Commit Graph

30845 Commits

Author SHA1 Message Date
c1aaf4cb38 Revert "disallow pairing ICE-TCP with a local ip address"
This reverts commit 712ebbb5b73baf30f11711efdceb6f08248fac38.
There is apparently more usage in the wild than anticipated.

Bug: chromium:1068705
Change-Id: If2f3907e509570d305670206d8d3724413964208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174420
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31163}
2020-05-05 06:59:45 +00:00
dad6a940e1 Add helper frame generator factories for the pc framework tests.
Bug: webrtc:11534
Change-Id: I569fb9e78aa38f0a17f4e4a261dd93c4b8ba9ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174340
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31162}
2020-05-04 18:56:22 +00:00
3c5450e693 Add support for PendingTaskSafetyFlag to ToQueuedTask.
This keeps usage of ToQueuedTask consistent and avoids callers having
to add additional boiler plate when using the safety flag.

From this:

tq->PostTask(ToQueuedTask([safety = my_safety_flag_]() {
  if (!safety->alive())
    return;
  Foo();
});

to this:

tq->PostTask(ToQueuedTask(my_safety_flag_, []() {
  Foo();
});


Bug: none
Change-Id: I205af56a64dd9839eb845321083d533140d614ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174262
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31161}
2020-05-04 18:20:10 +00:00
9e46cf5cc5 Introduce a RunLoop class that supports the TaskQueue interface
on the current thread.

This simplifies writing async tests that use TaskQueue and doesn't
require spinning up a new thread for simple things. The implementation
is currently based on rtc::Thread, which could also be useful in
some circumstances while migrating code over to TQ.

Remove PressEnterToContinue from the test_common files since
it's very specific and only used from one file.

Bug: none
Change-Id: I8b2c6c40809271a109ec17cf7e1120847645d58a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174260
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31160}
2020-05-04 18:10:00 +00:00
d7197080c0 Add unit tests for audio channel send frame transformer delegate.
Bug: webrtc:11380
Change-Id: I58a3983d3f16be8ed6a95ea2b9ce759bc3b3a7b7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174003
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31159}
2020-05-04 16:50:12 +00:00
1b900b1322 Removed unused function EncodedFrame::SetEncodedSize.
Bug: none
Change-Id: I5b4ce351193198c14cf3c336f910eb1d910f034c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31158}
2020-05-04 16:44:12 +00:00
701ccf97c9 Add unit tests for audio receive channel frame transformer delegate.
Bug: webrtc:11380
Change-Id: I4630b75c83886d722e7be64d50a9790c20956ba4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174004
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31157}
2020-05-04 15:44:08 +00:00
d9255b1840 [getStats] Fix DCHECK crash in MergeInfoAboutOutboundRtpSubstreams().
It seems possible that getStats() and merging RTX/FlexFEC substream
stats into media substream stats can race with the creation or
destruction of the media substream that the RTX/FlexFEC substream is
associated with.

In other words, the DCHECK that ensures that there exists a stats object
to merge into is not always valid. Because there is no media stats
object to merge in to, and outbound-rtp stats objects only exists per
media SSRCs, the sensible thing to do is to RTC_LOG and ignore the
substream stats.

Bug: webrtc:11545
Change-Id: I4061d7190da7ab8bd33fa1fd92c9d819f35d76c7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174360
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31156}
2020-05-04 15:25:34 +00:00
455e80271c Define MockTransformableFrame in test/.
The mock is to be used in frame transformer unit tests.

Bug: webrtc:11380
Change-Id: Id3f6ec71712333232873d8de30e3c7392dc7f5e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174002
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31155}
2020-05-04 15:17:54 +00:00
07ed0f4f93 Add more unit tests for sender video with frame transformer.
Bug: webrtc:11380
Change-Id: Iaf499420f3512fd78421e234a646d53f8fc649bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174005
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31154}
2020-05-04 15:04:15 +00:00
a81e9c82fc Wrap WebRTC OBJC API types with RTC_OBJC_TYPE.
This CL introduced 2 new macros that affect the WebRTC OBJC API symbols:

- RTC_OBJC_TYPE_PREFIX:
  Macro used to prepend a prefix to the API types that are exported with
  RTC_OBJC_EXPORT.

  Clients can patch the definition of this macro locally and build
  WebRTC.framework with their own prefix in case symbol clashing is a
  problem.

  This macro must only be defined by changing the value in
  sdk/objc/base/RTCMacros.h  and not on via compiler flag to ensure
  it has a unique value.

