This CL introduces a new behavior for leveraging early information
about the delay that is acquired before the standard delay estimate
has been established.
To simplify the process of setting the parameters for that, the CL
also surfaces the delay estimator parameters to the config struct.
Bug: webrtc:9720,chromium: 880686
Change-Id: If886813f70cd805bd37752c63913d28398f1c6fe
Reviewed-on: https://webrtc-review.googlesource.com/97860
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24614}
This CL replaces std::o?stringstream with rtc::StringBuilder where that's possible to do without changing any of the surrounding code. It also updates includes and build files as appropriate.
The CL was generated by running 'git grep -l -P std::o?stringstream | xargs perl -pi -e "s/std::o?stringstream/rtc::StringBuilder/g"'. Then I've manually updated the #includes and BUILD files, run 'git cl format' and unstaged any file that would need more complex fixes.
Bug: webrtc:8982
Change-Id: Ibc32153f4a3fd177e260b6ad05ce393972549357
Reviewed-on: https://webrtc-review.googlesource.com/98460
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24605}
This CL enables -Wexit-time-destructors and -Wglobal-constructors on
rtc_static_library and rtc_source_set build targets.
It also adds the possibility to suppress these warnings because
they trigger in a few places.
The long term goal is to avoid regressions on this and remove all the
suppressions.
Bug: webrtc:9693
Change-Id: I4c1ecc137ef9e87ec5e66981ce95d96fb082727c
Reviewed-on: https://webrtc-review.googlesource.com/98380
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24604}
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
Bug: webrtc:3380
Change-Id: I39c6b35d24182475b33a7a321cdf3b3ac9b8979a
Reviewed-on: https://webrtc-review.googlesource.com/97861
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24602}
This restricts the maximum burst length to receive times 100 ms apart.
Currently packets will be considered part of the same burst as long as
they are received within 5 ms from the previous packet. This can happen
when recovering from large network buffers.
Bug: webrtc:9718
Change-Id: I73027ddaae922cb7bb9a477cf29b4f0036ce6966
Reviewed-on: https://webrtc-review.googlesource.com/98280
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24601}
This is a reland of 711a31aead9007e42dd73c302c8ec40f9e931619
Changes since original landing:
Rename methods only used by tests, mainly via FakeClock,
MessageQueueManager::ProcessAllMessageQueues
--> ProcessAllMessageQueuesForTesting
MessageQueue::IsProcessingMessages
--> IsProcessingMessagesForTesting
Fix the handling of null rtc::Thread::Current() in
ProcessAllMessageQueuesInternal().
Add override Thread::IsProcessingMessagesForTesting() to return false
for the wrapped main thread, unless it's also the current thread. In
tests, the main thread is typically not processing any messages,
but blocked in an Event::Wait().
Original change's description:
> Explicitly wrap main thread in test_main.cc.
>
> Bug: webrtc:9714
> Change-Id: I6ee234f9a0b88b3656a683f2455c3e4b2acf0d54
> Reviewed-on: https://webrtc-review.googlesource.com/97683
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24560}
Bug: webrtc:9714
Change-Id: I6f022d46aaf1e28f86f09f2d68c1803b69770126
Reviewed-on: https://webrtc-review.googlesource.com/98060
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24596}
This allows VP9 encoder wrapper to control frame rate of each spatial
layer. The wrapper configures encoder to skip encoding spatial layer
when actual frame rate exceeds the target frame rate of that layer.
Target frame rate of high spatial layer is expected to be equal or
higher then that of low spatial layer. For now frame rate controller
is only enabled in screen sharing mode.
Added unit test which configures encoder to produce 3 spatial layers
with frame rates 10, 20 and 30fps and verifies that absolute delta of
final and target rate doesn't exceed 10%.
Bug: webrtc:9682
Change-Id: I7a7833f63927dd475e7b42d43e4d29061613e64e
Reviewed-on: https://webrtc-review.googlesource.com/96640
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24593}
Fixes issues when using the display arrangement utility. The region
outside of the screens had random data when re-arranging displays
while capturing. This CL makes sure this region is always black.
Bug: webrtc:9703
Change-Id: I1481dd0f1b4584e75926755f9b8a6e5161cd5904
Reviewed-on: https://webrtc-review.googlesource.com/97184
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24589}
As downside it disallows to destroy RtcpTransceiver on the TaskQueue
without prio call to the Stop function
BUG: webrtc:8239
Change-Id: I236b9aff7a0746044dd764c619174cc5ac03d27f
Reviewed-on: https://webrtc-review.googlesource.com/98120
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24587}
This feature went to stable with M69. Switch is in M69 and M70 banches.
Since tot is now M71 and we have not seen any issues, let's clean this
up.
