d7896be7f0c69aa6deb61475117021ca6ab09
Reviewed-on: https://webrtc-review.googlesource.com/60002
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22312}
 2018-03-06 15:48:41 +00:00
Danil Chapovalov
e3927c5885 Allow to turn RtcpTransciever on and off at runtime. 
Bug: webrtc:8239
Change-Id: I8678d1ee9cd0da194a1243d40b508bb62cb3f257
Reviewed-on: https://webrtc-review.googlesource.com/60180
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22311}
 2018-03-06 15:19:11 +00:00
Ilya Nikolaevskiy
70473fcac4 Reland "Add hugeFramesSent GetStats metric" 
This is a reland of f9f71b91ae073fdd2b89ff9df1204835aa3137eb
after the change in chromium tests.

Chromium change done here:
https://chromium-review.googlesource.com/c/chromium/src/+/950776

Original reviewed on: https://webrtc-review.googlesource.com/c/src/+/54420

No changes to the original patchset were done.

TBR=hta@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,solenberg@webrtc.org

Bug: webrtc:8901
Change-Id: Ic88c3cb963dceea0426eb90519743e3c1a4533c1
Reviewed-on: https://webrtc-review.googlesource.com/60140
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22310}
 2018-03-06 13:38:11 +00:00
Harald Alvestrand
1e0c804f19 Make DTMF sender wait before sending the first tone. 
This is according to WebRTC spec.

Bug: chromium:819118
Change-Id: I4c0c8e69de812730f9f5c0a1b86d08cf05a472ad
Reviewed-on: https://webrtc-review.googlesource.com/60181
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22309}
 2018-03-06 13:23:11 +00:00
Sam Zackrisson
52f8188f5d Revert "Deprecate the adaptive level controller" 
This reverts commit 6f37ed78d99daa36e964ff0a65b205f0916d9949.

Reason for revert: <INSERT REASONING HERE>

Original change's description:
> Deprecate the adaptive level controller
> 
> Level control handled by default-on AGC.
> 
> Bug: none
> Change-Id: I405daeceece12c896d41156b649fcfd556726f77
> Reviewed-on: https://webrtc-review.googlesource.com/59682
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Alex Loiko <aleloi@webrtc.org>
> Commit-Queue: Sam Zackrisson <saza@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22305}

TBR=solenberg@webrtc.org,saza@webrtc.org,aleloi@webrtc.org

Change-Id: Ic52f41fcbebfd2291a51b17ac788313e1ceef163
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/60240
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22308}
 2018-03-06 11:54:22 +00:00
Niels Möller
2e1d784956 Delete the VideoCodec::plName string. 
It holds the same information as codecType, but in different format.

Bug: webrtc:8830
Change-Id: Ia83e2dff4fd9a5ddb489501b7a1fe80759fa4218
Reviewed-on: https://webrtc-review.googlesource.com/56100
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22307}
 2018-03-06 11:17:41 +00:00
Karl Wiberg
f9fbce9cc4 Add back OWNERS entris that went missing 
The OWNERS entries for style-guide.md were removed in
https://webrtc-review.googlesource.com/1560, apparently by accident;
add them back.

And while we're at it, add an entry for native-api.md as well.

Bug: none
Change-Id: I924b9e8dab297e668055af3042ac70123e4c00dc
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/60001
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22306}
 2018-03-06 10:31:11 +00:00
Sam Zackrisson
6f37ed78d9 Deprecate the adaptive level controller 
Level control handled by default-on AGC.

Bug: none
Change-Id: I405daeceece12c896d41156b649fcfd556726f77
Reviewed-on: https://webrtc-review.googlesource.com/59682
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22305}
 2018-03-06 10:20:01 +00:00
philipel
9771c5050d Clear the FrameBuffer if it's full and a keyframe is being inserted. 
Bug: webrtc:7705, webrtc:8593, chromium:706599, chromium:807624
Change-Id: Ie4e3e217bc2930fe511f8b6ad3a36afed484ab5f
Reviewed-on: https://webrtc-review.googlesource.com/59321
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22304}
 2018-03-06 09:11:11 +00:00
Max Morin
8ddc2e6258 Revert "Add hugeFramesSent GetStats metric" 
This reverts commit f9f71b91ae073fdd2b89ff9df1204835aa3137eb.