- RCT_OBJC_TYPE:
  Macro used internally to reference API types. Declaring an API type
  without using this macro will not include the declared type in the
  set of types that will be affected by the configurable
  RTC_OBJC_TYPE_PREFIX.

Manual changes:
https://webrtc-review.googlesource.com/c/src/+/173781/5..10

The auto-generated changes in PS#5 have been done with:
https://webrtc-review.googlesource.com/c/src/+/174061.

Bug: None
Change-Id: I0d54ca94db764fb3b6cb4365873f79e14cd879b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173781
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31153}
2020-05-04 15:01:26 +00:00
ce1320cc4d Add WaitForPreviouslyPostedTasks to TaskQueueForTest.
Add an utility function to TaskQueueForTest to execute all already
posted tasks on the queue.

Bug: webrtc:11380
Change-Id: I6cf75bc543cfd2dd1c363935134d3f7bd55eec58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174140
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31152}
2020-05-04 13:47:35 +00:00
1148fd5cef Define MockFrameTransformer in test/.
Add MockFrameTransformer to test/, and remove definitions from unit test
files.

Bug: webrtc:11380
Change-Id: Ia709883e8d000852e3f71e7bfb87877072e22aeb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174001
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31151}
2020-05-04 13:45:22 +00:00
8ae18adb66 Remove unneeded dependency on CallStats.
Bug: none
Change-Id: I348ec88b3d978dac9813fb96368570f647e1e785
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174280
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31150}
2020-05-04 13:12:42 +00:00
14a23a32c4 Add field trial to force playout delay
This CL adds the field trial WebRTC-ForcePlayoutDelay with parameters
min_ms and max_ms. If both of these values are set, the playout delay
of any received packet will be overridden by the specified values.

Bug: None
Change-Id: I353282097e3ffa437dfc5affdfdf7780b09474e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174180
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31149}
2020-05-04 09:03:34 +00:00
3745d3fc93 [Adaptation] Use ResourceAdaptationProcessorInterface* instead of impl.
This replaces references to the ResourceAdaptationProcessor with
references to its interface. This would make it possible to have
alternative implementations or inject fake/mock implementations for
testing.

The VideoStreamAdapter is still responsible for constructing the
ResourceAdaptationProcessor, but beyond construction it is agnostic
towards the implementation.

With this CL, I claim https://crbug.com/webrtc/11222 complete.

TBR=ilnik@webrtc.org

Bug: webrtc:11222
Change-Id: I6e7a73bf1d0b5e97bc694f66180a747b27ffb018
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174160
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31148}
2020-04-30 09:16:41 +00:00
722fa4d509 [Adaptation] Misc tests for processor, input provider and restrictions.
This CL adds miscellaneous unit tests for the
ResourceAdaptationProcessor, the VideoSourceRestrictions comparators and
the VideoStreamInputStateProvider.

Bug: webrtc:11172
Change-Id: If95f69644aaf2b43e3b19d5729bedef0b438c77b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174101
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31147}
2020-04-29 15:59:14 +00:00
bb826c9142 Make echo metrics optional
This makes it optional for an echo detector to report metrics through
the getStats interface.

Bug: webrtc:11539
Change-Id: I1fef93b7bf534637b69c16971d38709b3e849a08
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31146}
2020-04-29 14:29:27 +00:00
6a92e0ebba Android: Allow for re-assigning ScopedJavaGlobalRef
Currently, ScopedJavaGlobalRef can only be set at creation and never
changed. This CL makes it possible to re-set these.

Bug: b/153389044
Change-Id: I6be92dae83a9f5f3d87aa44dde226b874f4ca0a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174041
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31145}
2020-04-29 13:36:32 +00:00
b5a013815f Rename done() into condition(), because it is actually condition in TimeController API
Bug: None
Change-Id: Ia3a742d1d2ad1238223f4da7ae843a8d22108ec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174060
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31144}
2020-04-29 10:29:09 +00:00
91aa73255e [Adaptation] Add OnAdaptationApplied(), remove ResourceListenerResponse.
This CL is part of the Call-Level Adaptation Processing design doc:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing

The ResourceListenerResponse was used to make the QualityScaler
not clear QP samples and instead increase its frequency of checking for
QP under certain circumstances, see enum description:
https://webrtc.googlesource.com/src.git/+/c70b1028d47c1aee4892545190cd66e97d09cd55/call/adaptation/resource.h#33

Because the QualityScaler depends on whether and how adaptation
happened it should listen to adaptation happening.

This CL moves the logic that was previously in VideoStreamAdapter closer
to the QualityScaler: QualityScalerResource::OnAdaptationApplied().

