Commit Graph

29482 Commits

Author SHA1 Message Date
6153e15d31 Roll chromium_revision a989226e28..81b1889c8c (731140:731328)
Change log: a989226e28..81b1889c8c
Full diff: a989226e28..81b1889c8c

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/49cfb9bdc2..25614ff33a
* src/third_party/depot_tools: a1266b63b5..59a3b2fd5d
* src/third_party/harfbuzz-ng/src: 64a45be519..82545c5e2b
DEPS diff: a989226e28..81b1889c8c/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I623a817f55cf7afa004d6b4b7f9ab16d7463d3be
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166020
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30250}
2020-01-14 14:35:38 +00:00
ecc5b93b13 AEC3: Restrict default logging of some delay changes to VERBOSE
It leads to overly verbose test output. Example:
https://chromium-swarm.appspot.com/task?id=49bc386e0545ef10

Bug: webrtc:11278
Change-Id: I4a1c565f3aab94d98910722b23dcadc5fcde602a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165962
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30249}
2020-01-14 12:52:47 +00:00
3e66a498c3 Use RTX SSRCs in scenario test framework.
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
2020-01-14 12:04:56 +00:00
2a92d2b461 Cleanup: Prepares for simulated time peer connection tests.
This CL contains some preparatory cleanup that can be done
outside the main CL.

Bug: webrtc:11255
Change-Id: Ib0dcd81d352bafc446dcd2f7f82ba81f5e82e210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165766
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30247}
2020-01-14 09:55:42 +00:00
b580bff520 Roll chromium_revision d6f6958da9..a989226e28 (731013:731140)
Change log: d6f6958da9..a989226e28
Full diff: d6f6958da9..a989226e28

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/71813e2ccf..49cfb9bdc2
DEPS diff: d6f6958da9..a989226e28/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I9b42a9df1f3fba62cb529e3e98d986f0af194994
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165940
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30246}
2020-01-14 08:37:05 +00:00
1546f99572 Fixed timeout overflow in sctp reliability test.
Sometimes some tests failed due to test long execution while timeout
was computed to negative value.

Bug: None
Change-Id: Icb666170323f6b757a409db575d36116f57632d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165691
Commit-Queue: Yura Yaroshevich <yura.yaroshevich@gmail.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30245}
2020-01-14 08:18:25 +00:00
3e3c551ac6 Suppress C5041 constexpr warning for MSVC 2019
Disable the C5041 warning which makes the build fail. This is a
C++17-only change and WebRTC doesn't support C++17 yet, so the code is
technically correct, but fails to build on MSVC 2019 and
warning-as-error active.

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

No-Presubmit: True
Bug: webrtc:11275,webrtc:11276
Change-Id: I891a894ee87252f96e84fd8d282576f46907256f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30244}
2020-01-14 07:44:35 +00:00
2ea27968d3 Extract an interface from the perf results logger.
The new interface is called PerfTestResultWriter and is currently
implemented by PerfResultsLogger (renamed PerfTestGraphJsonWriter).

I plan to introduce a second implementation of the perf logger that
uses the new Histogram C++ API. I add a flag that chooses
between the two implementations so I can try it out (perhaps by
setting up a second, limited run of webrtc_perf_tests on the perf
bots that uses the new implementation). The histogram C++
implementation will come in the next patch.

As a side effect, I disentangled the plottable counter printer from
the perf result printer so it will work for both implementations.
The only thing they had in common was that both wrote JSON anyway.

See the bug for details on the new API.

Bug: chromium:1029452
Change-Id: Icb21b25ced08ea73aeecd221e9d51f2adf3dab1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165389
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30243}
2020-01-14 06:05:02 +00:00
145cfc5025 Roll chromium_revision 69c66e4366..d6f6958da9 (730870:731013)
Change log: 69c66e4366..d6f6958da9
Full diff: 69c66e4366..d6f6958da9

No dependencies changed.
No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I5b2bf79e71bc85ae0fe351a6365303462306e97f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165880
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30242}
2020-01-14 02:37:17 +00:00
2cbfe17129 Roll chromium_revision 0792dc5faa..69c66e4366 (730752:730870)
Change log: 0792dc5faa..69c66e4366
Full diff: 0792dc5faa..69c66e4366