Reason for revert: Looks like it's breaking WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise, see https://ci.chromium.org/buildbot/chromium.webrtc.fyi/Mac%20Tester/48322 (win and lin testers are also failing on the same test).

[ RUN      ] WebRtcBrowserTest.RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise
[12743:4099:0305/082149.300326:WARNING:notification_platform_bridge_mac.mm(510)] AlertNotificationService: XPC connection invalidated.
[12743:88323:0305/082150.773242:WARNING:embedded_test_server.cc(228)] Request not handled. Returning 404: /favicon.ico
[12743:775:0305/082150.774044:INFO:CONSOLE(13)] "Requesting doGetUserMedia: constraints: {"audio":true,"video":true}", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082150.969262:INFO:CONSOLE(13)] "Returning request-callback-granted to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082150.983959:INFO:CONSOLE(13)] "Returning ok-got-stream to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.741587:INFO:CONSOLE(13)] "Requesting doGetUserMedia: constraints: {"audio":true,"video":true}", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.749225:INFO:CONSOLE(13)] "Returning request-callback-granted to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.754982:INFO:CONSOLE(13)] "Returning ok-got-stream to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.761516:INFO:CONSOLE(13)] "Returning ok-peerconnection-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:775:0305/082151.762047:WARNING:RTCPeerConnection.cpp(1151)] mediaConstraints is not a supported argument to addStream.
[12752:775:0305/082151.762096:WARNING:RTCPeerConnection.cpp(1153)] mediaConstraints was
[12743:775:0305/082151.762953:INFO:CONSOLE(13)] "Added local stream.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.763010:INFO:CONSOLE(13)] "Returning ok-added to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.767078:INFO:CONSOLE(13)] "Returning ok-peerconnection-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12755:775:0305/082151.767614:WARNING:RTCPeerConnection.cpp(1151)] mediaConstraints is not a supported argument to addStream.
[12755:775:0305/082151.767660:WARNING:RTCPeerConnection.cpp(1153)] mediaConstraints was
[12743:775:0305/082151.768452:INFO:CONSOLE(13)] "Added local stream.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.768523:INFO:CONSOLE(13)] "Returning ok-added to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.776171:INFO:CONSOLE(13)] "Returning ok-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.777197:INFO:CONSOLE(13)] "Returning ok-created to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:42755:0305/082151.777736:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 104 to 127
[12752:42755:0305/082151.777766:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 106 to 125
[12752:42755:0305/082151.777829:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 103 to 124
[12752:42755:0305/082151.777850:WARNING:mediasession.cc(353)] Duplicate id found. Reassigning from 105 to 123
[12743:775:0305/082151.778835:INFO:CONSOLE(13)] "createOffer(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.779780:INFO:CONSOLE(13)] "Returning ok-{"type":"offer","sdp":"v=0\r\no=- 3491235150284933882 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:3632917417 cname:J9N+OjIJeArKjXXh\r\na=ssrc:3632917417 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 6e608085-751b-4945-8982-6f4aedf7bef6\r\na=ssrc:3632917417 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:3632917417 label:6e608085-751b-4945-8982-6f4aedf7bef6\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 124 127 123 125 107 108\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420032\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=127\r\na=rtpmap:125 red/90000\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 ulpfec/90000\r\na=ssrc-group:FID 1955312265 3021315394\r\na=ssrc:1955312265 cname:J9N+OjIJeArKjXXh\r\na=ssrc:1955312265 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:1955312265 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:1955312265 label:7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:3021315394 cname:J9N+OjIJeArKjXXh\r\na=ssrc:3021315394 msid:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU 