21752 Commits

Author SHA1 Message Date
Dino Radaković
e9d2e4d3fb Provide the option of injecting rtc::TaskQueue when creating RtcEventLogImpl via factory methods.
Bug: webrtc:9004
Change-Id: Ia7cc96074dbf84f576e5fb0762866b213ec8e69f
Reviewed-on: https://webrtc-review.googlesource.com/63022
Commit-Queue: Dino Radaković <dinor@webrtc.org>
Reviewed-by: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22520}
2018-03-20 18:06:18 +00:00
Mirko Bonadei
fe48ee9e2c Fixing zlib license generation.
WebRTC license has been added in a wrong copy/paste in https://webrtc-review.googlesource.com/c/src/+/18244.

Bug: None
Change-Id: I7e16186fd0aac39bb1a5b934a7f5ba8ea31ef3ad
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/63341
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22519}
2018-03-20 16:57:08 +00:00
Sebastian Jansson
b38b05b4b8 Adding srte as owner in modules/congestion_controller.
Adding srte@webrtc.org as owner in modules/congestion_controller.

Bug: None
Change-Id: Ib96f9d8cbb13d34d9193f4f9317a1c884049179d
Reviewed-on: https://webrtc-review.googlesource.com/63340
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22518}
2018-03-20 16:29:08 +00:00
Paulina Hensman
89dd7bf924 Move android audio device code into sdk/android
This CL adds a stand-alone Android AudioDeviceModule in the
sdk/android folder. It's forked from modules/audio_device/android/
and then simplified for the Android case. The stand-alone Android
ADM is available both in the native_api and also under a field trial
in the Java API.

Bug: webrtc:7452
Change-Id: If6e558026bd0ccb52f56d78ac833339a5789d300
Reviewed-on: https://webrtc-review.googlesource.com/60541
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22517}
2018-03-20 16:04:33 +00:00
Niels Möller
4d22a6d8db Delete unneeded includes of wav_file.h and file_wrapper.h.
Bug: None
Change-Id: I9191950d9c9449656cc0f206daac3aff2e0ed0c3
Reviewed-on: https://webrtc-review.googlesource.com/63180
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22516}
2018-03-20 15:59:27 +00:00
Christoffer Rodbro
8ef59a431f Added data member access methods to FakeNetworkPipe.
Give internal test tools access to FakeNetworkPipe data members
by adding a set of access methods.

Also deleted copy assignment operator for NetworkPacket.

Bug: None
Change-Id: I451a21e0cc6ec82ea830cf197c7a4cef0789623c
Reviewed-on: https://webrtc-review.googlesource.com/63301
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22515}
2018-03-20 15:58:21 +00:00
Sebastian Jansson
3dc0125cf7 Moving ConfigureEncoderTask to the calling scope.
This CL replaces ConfigureEncoderTask with a slightly simpler struct
that is defined within the scope of the function using it. This makes
it more clear that it is only used once and slightly reduces the amount
of code.

Bug: None
Change-Id: I181a3da90540aa514048ff77fc9a9e9cf19d8f34
Reviewed-on: https://webrtc-review.googlesource.com/63026
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22514}
2018-03-20 14:05:55 +00:00
Autoroller
68a716878b Roll chromium_revision ee966518c2..8c0344a12e (544233:544337)
Change log: ee966518c2..8c0344a12e
Full diff: ee966518c2..8c0344a12e

Changed dependencies:
* src/build: 9390b4c944..ba340a5fd7
* src/buildtools: 2888931260..3748a2a908
* src/ios: 4c6eb173f2..020fb857ff
* src/testing: 25ec0a32b3..d820d168d0
* src/third_party: 7610f6ef81..a1c4df95ad
* src/third_party/depot_tools: 544b744621..0b89935ad6
* src/tools: 90a1ffadca..5c94c75feb
DEPS diff: ee966518c2..8c0344a12e/DEPS

No update to Clang.