No dependencies changed.
No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I0187c15ad63120d2d88582137017a4b52d7e5f71
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165840
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30241}
2020-01-13 22:44:43 +00:00
76294b6709 Roll chromium_revision 210e790756..0792dc5faa (730612:730752)
Change log: 210e790756..0792dc5faa
Full diff: 210e790756..0792dc5faa

Changed dependency
* src/third_party/depot_tools: 5e96ad12ac..a1266b63b5
DEPS diff: 210e790756..0792dc5faa/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Iab86c192804333fe7b7113224a6e5ce562f166ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165821
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30240}
2020-01-13 20:40:08 +00:00
bb6677709b Roll chromium_revision b581de5b1b..210e790756 (730447:730612)
Change log: b581de5b1b..210e790756
Full diff: b581de5b1b..210e790756

Changed dependencies
* src/buildtools/linux64: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/mac: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/buildtools/win: git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5..git_revision:0c5557d173ce217cea095086a9c9610068123503
* src/third_party/depot_tools: 7a8bf94894..5e96ad12ac
DEPS diff: b581de5b1b..210e790756/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I39e364b2aec00a902f8d665716c36e1fd48385da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165820
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30239}
2020-01-13 18:47:00 +00:00
b8c775aeaf Refactoring AudioSender api out of AudioSendStream for more abstraction to reuse AudioTransportImpl for voip api
Bug: webrtc:11251
Change-Id: Id3b6ff1814931d8250c4aaac59e494521fbe93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164560
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tim Na <natim@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30238}
2020-01-13 18:31:30 +00:00
8234b92ba3 Move DegradationPreference logic out of VideoSourceSinkController.
The DegradationPreference logic is moved into
OveruseFrameDetectorResourceAdaptationModule. This makes the adaptation
module solely responsible for degradation preference, and the
VideoStreamEncoder the only bridge between the adaptation module and the
VideoSourceSinkController.

The adaptation module is now unaware of the existence of a controller.
It only "speaks" VideoSourceRestrictions, which is a big milestone in
making adaptation modules injectable.

A follow-up CL will explore the possibility of reconfiguring the
controller's source and which degradation preference to use to the
encoder queue. This would allow us to make several classes
single-threaded, but it is a change in behavior and should be done in a
separate CL.

Bug: webrtc:11222
Change-Id: Ib7f640e12789da5f801177926c2072a51818f261
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165684
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30237}
2020-01-13 17:24:48 +00:00
e270ff1c41 [iOS] Reset VT session when H264 decoder malfunction error happen
Bug: webrtc:11268
Change-Id: I6932cfbe53dc7b922a90604de799f259526b4c8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165785
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30236}
2020-01-13 14:57:36 +00:00
6ea2c6ae87 Cleanup: Merges Thread and MessageQueue.
Since rtc::Thread is the only class inheriting from rtc::MessageQueue
and most members of MessageQueue are public or protected the split is
not adding much value. In preparation for future cleanup, this cl merges
the two classes.

Bug: webrtc:9883
Change-Id: Ia0efb4349f66f653aa34fa4d244998f187e3ce36
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165340
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30235}
2020-01-13 13:53:20 +00:00
7d43801a07 Delete RtpGenericDepacketizer as no longer used
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
48148dc840 Change log level of AEC3 buffer info to VERBOSE
Otherwise, test logs become very verbose:
https://chrome-swarming.appspot.com/task?id=49b6fa6ac93e2310
See linked issue.

Bug: webrtc:11278
Change-Id: I778ee4826de6c1b23d47a5d5ce302d074900ce6b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165786
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30233}
2020-01-13 13:31:27 +00:00
fa73393574 In TaskQueueWin fix race in canceling MutlimediaTimer
Bug: webrtc:11232
Change-Id: I371f0b78a572c94f2eefd8e0859eed88bce9e37e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165762
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30232}
2020-01-13 13:02:16 +00:00
fae640003c Add saza@ and peah@ to OWNERS of some audio files
Bug: None
Change-Id: Ibab0528b09bf2c4f0af4fd383a7b5e93e6c55f6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165784
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30231}
2020-01-13 12:31:21 +00:00
54891af88d More lenient double comparison for RunningStatistics.FullSimpleTest
The test was failing (in a flaky fashion, interestingly), comparing:
* 50.5 whose mantissa is:
1001010000000000000000000000000000000000000000000000
* with 50.500000000000036 whose mantissa is:
1001010000000000000000000000000000000000000000000101

since EXPECT_DOUBLE_EQ() only allows 4 ULPs difference.
We don't need this kind of precision.