This would allow the VideoStreamAdapter to operate on a separate task
queue in the future, with no dependencies on any stream-specific
resources that might operate on other task queues.

Bug: webrtc:11172, webrtc:11521
Change-Id: I07971a8a5fab5715f4ccb7d2c63f1b92bd47170f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173090
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@{#31143}
2020-04-29 09:08:46 +00:00
4381af48b4 Change connection ASSERT into metrics for the PC level framework.
Bug: webrtc:11504
Change-Id: I48b2f44a52b18fd4bb3e75e9ccdcd842ec1faaed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174022
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31142}
2020-04-28 09:28:13 +00:00
3b9fe99285 Add cpu_usage metrics.
Implemented an analogue of the cpu_usage metrics from third_party/webrtc/video/video_analyzer.h for third_party/webrtc/test/pc/e2e/analyzer/video/default_video_quality_analyzer.h

Bug: webrtc:11496
Change-Id: Ifdc9daa3351f1df5db98beb8f7dc7156fc7c2a16
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174020
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31141}
2020-04-28 09:24:30 +00:00
012aa375b1 Asynchronous QualityScaler: Callback-based CheckQpTask.
This CL breaks up the CheckQp() operation into several steps managed
by the inner helper class CheckQpTask, making responding to high or
low QP an asynchronous operation. Why? Reconfiguring the stream in
response to QP overuse will in the future be handled on a separate
task queue. See Call-Level Adaptation Processing for more details:
https://docs.google.com/document/d/1ZyC26yOCknrrcYa839ZWLxD6o6Gig5A3lVTh4E41074/edit?usp=sharing

Instead of "bool AdaptDown()" when high QP is reported,
synchronously returning true or false depending on the result of
adaptation, this CL introduces
  void QualityScalerQpUsageHandlerInterface::OnReportQpUsageHigh(
      rtc::scoped_refptr<QualityScalerQpUsageHandlerCallback>);
Where
  QualityScalerQpUsageHandlerCallback::OnQpUsageHandled(
      bool clear_qp_samples);
Instructs the QualityScaler whether to clear samples before
checking QP the next time or to increase the frequency of checking
(corresponding to AdaptDown's return value prior to this CL).

QualityScaler no longer using AdaptationObserverInterface, this class
is renamed and moved to overuse_frame_detector.h.

The dependency between CheckQpTasks is made explicit with
CheckQpTask::Result and variables like observed_enough_frames_,
adapt_called_ and adapt_failed_ are moved there and given more
descriptive names.

Bug: webrtc:11521
Change-Id: I7faf795aeee5ded18ce75eb1617f88226e337228
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173760
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31140}
2020-04-28 09:00:15 +00:00
9abc6bd8aa Reduce audiosendstream dependency on rttstats (dead code).
Change-Id: I4b05321548b6584424f23c45b0e95b4c03fe67c1
Bug: none
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148529
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31139}
2020-04-27 13:59:45 +00:00
b37e59d198 Add unittests for APM with submodule creation disabled
This introduces a function AudioProcessingImpl::SetCreateOptionalSubmodulesForTesting to simulate the exclusion of build-optional submodules, and tests of the currently only excludable submodule.

Bug: webrtc:11292
Change-Id: If492606205c9fdc669a6dce3a8989a434aeeed1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173746
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31138}
2020-04-27 11:47:15 +00:00
62a0d647d9 Remove deprecated constant.
Bug: None
Change-Id: I45957ad5e0f5fe0fd129bbae7aaef40a23142374
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173980
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31137}
2020-04-27 10:32:45 +00:00
74fc574cbc Fork a few VideoReceiveStream related classes.
We'll need to deprecate the previous classes due to being used externally
as an API.

Bug: webrtc:11489
Change-Id: I64de29c8adae304d0b7628e24dd0abc5be6387ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173960
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31136}
2020-04-27 09:25:47 +00:00
b261118156 Fix a typo for decoder naming
Bug: None
Change-Id: I1e1e7fe1d3efb6e7f302d7633673418b5de7212c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173940
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31135}
2020-04-27 08:03:47 +00:00
f7f7cc93aa Extend IceControllerFactoryArgs with field trial string
This patch adds a field trial string for the IceController
to the factory interface, the string is from the
"WebRTC-IceControllerFieldTrials" key.

This makes it possible to add new field trials
using that key as needed.