7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\na=ssrc:3021315394 mslabel:Nb7epCcAUJwMKKzTV36r1RGOlNSIxBooVpgU\r\na=ssrc:3021315394 label:7cd84f82-8fa1-42ca-b0ac-326aa0925f05\r\nm=application 9 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\na=ice-ufrag:ySYi\r\na=ice-pwd:5E4b4cjl+QFLqPoIgvleZ4m4\r\na=ice-options:trickle\r\na=fingerprint:sha-256 94:ED:E9:BB:45:FF:BE:85:C2:98:E5:45:3A:AB:A9:4B:3B:F0:04:D7:B1:05:45:E9:6D:14:3C:FE:62:5C:23:03\r\na=setup:actpass\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n"} to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.781514:INFO:CONSOLE(13)] "setLocalDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12755:41731:0305/082151.782411:WARNING:channel.cc(1039)] Trying to cache the Absolute Send Time extension id but the SRTP is not active.
[12752:43011:0305/082151.884258:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620da600:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12752:43011:0305/082151.884438:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(1, 65536) failed: 0
[12752:43011:0305/082151.884481:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(2, 65536) failed: 0
[12752:43011:0305/082151.884513:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7fab620dd000:video:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12755:41731:0305/082151.922410:WARNING:channel.cc(1039)] Trying to cache the Absolute Send Time extension id but the SRTP is not active.
[12743:775:0305/082151.924626:INFO:CONSOLE(13)] "createAnswer(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.925506:INFO:CONSOLE(13)] "Returning ok-{"type":"answer","sdp":"v=0\r\no=- 6096510228474213355 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video data\r\na=msid-semantic: WMS 7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\nm=audio 9 UDP/TLS/RTP/SAVPF 111 103 104 9 0 8 106 105 13 110 112 113 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus/48000/2\r\na=rtcp-fb:111 transport-cc\r\na=fmtp:111 minptime=10;useinbandfec=1\r\na=rtpmap:103 ISAC/16000\r\na=rtpmap:104 ISAC/32000\r\na=rtpmap:9 G722/8000\r\na=rtpmap:0 PCMU/8000\r\na=rtpmap:8 PCMA/8000\r\na=rtpmap:106 CN/32000\r\na=rtpmap:105 CN/16000\r\na=rtpmap:13 CN/8000\r\na=rtpmap:110 telephone-event/48000\r\na=rtpmap:112 telephone-event/32000\r\na=rtpmap:113 telephone-event/16000\r\na=rtpmap:126 telephone-event/8000\r\na=ssrc:3234277340 cname:PfS0qqt1exijuETX\r\na=ssrc:3234277340 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 9ddb9a77-20aa-42ba-8540-9e32f3dbb0af\r\na=ssrc:3234277340 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:3234277340 label:9ddb9a77-20aa-42ba-8540-9e32f3dbb0af\r\nm=video 9 UDP/TLS/RTP/SAVPF 96 97 98 99 100 101 102 124 127 123 125 107 108\r\nc=IN IP4 0.0.0.0\r\na=rtcp:9 IN IP4 0.0.0.0\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time\r\na=extmap:4 urn:3gpp:video-orientation\r\na=extmap:5 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01\r\na=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay\r\na=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type\r\na=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/video-timing\r\na=sendrecv\r\na=rtcp-mux\r\na=rtcp-rsize\r\na=rtpmap:96 VP8/90000\r\na=rtcp-fb:96 goog-remb\r\na=rtcp-fb:96 transport-cc\r\na=rtcp-fb:96 ccm fir\r\na=rtcp-fb:96 nack\r\na=rtcp-fb:96 nack pli\r\na=rtpmap:97 rtx/90000\r\na=fmtp:97 apt=96\r\na=rtpmap:98 VP9/90000\r\na=rtcp-fb:98 goog-remb\r\na=rtcp-fb:98 transport-cc\r\na=rtcp-fb:98 ccm fir\r\na=rtcp-fb:98 nack\r\na=rtcp-fb:98 nack pli\r\na=rtpmap:99 rtx/90000\r\na=fmtp:99 apt=98\r\na=rtpmap:100 H264/90000\r\na=rtcp-fb:100 goog-remb\r\na=rtcp-fb:100 transport-cc\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=fmtp:100 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=420032\r\na=rtpmap:101 rtx/90000\r\na=fmtp:101 apt=100\r\na=rtpmap:102 H264/90000\r\na=rtcp-fb:102 goog-remb\r\na=rtcp-fb:102 transport-cc\r\na=rtcp-fb:102 ccm fir\r\na=rtcp-fb:102 nack\r\na=rtcp-fb:102 nack pli\r\na=fmtp:102 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=4d0032\r\na=rtpmap:124 rtx/90000\r\na=fmtp:124 apt=102\r\na=rtpmap:127 H264/90000\r\na=rtcp-fb:127 goog-remb\r\na=rtcp-fb:127 transport-cc\r\na=rtcp-fb:127 