Bug: webrtc:11134
Change-Id: I811178b0762dbcd61d4f2d3f047ea0b59847fa57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165761
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30230}
2020-01-13 12:21:12 +00:00
53cd9e2645 Separates simulated TaskQueue and simulated ProcessThread.
The overlap in functionality is quite limited and separating the
functionality makes it a bit easier to follow each. This prepares
for adding a SimulatedThread class in a follow up CL.

Bug: webrtc:11255
Change-Id: I83c754bd570113dfb582098bb4d39e27bb4f4d87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165688
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30229}
2020-01-13 11:26:42 +00:00
ce0ea49001 VideoStreamEncoder configuring source/sink with VideoSourceController.
This is part of the work for making VideoStreamEncoder responsible for
configuring its source/sink and limiting the responsibility of
OveruseFrameDetectorResourceAdaptationModule to only output relevant
VideoSourceRestrictions.

BEFORE THIS CL

Prior to this CL, OveruseFrameDetector was responsible for performing
AddOrUpdateSink() on the source, which it did using its nested class
VideoSourceProxy.

AddOrUpdateSink() could happen for both adaptation and non-adaptation
related reasons. For example:
- Adaptation related: AdaptUp() or AdaptDown() happens, causing updated
  VideoSourceRestrictions.
- Non-adaptation related: VideoStreamEncoder asks the module to
  reconfigure the source/sink for it, such as with
  SetMaxFramerateAndAlignment() or SetWantsRotationApplied().

AFTER THIS CL

AddOrUpdateSink() is performed by VideoSourceController, which is owned
by VideoStreamEncoder. Any reconfiguration has to go through the
VideoStreamEncoder. This means that:
- Non-adaptation related settings happen between VideoStreamEncoder and
  VideoSourceController directly (without going through the adaptation
  module).
- Adaptation related changes can be expressed in terms of
  VideoSourceRestrictions. OveruseFrameDetectorResourceAdaptationModule
  only has to output the restrictions and not know or care about other
  source/sink settings.

For now, VideoSourceController has to know about DegradationPreference.
In a future CL, the DegradationPreference logic should move back to
the adaptation module. The VideoSourceRestrictions are fully capable of
expressing all possible source/sink values without the "modifier" that
is the degradation preference.

Bug: webrtc:11222
Change-Id: I0f058c4700ca108e2d9f212e38b61f6f728aa419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162802
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Cr-Commit-Position: refs/heads/master@{#30228}
2020-01-13 11:14:04 +00:00
d238200882 Introduce ResourceAdaptationModuleListener and VideoSourceRestrictions.
The VideoSourceRestrictions describe the maximum pixels per frame and
max frame rate of a video source.

This CL makes the ResourceAdaptationModuleInterface responsible for the
reconfiguration of video sources. The VideoSourceRestrictions is the
output of an adaptation module, and the ResourceAdaptationModuleListener
handles the callback for when the source restrictions change.

The OveruseFrameDetectorResourceAdaptationModule is updated to output
its changes using these interfaces, and VideoStreamEncoder - now a
listener - is made responsible for triggering the reconfiguring the
video source.

Performing the reconfiguration still requires interacting with the
VideoSourceProxy - it is still partially responsible for keeping track
of rtc::VideoSinkWants settings and performing AddOrUpdateSink(). For
now this may look a bit weird: the VideoSourceProxy tells the
VideoStreamEncoder about the new restrictions, and then the
VideoStreamEncoder tells the VideoSourceProxy to apply these
restrictions on the source/sink. This exercises the listener though, and
unblocks the next CL.