Bug: chromium:1024965
Change-Id: I50498e45da3c49b8e1d620c90c674eedc15dc16e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173900
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31134}
2020-04-27 07:00:04 +00:00
cc73ed3e70 APM: Add build flag to allow building WebRTC without APM
This CL adds a build flag to allow building the non-test parts
of WebRTC without the audio processing module.
The CL also ensures that the WebRTC code correctly handles
the case when no APM is available.

Bug: webrtc:5298
Change-Id: I5c8b5d1f7115e5cce2af4c2b5ff701fa1c54e49e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171509
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31133}
2020-04-26 23:06:44 +00:00
86bd33a1e7 Fix the name of the file generated by generate_sslroots.py.
WebRTC filenames use underscores to separate words so the ssl roots
file is rtc_base/ssl_roots.h instead of rtc_base/sslroots.h.

Bug: chromium:978779
Change-Id: I2fa11c38a566e177775deb3d42230d956efc8ccc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173800
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31132}
2020-04-25 13:37:30 +00:00
95d9a1a3d7 Update set of known root certificates.
This has been automatically generated by running [1].

See https://codereview.webrtc.org/1503473002 for some background about
the generator script.

[1] - https://cs.chromium.org/chromium/src/third_party/webrtc/tools_webrtc/sslroots/generate_sslroots.py

Bug: chromium:978779
Change-Id: I78cf8947b3363738dd0e21182348253dbad95f02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173821
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31131}
2020-04-24 20:40:45 +00:00
1b8ef63876 Add an optional override for AudioRecord device
This is important when we have multiple named devices connected over
USB (eg. "Webcam", "Microphone", "Headset") and there is some way to
choose a specific input device to route from.

Bug: b/154440591
Change-Id: I8dc1801a5e4db7f7bb439e855d43897c1f7d8bc4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173748
Commit-Queue: Robin Lee <rgl@google.com>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31130}
2020-04-24 17:24:54 +00:00
c8660b1650 Open visibility of some PC level framework components
Bug: webrtc:11479
Change-Id: I10567f2766e30825b4d28133002e04dcd0afba21
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173901
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31129}
2020-04-24 16:27:44 +00:00
3e1ac54407 Refactor video dumping and rendering in PC level test.
Move creation of video sinks for dumping and showing rendered video on
screen into video quality analyzer injection helper to eliminate need
to search for video config in on track callback, which makes this more
reliable and reusable.

Bug: webrtc:11479
Change-Id: I6bb5409688fd39268f9f97bde4c9b0833a64396b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173820
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31128}
2020-04-24 09:59:50 +00:00
90ecee1ed9 Make AudioEncoder::GetFrameLengthRange() pure virtual.
In order for WebRTC to be able to include packet overhead in its
bitrate calculations, the AudioEncoder::GetFrameLengthRange()
function must be implemented by all audio encoders. Making this
member function pure virtual as per the following PSA:

https://groups.google.com/forum/#!topic/discuss-webrtc/qscwYr38je0

Bug: webrtc:11427
Change-Id: I30d297ef05f57453bfc257624729559057cad118
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171517
Commit-Queue: Ali Tofigh <alito@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31127}
2020-04-24 09:22:57 +00:00
cda577fd59 Enable simulcast statistics
Bug: webrtc:9547
Change-Id: I8b2920dacfac0085449a797f2831b86e2e5e65b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173749
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31126}
2020-04-24 08:32:13 +00:00
1fb4a05e9e Reland "Launch external ref control for vp9 encoder"
This reverts commit 9665b7d1017bc5b44ffe550c4625921d0315df90.

Reason for revert: Fixes are in the PS#2

Original change's description:
> Revert "Launch external ref control for vp9 encoder"
> 
> This reverts commit 9427b51d6ff50af73c217cb725b1c59b9d701796.
> 
> Reason for revert: Breaks downstream tests
> 
> Original change's description:
> > Launch external ref control for vp9 encoder
> > 
> > Change field trial condition to killswitch instead.
> > 
> > Finch trial is going to 100% public today.
> > 
> > Bug: chromium:1027108,webrtc:11319
> > Change-Id: I29494a7c8515a454706983dd15ae444d3f85271f
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173752
> > Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> > Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#31122}
> 
> TBR=ilnik@webrtc.org,ssilkin@webrtc.org
> 
> Change-Id: I44436febb2b646cdd350fa9afee1c3a7ea307d04
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1027108, webrtc:11319
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173761
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31123}

TBR=ilnik@webrtc.org,ssilkin@webrtc.org

Change-Id: I8aed0edca2015297da512aa084515812103c6f48
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1027108, webrtc:11319
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173780
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31125}
2020-04-23 13:21:45 +00:00
e110a44628 Delete uri for the Generic Frame Descriptor v1
Bug: webrtc:11358
Change-Id: I0c3c3a7f682f172b92dcdcbc6c13d353e1e48ada
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173747
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31124}
2020-04-23 12:44:03 +00:00
9665b7d101 Revert "Launch external ref control for vp9 encoder"
This reverts commit 9427b51d6ff50af73c217cb725b1c59b9d701796.