ccm fir\r\na=rtcp-fb:127 nack\r\na=rtcp-fb:127 nack pli\r\na=fmtp:127 level-asymmetry-allowed=1;packetization-mode=1;profile-level-id=640032\r\na=rtpmap:123 rtx/90000\r\na=fmtp:123 apt=127\r\na=rtpmap:125 red/90000\r\na=rtpmap:107 rtx/90000\r\na=fmtp:107 apt=125\r\na=rtpmap:108 ulpfec/90000\r\na=ssrc-group:FID 3517790794 302440277\r\na=ssrc:3517790794 cname:PfS0qqt1exijuETX\r\na=ssrc:3517790794 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:3517790794 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:3517790794 label:2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:302440277 cname:PfS0qqt1exijuETX\r\na=ssrc:302440277 msid:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN 2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\na=ssrc:302440277 mslabel:7RUsJxsD5w2UmR1xiHiKSWHyYY8RVKSGT8hN\r\na=ssrc:302440277 label:2ffa90e8-b71e-463b-a105-d5a65e0ac205\r\nm=application 9 DTLS/SCTP 5000\r\nc=IN IP4 0.0.0.0\r\nb=AS:30\r\na=ice-ufrag:X54X\r\na=ice-pwd:VsLK5tJ8so82vOn1y+R72WBi\r\na=ice-options:trickle\r\na=fingerprint:sha-256 EC:A9:2D:A2:D9:44:F0:A4:EE:58:FC:32:DF:C4:8C:B0:FC:25:C3:08:BE:7E:D7:59:B8:A0:20:16:DA:5A:A5:7F\r\na=setup:active\r\na=mid:data\r\na=sctpmap:5000 webrtc-datachannel 1024\r\n"} to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.925954:INFO:CONSOLE(13)] "Receiving remote stream...", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.926204:INFO:CONSOLE(13)] "setRemoteDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.939935:INFO:CONSOLE(13)] "Returning ok-verified to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.940232:INFO:CONSOLE(13)] "setLocalDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12752:43011:0305/082151.942049:WARNING:p2ptransportchannel.cc(1093)] SetOption(1, 65536) failed: 0
[12752:43011:0305/082151.942084:WARNING:p2ptransportchannel.cc(1093)] SetOption(2, 65536) failed: 0
[12743:775:0305/082151.946009:INFO:CONSOLE(13)] "Receiving remote stream...", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.946327:INFO:CONSOLE(13)] "setRemoteDescription(): success.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.946367:INFO:CONSOLE(13)] "Returning ok-accepted-answer to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082151.950048:INFO:CONSOLE(368)] "Still ICE gathering - waiting...", source: http://127.0.0.1:50666/webrtc/peerconnection.js (368)
[12755:41731:0305/082152.030690:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(1, 65536) failed: 0
[12755:41731:0305/082152.030759:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(2, 65536) failed: 0
[12755:41731:0305/082152.030785:WARNING:p2ptransportchannel.cc(638)] Jingle:Port[0x7f8a9c809a00:audio:1:0:local:Net[en0:192.168.195.x/24:Ethernet]]: SetOption(5, 0) failed: 0
[12743:775:0305/082152.048464:INFO:CONSOLE(13)] "Returning [{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 65179 typ host generation 0 ufrag X54X network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag X54X network-id 1","sdpMid":"audio","sdpMLineIndex":0}] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.049868:INFO:CONSOLE(13)] "Returning ok-received-candidates to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.050468:INFO:CONSOLE(13)] "Returning [{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 60484 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 62030 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"video","sdpMLineIndex":1},{"candidate":"candidate:2504140996 1 udp 2122260223 192.168.195.156 50175 typ host generation 0 ufrag ySYi network-id 1","sdpMid":"data","sdpMLineIndex":2},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"audio","sdpMLineIndex":0},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"video","sdpMLineIndex":1},{"candidate":"candidate:3686913076 1 tcp 1518280447 192.168.195.156 9 typ host tcptype active generation 0 ufrag ySYi network-id 1","sdpMid":"data","sdpMLineIndex":2}] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.052841:INFO:CONSOLE(13)] "Returning ok-received-candidates to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.053385:INFO:CONSOLE(13)] "Returning ["codec","inbound-rtp","outbound-rtp","peer-connection","stream","track","data-channel","transport","local-candidate","remote-candidate","candidate-pair","certificate"] to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