The next CL should move all "configuring the source" logic to the
VideoStreamEncoder instead, so that the only information that is tracked
by OveruseFrameDetectorResourceAdaptationModule is what it actually
outputs to the listener. See the next CL
(https://webrtc-review.googlesource.com/c/src/+/162802) where a
VideoSourceController is introduced, to be owned by the
VideoStreamEncoder rather than the adaptation module.

Bug: webrtc:11222
Change-Id: I450ce74f51d96c4b98009a06134db671893d8fdc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162522
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30227}
2020-01-13 10:57:00 +00:00
2869638b4d Avoid [[nodiscard]] warning C4834 with MSVC 2019
Avoid a warning-as-error of MSVC 2019 due to a test ignoring a
[[nodiscard]] return value:

C4834: discarding return value of function with 'nodiscard' attribute
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d

Bug: webrtc:11275
Change-Id: I6b70d85769f311814393412830f48d0d8bfef63d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164467
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30226}
2020-01-13 10:07:35 +00:00
f5ecb5f22e Revert "Reland "Reland "Reland "Distinguish between send and receive video codecs""""
This reverts commit 9cad4dccc96f09d76dce3766a076052df7d75ab8.

Reason for revert: Breaks downstream tests.

Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive video codecs"""
> 
> This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c
> 
> This CL lands all code except the code that activates the change,
> see media/engine/webrtc_video_engine.cc
> Once downstream projects are fixed, there will be a one-line change to
> activate the change to distinguish between send and receive video codecs.
> 
> Original change's description:
> > Reland "Reland "Distinguish between send and receive video codecs""
> >
> > This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> > >
> > > Reason for revert: Downstream test updated.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive video codecs""
> > > >
> > > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > > >
> > > > Reason for revert: Breaks another downstream test.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > > >
> > > > > Reason for revert: Downstream tests have been updated.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive video codecs"
> > > > > >
> > > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > > >
> > > > > > Reason for revert: Breaks downstream test.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive video codecs
> > > > > > >
> > > > > > > Even though send and receive codecs are the same,
> > > > > > > they might have different support in HW.
> > > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > > track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30079}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30097}
> >
> > Bug: chromium:1029737
> > Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30120}
> 
> Bug: chromium:1029737
> Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30219}

TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org

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

Bug: chromium:1029737
Change-Id: I377f82866e56862f57383f96a3b96719344eef9e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165780
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30225}
2020-01-13 09:03:37 +00:00
081f7a3498 Add missing header for dchecked_cast on UWP
Add missing #include to fix some build error on `winuwp` with some code
using rtc::dchecked_cast<> under an `#ifdef (WINUWP)`, resulting in an
undefined symbol error.

Bug: webrtc:11194
Change-Id: Iad9e74c3e92ed6cf1461f34cdd9329d13f5d62e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161721
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30224}
2020-01-13 07:53:49 +00:00
41d96d26ee Flip goog_ping_announce default to false
BUG: webrtc:11100
Change-Id: I37a23b32b339c000cc2e88793c31732f7f1d259d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165686
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30223}
2020-01-13 06:38:35 +00:00
5590ec063f Roll chromium_revision d794106d9d..b581de5b1b (730346:730447)
Change log: d794106d9d..b581de5b1b
Full diff: d794106d9d..b581de5b1b

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/32c9791b8a..71813e2ccf
* src/third_party/depot_tools: fc132e61db..7a8bf94894
* src/third_party/sqlite4java: 889660698187baa7c8b0d79f7bf58563125fbd66..LofjKH9dgXIAJhRYCPQlMFywSwxYimrfDeBmaHc-Z5EC
DEPS diff: d794106d9d..b581de5b1b/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I7c06ddf990c474892f71ef81e45d1520b8798e6f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165730
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30222}
2020-01-12 14:38:41 +00:00
9d4bbc216b Using tasks to process packets in FakeNetworkSocketServer.
This way we can rely on existing task scheduling and execution
functionality, reducing the required functionality to support the
fake socket server.

This prepares for support simulated time execution of peer
connection level tests.