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

Change-Id: I9adecfd3865b9844c114e61cedb3e5a246cee06c
Reviewed-on: https://webrtc-review.googlesource.com/63282
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22513}
2018-03-20 13:25:05 +00:00
philipel
2e0da5aa38 Remove EncodedFrame picture_id/spatial_layer references.
Bug: webrtc:8909
Change-Id: I8499968829aaa8069b92b4a3cda81bfea759f2a4
Reviewed-on: https://webrtc-review.googlesource.com/63262
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22512}
2018-03-20 12:17:35 +00:00
Sergey Silkin
f18072e556 Enable SVC based on number of SSRCs.
Number of spatial layers is set equal to number of SSRCs. The maximum
value is limited to 3. If spatial layering is enabled, i.e. number of
spatial layers is greater than 1, then number of temporal layers is set
to 3. Otherwise number of temporal layers is set to 1.

Number of spatial and temporal layers can be overwritten through field
trial.

Bug: webrtc:8931
Change-Id: I37bd7fe053529683dc3e91b4e544fbdb44429340
Reviewed-on: https://webrtc-review.googlesource.com/59440
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22511}
2018-03-20 12:14:45 +00:00
henrika
eb98c727e6 Minor improvements in ADM unittest for Windows.
This CL resolves some minor issues related to running ADM unittests on Windows.
It is rather common on Windows that devices can't be opened up in mono mode and
some tests have been hardcoded to use mono and that leads to crashes and/or error
logs. Now, all tests runs in stereo as well.

NOTRY=TRUE

Bug: None
Change-Id: Iebf11a6ff63c19ff1be45575a8e0a3df4e112bd4
Reviewed-on: https://webrtc-review.googlesource.com/62940
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22510}
2018-03-20 11:57:35 +00:00
Niels Möller
faed53837d Delete obsolete alias RateLimiter and rtc_base/ratelimiter.h
Usage in Chrome was eliminated in cl
https://chromium-review.googlesource.com/c/chromium/src/+/966602.

Bug: webrtc:6424
Change-Id: I84bdf830724f4437ed54c37013f676bb40e4dbfc
Reviewed-on: https://webrtc-review.googlesource.com/62800
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22509}
2018-03-20 10:41:15 +00:00
Rasmus Brandt
9047dac757 Disable flaky test SendSideCongestionControllerTest/PacerQueueEncodeRatePushback.
TBR=srte@webrtc.org
NOTRY=TRUE

Bug: webrtc:9039
Change-Id: Ieb4cd437113e6291a326dff05dbcb96cbfdc06a6
Reviewed-on: https://webrtc-review.googlesource.com/63260
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22508}
2018-03-20 10:12:45 +00:00
Oleh Prypin
d328ec3dfa Change valueOf -> parseBoolean to avoid unneeded boxing
Bug: None
Change-Id: Id629c89f40d3f20b050c75d33caf308d77e9eb59
Reviewed-on: https://webrtc-review.googlesource.com/63181
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22507}
2018-03-20 09:52:35 +00:00
Autoroller
a98bb2de15 Roll chromium_revision 02de71e34e..ee966518c2 (544129:544233)
Change log: 02de71e34e..ee966518c2
Full diff: 02de71e34e..ee966518c2

Changed dependencies:
* src/base: 0f2a39f9e0..ef9c914584
* src/build: 50e3861371..9390b4c944
* src/ios: 1ac481cd0d..4c6eb173f2
* src/testing: f0027c8059..25ec0a32b3
* src/third_party: 5c40e5dfa5..7610f6ef81
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/2a467ca516..734f737c6b
* src/tools: b5a4beaa26..90a1ffadca
DEPS diff: 02de71e34e..ee966518c2/DEPS

No update to Clang.