[12743:775:0305/082152.061797:INFO:CONSOLE(13)] "Returning Test failed: Error: stats.hugeFramesSent is not a whitelisted member: 0
    at failTest (http://127.0.0.1:50666/webrtc/test_functions.js:46:15)
    at verifyStatsIsWhitelisted_ (http://127.0.0.1:50666/webrtc/peerconnection_getstats.js:386:13)
    at http://127.0.0.1:50666/webrtc/peerconnection_getstats.js:273:9 to test.", source: http://127.0.0.1:50666/webrtc/test_functions.js (13)
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc:533: Failure
Value of: base::StartsWith(result, "ok-", base::CompareCase::SENSITIVE)
  Actual: false
Expected: true
../../chrome/browser/media/webrtc/webrtc_browsertest_base.cc:138: Failure
Value of: value
  Actual: false
Expected: true
BrowserTestBase received signal: Segmentation fault: 11. Backtrace:
0   browser_tests                       0x0000000105c700cc base::debug::StackTrace::StackTrace(unsigned long) + 28
1   browser_tests                       0x0000000106271902 content::(anonymous namespace)::DumpStackTraceSignalHandler(int) + 226
2   libsystem_platform.dylib            0x00007fffa63ccb3a _sigtramp + 26
3   ???                                 0x0000000000000000 0x0 + 0
4   browser_tests                       0x0000000102ee29e3 WebRtcTestBase::VerifyStatsGeneratedPromise(content::WebContents*) const + 467
5   browser_tests                       0x0000000102edb4d1 WebRtcBrowserTest_RunsAudioVideoWebRTCCallInTwoTabsGetStatsPromise_Test::RunTestOnMainThread() + 817
6   browser_tests                       0x000000010627162d content::BrowserTestBase::ProxyRunTestOnMainThreadLoop() + 557
7   browser_tests                       0x0000000105da0d23 ChromeBrowserMainParts::PreMainMessageLoopRunImpl() + 4227
8   browser_tests                       0x0000000105d9fb9e ChromeBrowserMainParts::PreMainMessageLoopRun() + 62
9   browser_tests                       0x0000000104a3a3d3 content::BrowserMainLoop::PreMainMessageLoopRun() + 67
10  browser_tests                       0x0000000104df0dc7 content::StartupTaskRunner::RunAllTasksNow() + 39
11  browser_tests                       0x0000000104a38d35 content::BrowserMainLoop::CreateStartupTasks() + 661
12  browser_tests                       0x0000000104a3c8f0 content::BrowserMainRunnerImpl::Initialize(content::MainFunctionParams const&) + 96
13  browser_tests                       0x0000000104a36c94 content::BrowserMain(content::MainFunctionParams const&) + 180
14  browser_tests                       0x0000000105c3ebb9 content::ContentMainRunnerImpl::Run() + 377
15  browser_tests                       0x000000010784a8f4 service_manager::Main(service_manager::MainParams const&) + 2324
16  browser_tests                       0x0000000105c3e094 content::ContentMain(content::ContentMainParams const&) + 68
17  browser_tests                       0x0000000106271216 content::BrowserTestBase::SetUp() + 2550
18  browser_tests                       0x0000000105d2993e InProcessBrowserTest::SetUp() + 398
19  browser_tests                       0x0000000104032b51 testing::Test::Run() + 97
20  browser_tests                       0x0000000104033770 testing::TestInfo::Run() + 288
21  browser_tests                       0x0000000104033cd7 testing::TestCase::Run() + 263
22  browser_tests                       0x000000010403b167 testing::internal::UnitTestImpl::RunAllTests() + 903
23  browser_tests                       0x000000010403adb3 testing::UnitTest::Run() + 163
24  browser_tests                       0x0000000105d41c67 base::TestSuite::Run() + 167
25  browser_tests                       0x0000000105c63755 ChromeTestSuiteRunner::RunTestSuite(int, char**) + 37
26  browser_tests                       0x00000001062b6597 content::LaunchTests(content::TestLauncherDelegate*, unsigned long, int, char**) + 391
27  browser_tests                       0x0000000105c63c3c LaunchChromeTests(unsigned long, content::TestLauncherDelegate*, int, char**) + 348
28  browser_tests                       0x0000000105c636ce main + 94
29  libdyld.dylib                       0x00007fffa61bd235 start + 1
30  ???                                 0x000000000000000a 0x0 + 10