Bug: webrtc:11255
Change-Id: I7de64a099c2e355c70929ecff79b8ea3b98b70b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165398
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30221}
2020-01-12 12:53:30 +00:00
10d8758251 Roll chromium_revision bd2395cd43..d794106d9d (730226:730346)
Change log: bd2395cd43..d794106d9d
Full diff: bd2395cd43..d794106d9d

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/f6f813d450..32c9791b8a
* src/third_party/depot_tools: 13928b7e7f..fc132e61db
DEPS diff: bd2395cd43..d794106d9d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I6cc34f75c049bc75a92eddaf00e6dc0694d64837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165669
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30220}
2020-01-11 00:40:03 +00:00
9cad4dccc9 Reland "Reland "Reland "Distinguish between send and receive video codecs"""
This is a reland of 4e64e605894df287178c5a1b537fbe859b7d420c

This CL lands all code except the code that activates the change,
see media/engine/webrtc_video_engine.cc
Once downstream projects are fixed, there will be a one-line change to
activate the change to distinguish between send and receive video codecs.

Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
>
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
>
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are the same,
> > > > > > they might have different support in HW.
> > > > > > Distinguish between send and receive codecs to be able to keep
> > > > > > track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30097}
>
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30120}

Bug: chromium:1029737
Change-Id: Id4f1c6f6f0cf7b96fe93dd22d14310d286af31f3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165682
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30219}
2020-01-10 23:37:11 +00:00
0e3a3f6b1d Adding deadbeef to sctp/OWNERS and removing myself.
Bug: None
Change-Id: I572b65107797da8494f1956ab0a08a3221be4bb7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165002
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30218}
2020-01-10 21:09:39 +00:00
b0e0728159 Replaces SynchronousMethodCall with rtc::Thread::Invoke.
Given that we already have Thread:.Invoke that can be used with lambda,
SynchronousMethodCall doesn't add any value.

This simplification prepares for simulated time peer connection tests.

Bug: webrtc:11255
Change-Id: I478a11f15e30e009dae4a3fee2120f6d7a03355f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165683
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30217}
2020-01-10 19:29:34 +00:00
290de82b2a Cleanup: Replace MessageQueue pointers with Thread pointers.
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.

Bug: webrtc:9883
Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30216}
2020-01-10 19:03:12 +00:00
ec648f50ca Roll chromium_revision 54a7cb4bda..bd2395cd43 (730109:730226)
Change log: 54a7cb4bda..bd2395cd43
Full diff: 54a7cb4bda..bd2395cd43

Changed dependency
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9dbcda8385..f6f813d450
DEPS diff: 54a7cb4bda..bd2395cd43/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I50405f17a60be878e906f03e05605b5581f70578
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165666
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30215}
2020-01-10 18:50:23 +00:00
bcea217667 Unflake P2PTransportChannelTest.TurnToTurnPresumedWritable.
Some messages were processed after involved objects were destructed,
a.k.a. 'use after free'.

This CL fixes that by disconnecting signals before fixture destruction,
honoring CreateChannel/DestroyChannel symmetry and following what is
done in similar test cases.

Bug: webrtc:11269
Change-Id: I122aca70a9978b752edc01e5f31583f4425f3624
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165685
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30214}
2020-01-10 18:35:53 +00:00
b408bb7b95 Revert "In RtpBitrateConfigurator ignore new parameters when set to default values."
This reverts commit bcbdeedd432198c3d48effb2162af6344d885b14.

Reason for revert: Speculative revert after a perf regression.

Original change's description:
> In RtpBitrateConfigurator ignore new parameters when set to default values.
> 
> Bug: webrtc:11263
> Change-Id: Ia7539c7c142b059d0295849b916439bb647f112d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162207
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30191}

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

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

Bug: webrtc:11263
Change-Id: I17804655465b27523c462d2aba44519c820b8e04
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165687
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30213}
2020-01-10 16:39:51 +00:00
b42aeaa3fb Move RtpDepacketizerH264 into own files
Bug: webrtc:11152
Change-Id: Iab4975e9f378b177a2abf34559f9b74752e69843
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165582
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30212}
2020-01-10 15:33:54 +00:00
539f9b376e Use a TaskQueue for decoding in VideoStreamDecoderImpl.
Long term goal is to use the VideoStreamDecoder in the VideoReceiveStream so
that we can stop using legacy VideoCodingModule components and classes. This CL is
one of several in preparation for that.