Bug: webrtc:9634
Change-Id: I708bab55b0443d0873b09dd5b71cdfad72397a7a
Reviewed-on: https://webrtc-review.googlesource.com/98002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24581}
Second round of the new Windows ADM is now ready for review. Main
changes are:
Supports internal (automatic) restart of audio streams when an active
audio stream disconnects (happens when a device is removed).
Adds support for IAudioClient3 and IAudioClient2 for platforms which
supports it (>Win8 and >Win10).
Modifies the threading model to support restart "from the inside" on
the native audio thread.
Adds two new test methods for the ADM to emulate restart events or
stream-switch events.
Adds two new test methods to support rate conversion to ensure that
audio can be tested in loopback even if devices runs at different
sample rates.
Added initial components for low-latency support. Verified that it works
but disabled it with a flag for now.
Bug: webrtc:9265
Change-Id: Ia8e577daabea6b433f2c2eabab4e46ce8added6a
Reviewed-on: https://webrtc-review.googlesource.com/86020
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24578}
This allow to destroy the RtcpTransceiverImpl off the task queue
with assumption it is destroyed after task queue.
i.e. it allows to post destruction of RtcpTransceiverImpl to the TaskQueue without waiting.
BUG: webrtc:8239
Change-Id: I4bea7a6d2edc97061ebd00f2f275c1ab827bc3c5
Reviewed-on: https://webrtc-review.googlesource.com/97160
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24574}
In-lining GetNetworkParameters and MaybeUpdateCongestionWindow which
was left over from previous refactoring. This prepares for upcoming CLs
changing the behavior.
Bug: webrtc:9586
Change-Id: I6f038acdf97c3db2c85254a36592c617a5754a96
Reviewed-on: https://webrtc-review.googlesource.com/97605
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24570}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I443c0c77cef30a3f5712f72ce88db80d1fb29874
Reviewed-on: https://webrtc-review.googlesource.com/96642
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24567}
This is useful for debugging and has minimal effect on production code.
Bug: webrtc:9510
Change-Id: I3a71f484a0d4e54999e376b7924b73230d86cd96
Reviewed-on: https://webrtc-review.googlesource.com/97607
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24557}
Negative infinity is useful in comparisons as it represents a value
guaranteed to be less than any other timestamp. This avoid using
Timestamp::ms(0) to represent such a value or having to use optional
for the special case of first time execution.
To avoid ambiguity, Timestamp::Infinity is renamed to PlusInfinity.
Bug: webrtc:9709
Change-Id: Ib2debff91036e94d1dc903ec0c6faae30febde4e
Reviewed-on: https://webrtc-review.googlesource.com/79440
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24556}
so that it can be shared between different packetizers
and thus easier to extend
Bug: webrtc:9680
Change-Id: Ie5e904ad27afb8dd2ed35ef9e009f7f408017b2f
Reviewed-on: https://webrtc-review.googlesource.com/97661
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24555}
This reverts commit cd87e014f34069fd5a73c1ed5b74ddf251a95c2d.
Reason for revert: Somehow introduces a race where rtc::Thread auto-wrapping may be applied to the TestAudioDevice thread rather than the main thread. This causes failures when running video_engine_tests without any test filter.
Original change's description:
> Reland "Refactor TestAudioDeviceModule to not depend on EventTimerWrapper."
>
> This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
>
> Original change's description:
> > Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
> >
> > In addition, let the processing thread loop explicitly, and not use
> > the deprecated builtin looping in PlatformThread.
> >
> > Bug: webrtc:3380
> > Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> > Reviewed-on: https://webrtc-review.googlesource.com/96544
> > Commit-Queue: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> > Reviewed-by: Artem Titov <titovartem@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#24492}
>
> Bug: webrtc:3380
> Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
> Reviewed-on: https://webrtc-review.googlesource.com/97320
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24528}
TBR=henrika@webrtc.org,nisse@webrtc.org,titovartem@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
# Skip anyway, needed for a two-step revert.
No-try: True
Bug: webrtc:3380
Change-Id: Ia7c8cfab36b8259f150b5ccd0c28defd0e7237f6
Reviewed-on: https://webrtc-review.googlesource.com/97682
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24552}
It is called right after construction, so move the needed
implementation into the MediaOptimization constructor instead.
Bug: webrtc:9711
Change-Id: Ibca35670bf45a85538c34c8ead58ba855acc6b96
Reviewed-on: https://webrtc-review.googlesource.com/97325
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24540}
This was done by
* adding an EchoMetric class to EvaluationScore
* passing an echo metric binary path from the cmd arguments to the
EvaluationScoreWorkerFactory
* passing the render input filepath to the Evaluator.