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

Change-Id: I74fdbf9ca051199ecfbaae4b0471b4cda770a3b0
Reviewed-on: https://webrtc-review.googlesource.com/63140
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22506}
2018-03-20 01:21:15 +00:00
Autoroller
114155bcd8 Roll chromium_revision 163641576c..02de71e34e (544029:544129)
Change log: 163641576c..02de71e34e
Full diff: 163641576c..02de71e34e

Changed dependencies:
* src/base: e9b524587a..0f2a39f9e0
* src/build: 4410b727d4..50e3861371
* src/ios: 4fdd9e6dda..1ac481cd0d
* src/third_party: 3f6c9f2075..5c40e5dfa5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/fa0f0f2dd7..2a467ca516
* src/tools: b34742185b..b5a4beaa26
DEPS diff: 163641576c..02de71e34e/DEPS

No update to Clang.

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

Change-Id: Ib571050968cc85dc95c971463dd903b087e4fad9
Reviewed-on: https://webrtc-review.googlesource.com/63100
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22505}
2018-03-19 20:15:04 +00:00
Taylor Brandstetter
01cb5f2cee Fix issue where sockets bound to temporary IPv6 addresses are discarded.
Also removing the implicit InterfaceAddress constructor that takes an
IPAddress, so that issues like this won't happen in the future.

And adding a convenience "Network::AddIP" method that takes an
IPAddress, so that code doing that (previously relying on the implicit
constructor) will continue to work.

Bug: webrtc:8972
Change-Id: Id5cf0fca481cfee3f8ab83412fcb41886535bba2
Reviewed-on: https://webrtc-review.googlesource.com/59461
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22504}
2018-03-19 19:22:31 +00:00
Harald Alvestrand
3d976f6066 Discard link to media channel when audio sender stopped.
Bug: chromium:822799
Change-Id: Ib863cf048318b04369cc51ed1b1c8b03010a2fd2
Reviewed-on: https://webrtc-review.googlesource.com/62941
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22503}
2018-03-19 18:39:01 +00:00
Erik Språng
bb60a3a5fa Refactor VP8 TemporalLayers
This CL moves all temporal layer rate allocation from
DefaultTemporalLayers and ScreenshareLayers into SimulcastRateAllocator.
This means we don't need an extra call-out to the TemporalLayers
interface to get the last allocation, which simplifies the code path a
lot.

It also paves the wave for removing the TemporalLayersFactory interface
(in a separate cl), which will further simplify the ownership model.

Bug: webrtc:9012
Change-Id: I6540b1848efa1a136dce449f13902ad479d5ee37
Reviewed-on: https://webrtc-review.googlesource.com/62420
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22502}
2018-03-19 18:14:21 +00:00
Mirko Bonadei
d7573563a4 Fixing -Wstrict-prototypes warnings.
Bug: webrtc:8984
Change-Id: I9a7ffb0038f341bfec055f021fc203c7d45d72fa
Reviewed-on: https://webrtc-review.googlesource.com/60903
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@{#22501}
2018-03-19 16:57:21 +00:00
Niels Möller
def1ef5603 New equality operators, for structs related to webrtc::VideoCodec.
Added for the structs VideoCodecVP8, VideoCodecVP9, VideoCodecH264,
and SpatialLayer.

New operators are used to replace memcmp in VCMEncoderDataBase. Using
memcmp to compare structs is generally unreliable, since the struct
may contain random padding bytes due to alignment requirements
(affects at least VideoCodecH264). And in the case of VideoCodecVP8,
we need to exclude the tl_factory pointers from the comparison.

Bug: webrtc:8830
Change-Id: I40432ea7834e288f8c89ce0a28a630ae1800dff8
Reviewed-on: https://webrtc-review.googlesource.com/62761
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22500}
2018-03-19 15:54:21 +00:00
Henrik Boström
5b3541f9af RTCStatsCollector::GetStatsReport() with optional selector argument.
This implements the stats selection algorithm[1] in RTCStatsCollector by
obtaining the selector's inbound-rtp/outbound-rtp stats and performing
the stats traversal algorithm (TakeReferencedStats)[2] on a copy of the
cached report with the rtps as starting point.