Original change's description:
> Add hugeFramesSent GetStats metric
> 
> Bug: webrtc:8901
> Change-Id: I36021c1160c3426d3bfa0f37ff0adaa35710b93e
> Reviewed-on: https://webrtc-review.googlesource.com/54420
> Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22290}

TBR=solenberg@webrtc.org,ilnik@webrtc.org,hbos@webrtc.org,sprang@webrtc.org,stefan@webrtc.org,hta@webrtc.org

Change-Id: I6a7501c46f928281d357da37f9232bb92c5a4f19
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8901
Reviewed-on: https://webrtc-review.googlesource.com/60120
Reviewed-by: Max Morin <maxmorin@webrtc.org>
Commit-Queue: Max Morin <maxmorin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22303}
 2018-03-06 08:28:52 +00:00
Autoroller
87f827db3e Roll chromium_revision 4f34e798b7..a76818d7bf (540949:541053) 
Change log: 4f34e798b7..a76818d7bf
Full diff: 4f34e798b7..a76818d7bf

Changed dependencies:
* src/base: 820903a0fd..a84984c792
* src/build: 244cd3dc77..06a6e63084
* src/ios: a4b3be210d..5b99a0954d
* src/testing: 8d099e8020..d127d05715
* src/third_party: cea692b439..2d7c1c2b96
* src/tools: 5e6d7f0f14..394bf90f9c
DEPS diff: 4f34e798b7..a76818d7bf/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ib53d884e183ff63eee73bf56132e6396ccec3b17
Reviewed-on: https://webrtc-review.googlesource.com/60102
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22302}
 2018-03-06 08:15:41 +00:00
Karl Wiberg
6b54228e8b Adapt audio codec factory templates to the new codec pair ID arguments 
We use template magic to let them handle both the presence and absence
of the new argument. This will be removed in a later CL, when we can
assume that new argument is always present.

Bug: webrtc:8941
Change-Id: I2d47f7c8572a9f03e742401dcf491b948b161f63
Reviewed-on: https://webrtc-review.googlesource.com/58081
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22301}
 2018-03-06 01:13:10 +00:00
Autoroller
94ffe90a88 Roll chromium_revision 6562397ff8..4f34e798b7 (540826:540949) 
Change log: 6562397ff8..4f34e798b7
Full diff: 6562397ff8..4f34e798b7

Changed dependencies:
* src/base: ee4a0d7bd5..820903a0fd
* src/build: 0fc17e203f..244cd3dc77
* src/ios: 8111fe6961..a4b3be210d
* src/testing: a3fc4d9486..8d099e8020
* src/third_party: cec6995486..cea692b439
* src/third_party/android_tools: https://chromium.googlesource.com/android_tools.git/+log/9a70d48fcd..c22a664c39
* src/tools: 627a19c32b..5e6d7f0f14
DEPS diff: 6562397ff8..4f34e798b7/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: I04fff752e644e25964ebe120382d947da7dec52e
Reviewed-on: https://webrtc-review.googlesource.com/59900
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22300}
 2018-03-05 22:20:00 +00:00
Bjorn Terelius
b2bfba6922 Declare the RtpHeaderExtensionMap* as const in RtpHeaderParser::Parse. 
Bug: None
Change-Id: I38ba9f879dfd5b46f2209f107d20c41529fb645c
Reviewed-on: https://webrtc-review.googlesource.com/59801
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22299}
 2018-03-05 20:50:40 +00:00
Steve Anton
0807d152d5 Remove more dead code from BaseChannel 
This removes the following methods:
- SetAudioSend (directly accessed through MediaChannel now)
- "Early Media" (feature not used)
- GetStats (directly accessed through MediaChannel now)

Bug: None
Change-Id: Ifd075d030b0f5f41e94918979891592a731d5a91
Reviewed-on: https://webrtc-review.googlesource.com/59500
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22298}
 2018-03-05 20:23:00 +00:00
Taylor Brandstetter
0867260590 Fixing data race on vptr of Thread subclasses. 
Thread's constructor calls DoInit, which registers itself with the
MessageQueueManager. This could result in the vptr being read before
the subclass has had a chance to modify it (for example, if another
thread happens to call MessageQueueManager::Clear at the right time).

This is exactly why there's a "DoInit" method, which is intended to be
called by the fully instantiated subclass. This was being done between
MessageQueue/Thread, but not between Thread and its subclasses.