The echo score is supposed to be computed by the provided binary. It
should print the echo score in [0.0, 1.0] to a text file. It should
satisfy the cmd flags in its invocation in EchoMetric._Run()
Bug: webrtc:7494
Change-Id: I397013d6ed17659ea01d0623d98a14d4fcdcc161
Reviewed-on: https://webrtc-review.googlesource.com/97022
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24537}
Most of the code in neteq_rtpplay is moved into a factory class for
NetEqTest. The factory method takes the same argc and argv arguments as
neteq_rtpplay.
This CL also adds a small public API for neteq_test to allow easy
integration into external software.
Bug: webrtc:9667
Change-Id: I5241c1f51736cb6fbe47b0ad25f4bc83dabd727d
Reviewed-on: https://webrtc-review.googlesource.com/96100
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24531}
This is a reland of 9ea5765f78ed3d0d7b0d483e81f08fb8a2e1110a
Original change's description:
> Refactor TestAudioDeviceModule to not depend on EventTimerWrapper.
>
> In addition, let the processing thread loop explicitly, and not use
> the deprecated builtin looping in PlatformThread.
>
> Bug: webrtc:3380
> Change-Id: I5171ce3457b80f922c8284259882da63c8f146f1
> Reviewed-on: https://webrtc-review.googlesource.com/96544
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24492}
Bug: webrtc:3380
Change-Id: I671e3a60ace6ade765a8537b7e20e36f1782a60d
Reviewed-on: https://webrtc-review.googlesource.com/97320
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24528}
This CL removes //build/config/clang:find_bad_constructs from the
suppressed_configs list, which means that clang:find_bad_constructs
is now enabled on these translation units.
Bug: webrtc:9251, webrtc:163
Change-Id: I71350feb45c23d5a0e17e4828c514ce1c353f6a7
Reviewed-on: https://webrtc-review.googlesource.com/96700
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24524}
Thanks to the CL https://webrtc-review.googlesource.com/c/src/+/83822
there is now no need to destroy the stream asynchronously.
But the CL above introduced a leak, the streams were stopped but
not destroyed. This CL essentially fixes the leak, it is now safe
to destroy the stream right after being stopped as everything happen
in the same capture thread.
Bug: chromium:851883
Change-Id: I4bf7409246f3957d90040d0d8cf09e98f28d6559
Reviewed-on: https://webrtc-review.googlesource.com/96621
Reviewed-by: Brave Yao <braveyao@webrtc.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24519}
Sine a starting rate field was added to the constraints struct. Having
it in the initial config separately is reduntant. To simplify the code,
the extra field is removed. This is a follow up on:
https://webrtc-review.googlesource.com/c/src/+/92624
Bug: webrtc:9586
Change-Id: I9b01b16b2fc4b8479e83b7e998308be2295e0325
Reviewed-on: https://webrtc-review.googlesource.com/96801
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24514}
LappedTransform is only used in BandwidthAdaptationTest and therefore it
should not be anymore a visible target under common_audio.
This CL moves LappedTransform and other two classes it depends on (and which
are not used elsewhere) to modules/audio_coding/codecs/opus/test.
Bug: webrtc:9577, webrtc:5298
Change-Id: I1aa8052c2df2b2b150c279c0c9b1001474aed47a
Reviewed-on: https://webrtc-review.googlesource.com/96440
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24509}
Remove partition support for test helper and from tests.
Merge Init function into constructor
Replace extra macroses in favor of Bit helper function
Replace extra members in favor of local variables
Remove fixture
Bug: None
Change-Id: Ibf1600dda9f59abe5afd2bbe40c3e232a2d269ea
Reviewed-on: https://webrtc-review.googlesource.com/96940
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24508}
This is a reland of da0898dfae3b0a013ca8ad3828e9adfdc749748d
Original change's description:
> Add spatial index to EncodedImage.
>
> Replaces the VP8 simulcast index and VP9 spatial index formely part of
> CodecSpecificInfo.
>
> Bug: webrtc:9378
> Change-Id: I80eafd63fbdee0a25864338196a690628b4bd3d2
> Reviewed-on: https://webrtc-review.googlesource.com/83161
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#24485}
Tbr: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: Iff20b656581ef63317e073833d1a326f7118fdfd
Reviewed-on: https://webrtc-review.googlesource.com/96780
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24507}
This CL introduces the ability to control the usage of the shadow filter
output in the echo canceller output.
Bug: webrtc:9694,chromium:879451
Change-Id: I01f90de60de1799b32892051c176bda5e1a8d33e
Reviewed-on: https://webrtc-review.googlesource.com/97020
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24506}
The intelligibility enhancer is always disabled and it is the only non-test
target using the lapped transform in common_audio (which we planned to remove).
Bug: webrtc:9689, webrtc:5298
Change-Id: Ida65d3aa11ac366471e7e5cbc053108b376c67d8
Reviewed-on: https://webrtc-review.googlesource.com/96460
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24504}