Changes:
- RTCStatsCollector.GetStatsReport() with selector arguments added.
  - RequestInfo added, "callbacks_" is replaced by "requests_".
- RTCStatsReport.Copy() added.
- New test for sender selector and receiver selector,
  RTCStatsCollectorTest.GetStatsWithSelector.

[1] https://w3c.github.io/webrtc-pc/#dfn-stats-selection-algorithm
[2] https://cs.chromium.org/chromium/src/third_party/webrtc/pc/rtcstatstraversal.h

Bug: chromium:680172
Change-Id: I9eff00738a1f24c94c9c8ecd13c1304452e962cf
Reviewed-on: https://webrtc-review.googlesource.com/62141
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22499}
2018-03-19 15:32:16 +00:00
Sebastian Jansson
4a73cd436c Adding tests of TaskQueueCongestionControl field trial.
Adding TaskQueueCongestionControl field trial to parametrized end to
end tests. This ensures that enabling the field trial will not break the
functionality tested in the tests.

Bug: webrtc:8415
Change-Id: Ieac75b840f18af2d9d5d35f976e119a8b3e7bfc0
Reviewed-on: https://webrtc-review.googlesource.com/61722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22498}
2018-03-19 15:31:11 +00:00
Artem Titov
e62f600c42 Extend WavReader and WavWriter API.
Add ability to read and write wav files using rtc::PlatformFile instead
of file name.

Bug: webrtc:8946
Change-Id: If18d9465f2155a33547f800edbdac45971a0e878
Reviewed-on: https://webrtc-review.googlesource.com/61424
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22497}
2018-03-19 15:21:51 +00:00
Autoroller
451dfdf6db Roll chromium_revision 7a8a322ad7..163641576c (543921:544029)
Change log: 7a8a322ad7..163641576c
Full diff: 7a8a322ad7..163641576c

Changed dependencies:
* src/build: de87b44a7f..4410b727d4
* src/testing: 6f09eccc24..f0027c8059
* src/third_party: 432c55f861..3f6c9f2075
* src/tools: b3c542c8dc..b34742185b
DEPS diff: 7a8a322ad7..163641576c/DEPS

No update to Clang.

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

Change-Id: I78b3f37a864c9160ad7042037bd31ca5265020fe
Reviewed-on: https://webrtc-review.googlesource.com/62882
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22496}
2018-03-19 15:19:01 +00:00
philipel
0fa82a60e9 Moved FrameKey to api/video/encoded_frame.h and renamed it to VideoLayerFrameId.
Since we want the VideoStreamDecoder to callback with the last
continuous frame we need to move the FrameKey into the public API.

Bug: webrtc:8909
Change-Id: I39634145d848b8163778e31a1e0d04d91f9bbeb8
Reviewed-on: https://webrtc-review.googlesource.com/60864
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22495}
2018-03-19 15:13:11 +00:00
Oleh Prypin
9c1ee368e0 Fix low_bandwidth_audio_perf_test resource dependency on Android
The executable that's pushed to the device must depend on all
files that need to be on the device.

No-Try: True
TBR: phoglund@webrtc.org
Bug: chromium:755660
Change-Id: Iee041bd51e789e3ce6612fbda1582286a5cf4680
Reviewed-on: https://webrtc-review.googlesource.com/62961
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22494}
2018-03-19 15:02:51 +00:00
Alex Loiko
b9a02e523c Change place of UMA logging in AudioMixer.
And fix typo in UMA metric.

We have this pattern in the FrameCombiner component of the AudioMixer:

  if (number_of_streams <= 1) {
    // Copy or fill with zeros.
    return;
  }
  // Mix and limit
  LogMixingStats(/* args */);

When there is only one remote stream, info about active streams and
sample rate is not logged. This CL moves the call to log stats before
the 'return'.