Bug: webrtc:7408, webrtc:9378
Change-Id: Ifd7e4c3c7d38dbb7c4b0636aaad318c571a29158
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164525
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30211}
2020-01-10 14:31:22 +00:00
27064adc34 SimulcastEncoderAdapter: In passthrough mode set correct lenght for frame_types parameter
If in simulcast case some streams are disabled (especially the first one), the key-frame
requests might be ignorred by e.g. libvpx vp8 encoder wrapper.

Before this CL SimulcastEncoderAdapter always passes single frame type in Encode() call.
However, if underlying encoder used simulcast, it would've expected as many frame types
as there are streams.

Bug: none
Change-Id: I7f56a6540b67273b7d3cf9fa86dc76015b92d271
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165681
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30210}
2020-01-10 14:16:34 +00:00
4b07059139 [iOS] Reset VT session when H264 encoder malfunction error happen
Bug: webrtc:11268
Change-Id: I764eb37a386075838e981c6d5b820e25d77f1a80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165395
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30209}
2020-01-10 12:00:45 +00:00
6ddbe2c5b0 Extract results line plotting.
This will make RESULT lines still come out after we add a second JSON
writer implementation.

Bug: chromium:1029452
Change-Id: I5cba3151c21df2901f19305e9b71bc5c9638a0ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165399
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30208}
2020-01-10 11:29:55 +00:00
56452dd17b Roll chromium_revision aa827d6534..54a7cb4bda (729982:730109)
Change log: aa827d6534..54a7cb4bda
Full diff: aa827d6534..54a7cb4bda

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ce9e11f024..9dbcda8385
* src/third_party/depot_tools: 12f8d69f12..13928b7e7f
DEPS diff: aa827d6534..54a7cb4bda/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ib03d9edc7e52303c9c6c01e566940c05e7f2a010
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165662
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30207}
2020-01-10 10:42:55 +00:00
4b6df1777c Update Linux documentation links
Bug: None
Change-Id: Idbee56e8c6ed25fb90b2456c243e30ef72a0b68d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165642
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30206}
2020-01-10 07:51:46 +00:00
9b289b2c29 Roll chromium_revision d63380b813..aa827d6534 (729869:729982)
Change log: d63380b813..aa827d6534
Full diff: d63380b813..aa827d6534

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ae4bbcda1a..ce9e11f024
* src/third_party/depot_tools: 797d74a266..12f8d69f12
DEPS diff: d63380b813..aa827d6534/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I899224e29d8727fb1a73b6782d1b1e2e3e0e9608
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165641
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30205}
2020-01-10 00:37:44 +00:00
ab81316411 Housekeeping: Declare DataChannelController immovable
This should be done according to the C++ style guide.

Bug: none
Change-Id: I3f8d36339bbc7175bd67631e38820b5883e875d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165386
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30204}
2020-01-09 21:33:54 +00:00
be4f9ed113 Roll chromium_revision b57c714b1d..d63380b813 (729764:729869)
Change log: b57c714b1d..d63380b813
Full diff: b57c714b1d..d63380b813

Changed dependencies
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/85c4a438f6..ae4bbcda1a
* src/third_party/depot_tools: 081c5b5979..797d74a266
DEPS diff: b57c714b1d..d63380b813/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I48bd305729082e7b4ea053a42ec710c1ec28042f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165620
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30203}
2020-01-09 20:42:33 +00:00
9a83dd729b Batch process pending tasks in the libevent TaskQueue
This change improves performance under high load by processing
all pending tasks each time the thread is woken up by libevent.

Additionally, the pipe used to wake up the TaskQueue thread now
not be written to if there's already a pending write on the pipe.
This fixes a bug where under high load the pipe write buffer can
fill and cause tasks to get dropped.

Bug: webrtc:11259, webrtc:8876
Change-Id: Ic82978c71bf9e9a25f281ca4775d46168d161d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165420
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30202}
2020-01-09 19:49:23 +00:00
3b19b27343 Make the sdk:audio_session_objc target public.
Bug: webrtc:11237
Change-Id: I83360b2608a58c7ab9f0cb050aa289df178eb66f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165560
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Joe Chen <jsphchn@google.com>
Cr-Commit-Position: refs/heads/master@{#30201}
2020-01-09 17:35:57 +00:00