Reason for revert: Breaks downstream tests

Original change's description:
> Launch external ref control for vp9 encoder
> 
> Change field trial condition to killswitch instead.
> 
> Finch trial is going to 100% public today.
> 
> Bug: chromium:1027108,webrtc:11319
> Change-Id: I29494a7c8515a454706983dd15ae444d3f85271f
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173752
> Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31122}

TBR=ilnik@webrtc.org,ssilkin@webrtc.org

Change-Id: I44436febb2b646cdd350fa9afee1c3a7ea307d04
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1027108, webrtc:11319
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173761
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31123}
2020-04-23 09:25:19 +00:00
9427b51d6f Launch external ref control for vp9 encoder
Change field trial condition to killswitch instead.

Finch trial is going to 100% public today.

Bug: chromium:1027108,webrtc:11319
Change-Id: I29494a7c8515a454706983dd15ae444d3f85271f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173752
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31122}
2020-04-23 09:03:06 +00:00
f4ee036871 [InsertableStreams] Clear callback to audio receive channel in delegate.
Reset the frame transformer delegate's callback to
ChannelReceive::OnReceivedPayloadData when the channel is destroyed, to
prevent future callbacks from the delegate.

Bug: chromium:870644
Change-Id: Iaa2c1b7b26dc38709d3fe64a180ccc6a60a1ec9c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173751
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31121}
2020-04-22 18:58:35 +00:00
1120f7d895 Makes the max padding factor default 3.0x
Bug: webrtc:11508
Change-Id: Ib2e8f5216140f55a590f43fc52a50058703cc2bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173750
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31120}
2020-04-22 10:57:21 +00:00
9ce77fda75 Remove redundant Dcheck in vp9 decoder
Bug: chromium:1070146
Change-Id: Ia4a07cfd16c154e2be3478c020c01fbcaf1c5bb0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173743
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31119}
2020-04-22 09:06:26 +00:00
f355e1a0f6 Add glUseProgram to the list of GL functions requiring synchronization
We have observed an internal deadlock in libGLESv2_adreno where one
thread is in eglCreateContext and another thread in glUseProgram. We
have observed similar deadlocks before and started to synchronize all
access to the offending GL functions. Calls to eglCreateContext are
already synchronized, and this CL synchronizes calls to glUseProgram as
well.

Bug: b/153513005
Change-Id: I576e564aab44c9e429f2b1407105ed72942c309e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173742
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31118}
2020-04-22 07:09:15 +00:00
11f92bc81b Audio ingress implementation for voip api.
This is based on channel_receive.cc implementation where non-audio
related logics are trimmed off for smaller footprint in size.

Bug: webrtc:11251
Change-Id: I743c9f93f509fa6fcc12981fa73a6f01ce38348c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172821
Commit-Queue: Tim Na <natim@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31117}
2020-04-21 20:19:37 +00:00
f2b06ce5c8 Reduces unnecessary thread wakeups in TaskQueuePacedSender.
This CL is functionally a noop but may reduce thread wakupes in some
cases.

In particular, consider a send task scheduled for time T. While waiting
for that, a higher-priority packet than the top of the current queue is
added (e.g. an audio packet), and a send is executed immediately.

After sending, it resets the field indicating that a scheduled task is
expected at time T. It then polls NextSendTime() and schedules a new
task, likely at or very close to T. Causing unnecessary task queue
churn and behavior that is more difficult to reason about.

Bug: webrtc:10809
Change-Id: Ic5706f2cc06df3f27cc3e7b473d4de29a669473b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173700
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31116}
2020-04-21 17:09:37 +00:00
811efc06df Trigger bots.
TBR=peah@webrtc.org

No-Try: True
Bug: None
Change-Id: I3510de1bcb1a66ec5710bb42e27ac17216fb3579
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173744
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31115}
2020-04-21 13:15:09 +00:00
146137adee Merge OOURA BUILD.gn files.
No-Tree-Checks: True
No-Try: True
Bug: webrtc:11509
Change-Id: If66483a02085396e231986e39ef0dafcdd3f4bed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173740
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31114}
2020-04-21 09:06:37 +00:00