Bug: webrtc:8925
Change-Id: I7b54f61f628273631909dafbfafa21e155e18d4a
Reviewed-on: https://webrtc-review.googlesource.com/62860
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22493}
2018-03-19 14:10:51 +00:00
Sebastian Jansson
537012405b Replacing unique pointer with raw pointer in SSCC checks.
Replacing the unique pointer used for access checks with a raw pointer
pointing to the object owned by the unique pointer. This is to stop
tsan from detecting a race between .get() done on the task queue and
.reset() done in the destructor.

Bug: webrtc:8415
Change-Id: Iae2ea9a2d38f319e73146e6b1e360b11b1708c76
Reviewed-on: https://webrtc-review.googlesource.com/62560
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22492}
2018-03-19 13:46:41 +00:00
Oleh Prypin
04d49500e2 Revert "Using safe casts of allocation limits in Call."
This reverts commit 4a9b4d6332f596867d2a8fb34ff5b4befb9848eb.

Reason for revert: Breaks downstream projects

Original change's description:
> Using safe casts of allocation limits in Call.
> 
> Bug: None
> Change-Id: I71d0e1f92bf820d117b354dd7701c9c719cc2c0a
> Reviewed-on: https://webrtc-review.googlesource.com/61784
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22490}

TBR=nisse@webrtc.org,srte@webrtc.org

Change-Id: I720e97981574fd152cb7ed4204e29f9ea0b2e909
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/62920
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22491}
2018-03-19 13:29:52 +00:00
Sebastian Jansson
4a9b4d6332 Using safe casts of allocation limits in Call.
Bug: None
Change-Id: I71d0e1f92bf820d117b354dd7701c9c719cc2c0a
Reviewed-on: https://webrtc-review.googlesource.com/61784
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22490}
2018-03-19 12:18:51 +00:00
Sebastian Jansson
8d8cb56f3e Delete obsolete methods from MockRtpTransportControllerSend
Removing functions that has been removed from
RtpTransportControllerSendInterface from
MockRtpTransportControllerSend.

Deleted functions: GetPacerModule, GetModule,
SetTransportOverhead and AvailableBandwidth.

Bug: webrtc:8415
Change-Id: I24d460bd18d57966e3b333ce0c234c3e3dc19a9a
Reviewed-on: https://webrtc-review.googlesource.com/62762
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22489}
2018-03-19 11:42:11 +00:00
Oleh Prypin
b708e93ad0 Bring mb up to date with Chromium's changes
Bug: None
Change-Id: I87abffb0bba6ee945447f3e651a505d7602fa15d
Reviewed-on: https://webrtc-review.googlesource.com/62641
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22488}
2018-03-19 11:17:41 +00:00
Tommi
8d2c5a810f Detangling target dependencies in rtc_base_approved.
The eventual goal is to allow PlatformThread to use
SequencedTaskChecker, but getting to that point will require
some more detangling.

Here are (roughly) the steps taken in this CL:
* Make constructormagic a separate target.
* Move atomicops and arraysize to separate targets
* Move platform_thread_types to a separate target
* Move criticalsection to a separate target
* Move thread_checker to separate target
* Make sequenced_task_checker not depend on base_approved
* Move ptr_util to a separate target
* Move scoped_ptr to ptr_util
* Make rtc_task_queue_api not depend on base_approved
* Make sequenced_task_checker depend on rtc_task_queue_api
* Move rtc::Event to its own target
* Move basictypes.h to constructormagic
* Move format_macros and stringize_macros into constructormagic
* Rename constructormagic target to... macromagic
* Move stringencode to stringutils
* New target for safe_conversions
* Move timeutils to a new target.
* Move logging to a new target.
* Move platform_thread to a new target.
* Make refcount a new target (refcount, refcountedobject, refcounter).
* Remove rtc_base_approved from deps of TQ
* Remove a circular dependency between event tracer and platform thread.