Bug: webrtc:3911
Change-Id: I94d8855da56d9aaf22470ddca12d0b1dd5de249d
Reviewed-on: https://webrtc-review.googlesource.com/59466
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22297}
 2018-03-05 20:09:20 +00:00
Karl Wiberg
98cd810d31 Production code: Pass codec ID argument to audio codecs 
Just a null ID for now, but future CLs will fix that.

Bug: webrtc:8941
Change-Id: I393af0fef752ca3711421bdaf4b2e41cbe286bcf
Reviewed-on: https://webrtc-review.googlesource.com/58093
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22296}
 2018-03-05 18:55:19 +00:00
Honghai Zhang
2e7a459fe7 Reland "Fix race conditions in NetworkMonitor. This change makes the class thread-safe." 
This is a reland of 1f4cb9f22d87287cec331c4713c6969da50d8bd6

Original change's description:
> Fix race conditions in NetworkMonitor.
> This change makes the class thread-safe.
>
> Bug: b/73773043
> Change-Id: I1ad13e4f15907e3dd1fef1307f9c654e53b69b22
> Reviewed-on: https://webrtc-review.googlesource.com/57040
> Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22238}

Bug: b/73773043
Change-Id: I9279d514d0735327f9c133be445e5131aace5722
TBR: sakal@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/59240
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22295}
 2018-03-05 18:54:00 +00:00
Ilya Nikolaevskiy
94065690f6 Use round robin packet queue in the pacer by default 
Bug: webrtc:8968
Change-Id: Ibf7d7917cd8ac6093b0994a8ac206c6934c5d6e8
Reviewed-on: https://webrtc-review.googlesource.com/59325
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22294}
 2018-03-05 17:54:00 +00:00
Ying Wang
012b7e7473 Add a couple of logs. 
Bug: webrtc:8963
Change-Id: I462b0fe493306429fdec499f1324f06a80ae17ac
Reviewed-on: https://webrtc-review.googlesource.com/59681
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22293}
 2018-03-05 16:42:02 +00:00
Autoroller
dcb4cd1085 Roll chromium_revision bb5e87b52c..6562397ff8 (540725:540826) 
Change log: bb5e87b52c..6562397ff8
Full diff: bb5e87b52c..6562397ff8

Changed dependencies:
* src/base: e0b732554d..ee4a0d7bd5
* src/build: a81ca97020..0fc17e203f
* src/testing: 0dcf0bcd14..a3fc4d9486
* src/third_party: 0aeb75e705..cec6995486
* src/third_party/android_ndk: https://chromium.googlesource.com/android_ndk.git/+log/e951c37287..635bc38096
* src/third_party/depot_tools: d0de9616e5..462839ea99
* src/tools: 571b1312e0..627a19c32b
DEPS diff: bb5e87b52c..6562397ff8/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

Change-Id: Ib090ca2df65d4e3999b1057f2da7c34668d17e1f
Reviewed-on: https://webrtc-review.googlesource.com/59701
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22292}
 2018-03-05 16:20:16 +00:00
Artem Titov
6723cdc8a4 Revert "Separate test/fake_audio_device on API and implementation." 
This reverts commit 8ea5f9ae5b757aa3a0e6abe46f5c9ef3aaf4b337.

Reason for revert: breaks downstream project

Original change's description:
> Separate test/fake_audio_device on API and implementation.
> 
> Adding ability of injecting audio in end to end tests, that are using
> WebRTC. For this purpose as a 1st step test/fake_audio_device will
> be moved to production part of WebRTC source code and renamed to
> test_audio_device_module. Old header is replaced with alias to the
> new one and will be deleted after a while.
> 
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> 
> Bug: webrtc:8946
> Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
> Reviewed-on: https://webrtc-review.googlesource.com/58086
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22289}

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

Change-Id: I88d9c4f09cc576ed7c9182dcf0a873d25a8bab54
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8946
Reviewed-on: https://webrtc-review.googlesource.com/59720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22291}
 2018-03-05 15:36:23 +00:00
Ilya Nikolaevskiy
f9f71b91ae Add hugeFramesSent GetStats metric 
Bug: webrtc:8901
Change-Id: I36021c1160c3426d3bfa0f37ff0adaa35710b93e
Reviewed-on: https://webrtc-review.googlesource.com/54420
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22290}
 2018-03-05 15:09:12 +00:00
Artem Titov
8ea5f9ae5b Separate test/fake_audio_device on API and implementation. 
Adding ability of injecting audio in end to end tests, that are using
WebRTC. For this purpose as a 1st step test/fake_audio_device will
be moved to production part of WebRTC source code and renamed to
test_audio_device_module. Old header is replaced with alias to the
new one and will be deleted after a while.

Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c

Bug: webrtc:8946
Change-Id: I5284d1dd46ce9bf86cf43268e855520606fa8c5c
Reviewed-on: https://webrtc-review.googlesource.com/58086
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22289}
 2018-03-05 14:30:42 +00:00
Sam Zackrisson
a5797c2bf2 Clean up Android API audio settings 
This removes the routing for the deprecated audio control setting

Bug: none
Change-Id: If7a134ee487b80a653ba982768ba74ce2d539e0a
Reviewed-on: https://webrtc-review.googlesource.com/58941
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22288}
 2018-03-05 14:10:42 +00:00
Harald Alvestrand
f9d0f1d215 UMA counters for SDES x media type 
These counters will register whether the media sections
used with SDES are for audio, video or data.

Bug: chromium:804275
Change-Id: I1da3bb6625af755c0897bf4cd349655cb283fbb6
Reviewed-on: https://webrtc-review.googlesource.com/59400
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22287}
 2018-03-05 13:46:43 +00:00
Ying Wang
0133d46847 Use a smaller size of sequence number set, to improve performance 
Bug: webrtc:8857
Change-Id: I78b4e6d191b1b7eb96f5109323ef48b24b99c7c2
Reviewed-on: https://webrtc-review.googlesource.com/49361
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22286}
 2018-03-05 13:12:32 +00:00
Sebastian Jansson
8d9dcb1c89 Adding bps_or method to DataRate class. 
Bug: webrtc:8415
Change-Id: I64e46b63d82cb843f0710839c1fc22e2440ae7e1
Reviewed-on: https://webrtc-review.googlesource.com/59222
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22285}
 2018-03-05 11:47:52 +00:00
Rasmus Brandt
e90636adc2 Codec instantiation tests for testing device capabilities in batch. 
Change-Id: I465fb165bec9d6501015ec57d13fb556c4cb532b
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/58643
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22284}
 2018-03-05 10:01:02 +00:00
Kári Tristan Helgason
6375ef7dfb Enable VideoProcessorIntegrationTest on devices. 
These tests cannot run on simulators but should be enabled on real device
bots in order to catch regressions or crashes in the iOS codecs.

Bug: webrtc:8950
Change-Id: I8e877aa4368683073fdb4586cd6f4add4a1284ad
Reviewed-on: https://webrtc-review.googlesource.com/59040
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22283}
 2018-03-05 09:52:32 +00:00
Sam Zackrisson
4d3644979c Add stub draft of audio generator to APM 
This provides the empty shell of an AudioGenerator class.
It is intended to be used for debugging purposes and can be inserted
into the APM much like an AecDump. It allows for playing out diagnostic
audio unaffected by codecs and network jitter, while still capturing
API interaction like in a normal call.

NOTRY=True

Bug: webrtc:8882
Change-Id: I8132afc95cdba02ab233f44e22e0a5f530710ef7
Reviewed-on: https://webrtc-review.googlesource.com/53300
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22282}
 2018-03-05 09:28:52 +00:00
Ilya Nikolaevskiy
4c09d7af8a In webrtc_perf_tests always create screenshare stream after realtime video 
Bug: chromium:818127
Change-Id: Ifbb09a81d6a393d0861d6dc2c2e806b127bf76fa
Reviewed-on: https://webrtc-review.googlesource.com/59322
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22281}
 2018-03-05 09:09:22 +00:00
Niels Möller
3f693b9e75 Delete unused method SetPeriodicKeyFrames. 
Keyframe interval is configurable in codec settings, with no need for
a setter method to toggle it on or off.

Bug: webrtc:8830
Change-Id: Ic20d8829884ed22588f8f8c0cceddd76144a9858
Reviewed-on: https://webrtc-review.googlesource.com/56040
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22280}
 2018-03-05 08:54:32 +00:00
Sam Zackrisson
9e981f0e43 Clean up iOS API audio settings 
This removes the routing for the deprecated audio control setting

Change-Id: Id83ff548625279d5b34c9e3cadc097c25a00ef05
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/58900
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22279}
 2018-03-05 08:32:52 +00:00