Further steps will probably be to factor TaskQueue::Current() to not
be a part of the TaskQueue class itself and have it declared+implemented
in a target that's lower level than TQ itself. SequencedTaskChecker can
then depend on that target and avoid the TQ dependency. Once we're there,
PlatformThread will be able to depend on SequencedTaskChecker.

Attempted but eventually removed from this CL:
* Make TQ a part of rtc_base_approved
* Remove direct dependencies on sequenced_task_checker.
* Profit.

A few include-what-you-use updates along the way.
Fix a few targets that were depending on rtc_task_queue_api

Change-Id: Iee79aa2e81d978444c51b3005db9df7dc12d92a9
Bug: webrtc:8957
Reviewed-on: https://webrtc-review.googlesource.com/58480
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22487}
2018-03-19 11:02:01 +00:00
Oleh Prypin
7b2676fee9 Fix low_bandwidth_audio_perf_test binary dependency on Windows
The split in https://webrtc-review.googlesource.com/c/src/+/62660
broke it.

No-Try: True
Bug: chromium:755660
Change-Id: I664f022cac9f8e7e0bb64a7cb59992f030543aa6
Reviewed-on: https://webrtc-review.googlesource.com/62801
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22486}
2018-03-19 10:54:11 +00:00
Mirko Bonadei
d2c8332e2b Revert "Relaxing no-streams presubmit check (streams are allowed in tests)."
This reverts commit 73ac90863d339599e6fc42fc5228282f479ebc0d.

Reason for revert: Sometimes 'gn refs' exits with status 1.

Original change's description:
> Relaxing no-streams presubmit check (streams are allowed in tests).
> 
> It is actually fine to use streams in testonly code. This CL relaxes
> the presubmit check in order allow streams usage in tests.
> 
> Bug: webrtc:8982
> Change-Id: I18bbf079e804815956cd94ac761cc13022c0761e
> No-Try: True
> Reviewed-on: https://webrtc-review.googlesource.com/61701
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
> Cr-Commit-Position: refs/heads/master@{#22482}

TBR=phoglund@webrtc.org,mbonadei@webrtc.org,tommi@webrtc.org,srte@webrtc.org,tommi@chromium.org

Change-Id: I053b953896ca66be26835b60fb245d5ac0832294
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8982
Reviewed-on: https://webrtc-review.googlesource.com/62780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22485}
2018-03-19 10:32:02 +00:00
Anders Carlsson
7311918269 Add an example app for iOS native API.
Demonstrates how to use the iOS native API to wrap components into
C++ classes.

This CL also introduces a native API wrapper for the capturer.

The C++ code is forked from the corresponding CL for Android at
https://webrtc-review.googlesource.com/c/src/+/60540

Bug: webrtc:8832
Change-Id: I12d9f30e701c0222628e329218f6d5bfca26e6e0
Reviewed-on: https://webrtc-review.googlesource.com/61422
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22484}
2018-03-19 09:31:06 +00:00
Oleh Prypin
8cf0a87bc3 Reland "Split perf-test-specific resources in low_bandwidth_audio_test"
This is a reland of 4bbc150b18e961811991e3e524378e703b6d5b31

Now with explicitly specified `write_runtime_deps`

Original change's description:
> Split perf-test-specific resources in low_bandwidth_audio_test
>
> Bug: chromium:755660
> Change-Id: I7c60a47b26ad86892218497f28a09a04574077e6
> Reviewed-on: https://webrtc-review.googlesource.com/61961
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22439}

Bug: chromium:755660
TBR: phoglund@webrtc.org
Change-Id: I3d4bcc5156ee25de399ab23773ecb73cd995075c
Reviewed-on: https://webrtc-review.googlesource.com/62660
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22483}
2018-03-19 09:30:01 +00:00
Mirko Bonadei
73ac90863d Relaxing no-streams presubmit check (streams are allowed in tests).
It is actually fine to use streams in testonly code. This CL relaxes
the presubmit check in order allow streams usage in tests.

Bug: webrtc:8982
Change-Id: I18bbf079e804815956cd94ac761cc13022c0761e
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/61701
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@chromium.org>
Cr-Commit-Position: refs/heads/master@{#22482}
2018-03-19 09:08:51 +00:00
Oleh Prypin
9fa35e5285 Fix path to proto in py_event_log_analyzer/pb_parse.py
Bug: chromium:611808
No-Try: True
Change-Id: I173f0270a07896d9edddfef6b68592e6b404ecab
Reviewed-on: https://webrtc-review.googlesource.com/62680
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22481}
2018-03-19 07:42:35 +00:00
Autoroller
8a1b20a213 Roll chromium_revision 33aa22e76e..7a8a322ad7 (543816:543921)
Change log: 33aa22e76e..7a8a322ad7
Full diff: 33aa22e76e..7a8a322ad7

Changed dependencies:
* src/base: fcec2204ef..e9b524587a
* src/build: 7642603cf6..de87b44a7f
* src/ios: 35a984f205..4fdd9e6dda
* src/testing: 89d9194c5a..6f09eccc24
* src/third_party: 9bae82ef2a..432c55f861
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/16322a374a..fa0f0f2dd7
* src/third_party/depot_tools: a3a80b6908..544b744621
* src/tools: 0160a47e0e..b3c542c8dc
DEPS diff: 33aa22e76e..7a8a322ad7/DEPS

No update to Clang.

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

Change-Id: I93e31890399309d4abffaa1cda5df6870cb06bac
Reviewed-on: https://webrtc-review.googlesource.com/62466
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22480}
2018-03-17 03:17:12 +00:00
Autoroller
cdd2a97f5a Roll chromium_revision ce851e47bd..33aa22e76e (543685:543816)
Change log: ce851e47bd..33aa22e76e
Full diff: ce851e47bd..33aa22e76e

Changed dependencies:
* src/base: ebff846f9a..fcec2204ef
* src/build: e3927e7faf..7642603cf6
* src/ios: 55282d1a03..35a984f205
* src/testing: ab79f563a8..89d9194c5a
* src/third_party: c0d8139005..9bae82ef2a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6733f34199..16322a374a
* src/tools: a6bb2c2717..0160a47e0e
DEPS diff: ce851e47bd..33aa22e76e/DEPS

Clang version changed 325667:327688
Details: ce851e47bd..33aa22e76e/tools/clang/scripts/update.py

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

Change-Id: I04740d14cad49dda48b532b9279241783e1d301d
Reviewed-on: https://webrtc-review.googlesource.com/62460
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22479}
2018-03-16 21:20:21 +00:00
Sebastian Jansson
317a522876 Fixes to posting delayed process tasks in SSCC.
The task queue based SendSideCongestionController (SSCC) was accessing
a unique pointer to the task queue from the task queue itself. This
triggered a tsan check failure when resetting the same unique pointer.

Also move declaration of SSCC member in RtpTransportControllerSend last,
to ensure that it, and its TaskQueue, are destroyed before other members.

Bug: webrtc:8415
Change-Id: I75c93f41deab637f7e4766ac4b61713c86f866e9
Reviewed-on: https://webrtc-review.googlesource.com/62143
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22478}
2018-03-16 15:28:21 +00:00
Oskar Sundbom
4ccc1c4bb6 Don't destroy a receive stream's sink before reassigning it.
Bug: chromium:820901
Change-Id: If1f2ea82172154c8645baf5fbbba3acf17ddc19b
Reviewed-on: https://webrtc-review.googlesource.com/62346
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Oskar Sundbom <ossu@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22477}
2018-03-16 15:01:52 +00:00
Patrik Höglund
3bb1194fff Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'"
This reverts commit d3070f43b19f503246be4ebad425d87568a71ce0.

Reason for revert: Need to re-enable h264 tests.

Original change's description:
> Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'
> 
> This doesn't change behavior at the moment because Chromium's
> 'proprietary_codecs' is already conditional on 'is_chrome_branded'
> but this guards WebRTC's default from upstream changes like
> https://chromium-review.googlesource.com/c/chromium/src/+/835010/6/build/config/features.gni
> 
> TBR=phoglund@webrtc.org
> 
> Bug: webrtc:8675
> Change-Id: Ic2ae311b5fc70a4d1ac1aefe4cc27574e4fcee40
> Reviewed-on: https://webrtc-review.googlesource.com/36321
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21452}

TBR=phoglund@webrtc.org,oprypin@webrtc.org,hbos@webrtc.org

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

Bug: webrtc:8675
Change-Id: I97e5b25fb638e9d4731ac9610f9f6009a3789578
Reviewed-on: https://webrtc-review.googlesource.com/62380
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22476}
2018-03-16 14:34:21 +00:00
Autoroller
35468356d6 Roll chromium_revision e0e02de5a7..ce851e47bd (543578:543685)
Change log: e0e02de5a7..ce851e47bd
Full diff: e0e02de5a7..ce851e47bd

Changed dependencies:
* src/base: f623956778..ebff846f9a
* src/build: 7757432a81..e3927e7faf
* src/ios: 2bfb2ab258..55282d1a03
* src/testing: 654bce3959..ab79f563a8
* src/third_party: c5d96c8e03..c0d8139005
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/36d08eec1a..6733f34199
* src/tools: b0ce060f3b..a6bb2c2717
DEPS diff: e0e02de5a7..ce851e47bd/DEPS

No update to Clang.

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

Change-Id: I8faad77fb9f732699f16d42f6f9bf927076d8883
Reviewed-on: https://webrtc-review.googlesource.com/62365
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22475}
2018-03-16 14:25:41 +00:00
Niels Möller
bf3dbb4a69 Delete payload_type from VCMEncoderDatabase and vcm::VideoSender.
Bug: webrtc:8830
Change-Id: Ie6a874023618a5540e138b34edfcad1ce6e8d391
Reviewed-on: https://webrtc-review.googlesource.com/62102
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22474}
2018-03-16 13:43:01 +00:00
Niels Möller
5bf8ccdfa0 Delete encoder caching in WebRtcVideoSendStream.
This is a followup to https://webrtc-review.googlesource.com/61640,
which ensures that picture id and tl0 pic idx are continuous,
independent of how the encoder objects are created and destroyed.

The plan is to later move responsibility for encoder creation to
VideoSendStream::ReconfigureVideoEncoder, delegating work to
VideoStreamEncoder.

Bug: webrtc:8830
Change-Id: Idde5c91f24d3c0e3fa6a3bb26eb06f6800896a28
Reviewed-on: https://webrtc-review.googlesource.com/62082
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22473}
2018-03-16 13:20:21 +00:00
Åsa Persson
677f42c679 Enable ContinuousAfterStreamCountChangeSimulcastEncoderAdapter picture id tests.
Support added in: https://webrtc-review.googlesource.com/c/src/+/61640

The tests are no longer related to any field trial.

Bug: none
Change-Id: I42dbdf23fa44953a139177a6693630507152e2ef
Reviewed-on: https://webrtc-review.googlesource.com/62345
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22472}
2018-03-16 13:18:41 +00:00
Cameron Pickett
d132ce1f67 Remove unnecessary copies from AsyncInvoke
Currently, the way the AsyncInvoke is implemented, the lambda invoked is copied multiple times. This causes two problems: (1) a reduced performance where captured variables are copied unnecessarily, (2) lambdas with non-copyable captures are not possible to invoke.

This cl attempts to address both points.

Change-Id: I8d907287d6e4851330d469f184760d165fa8bc08
Bug: webrtc:9028
Reviewed-on: https://webrtc-review.googlesource.com/61346
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22471}
2018-03-16 13:09:18 +00:00