Defects are newly detected by the latest clang version.
This CL mutes them.
Rationale:
* They concern third party code we cannot update here.
* They block chromium roll (containing said clang version).
Bug: webrtc:11110
Change-Id: I7abdfee7e42fd8e89d2296f18690fbda449509d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160081
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29860}
This will be immediately useful to guarantee consistent state across
components referencing the pacer, but will be a net benefit overall
imo.
Bug: webrtc:10809
Change-Id: I49630696f757a832ccf2e4c8597193bf087ce53b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159885
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29859}
This reverts commit I5b9d9036aa90eb0c652f6b17ea1162dea0362640
using spin lock (Global lock) for highly used lock may cause deadlock on ios
Bug: None
Change-Id: Ia7594d665bc17717299245b1a6cfcff18f273e77
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160200
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29857}
In system_wrappers, two build targets depended on the Chromium's
//third_party/webrtc_overides folder. While this was acceptable
before, now that the WebRTC component build is landed [1] it can
create a path where parts of WebRTC get statically linked in
Chromium. To avoid this, this CL removes them and fixes the
problem in //third_party/webrtc_overides.
[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1874722
Bug: webrtc:9419
Change-Id: I94c739d15eb974371af8087986cee03794f327dc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159862
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29852}
This patch moves the logic for
- selection of connection to ping
- selection of connection to use
- selection of connection to prune
into own file and puts it behind a new interface called 'IceControllerInterface'.
BUG=webrtc:10647
Change-Id: I10228b3edd361d3200fa4a734d74a319560966c9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158205
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29850}
ADM2 for Windows is based on the CoreAudioUtil class in Chrome.
CoreAudioUtil in Chrome does not use a special string to identify
the Default Communication device but instead a combination of a
string (Default) and a role parameter [1].
When CoreAudioUtil was ported to WebRTC, I accidentally added an
invalid usage of a unique string to identify the default comm device
and it can lead to errors since there are then two different ways to
identify this device. It will also complicate life when we want to
merge changes from Chrome into WebRTC.
This CL removes usage of AudioDeviceName::kDefaultCommunicationsDeviceId
in WebRTC to reduce the risk of errors.
[1] https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.cc?q=core_audio_ut&sq=package:chromium&g=0&l=464
Excluding flaky bot win_x86_msvc_dbg and using Tbr.
Tbr: thaloun@chromium.org
No-Try: True
Bug: webrtc:11107
Change-Id: Ie6687adbe9c3940a217456e4025967f71d86214c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160047
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29848}
A bug currently causes the packet queue to not get any trials enabled
unless an injected key value map is used.
Bug: None
Change-Id: I5c21aa296e8a202a63e81a57c5d13297ad7333bd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160012
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29845}
This is a manual roll of [1]:
"""
Moved from manual deps into //third_party/android_deps:
* Guava-jre
* AutoService
* ErrorProne
It looks like this CL adds other libraries, but they are just those
that already existed within errorprone-ant.jar.
This updates how ErrorProne is invoked to the JDK9+ method of being a
proper javac plugin. This move necessitated moving the above libraries
into android_deps, because the version of Guava that was already in
android_deps was conflicting with our non-android_deps one.
"""
On top of that, errorprone flags have been removed,
since they aren't recognized anymore:
"error: invalid flag: -Xep:ParameterNotNullable:ERROR"
A follow-up CL will re-activate them with proper invokation.
[1] https://chromium-review.googlesource.com/c/chromium/src/+/1885951
Manual chromium roll: Compile using JDK 11.
Bug: webrtc:11102, chromium:693079
Change-Id: I6fdc700e71bcf39efae948d6195c97700c9cb978
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160011
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29842}
Keeps behavior for old periodic processing.
Rounding sleep time reduced chance for small bursts of busy-looping when
time approaches 0.
Also fixes a DCHECK which may trigger if there are rounding errors in
the timing.
Bug: webrtc:10809
Change-Id: Iba8450f906fd6ab3b1da97e04507b16ac6bbde3f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160000
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29841}
It's possible for an device to be counted but getting its name fails, in which case the utility function returns true but would continue from its loop filling the AudioDeviceNames vector, leading to a smaller output than the later code expects.
No-Try: True
Bug: b/144729866
Change-Id: If902cada4ef2911bc24fbec0f169da75ff6e6a83
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29840}
On top of adding unittests for the remixing, the CL
moves the code tested to a separate file in order
to allow it to be tested.
Bug: webrtc:11007
Change-Id: I531736517bbcc715b3c1bf3a4256c42208c5b778
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155740
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29839}
This CL refactors the analog AGC functionality. In particular it:
-Breaks then tight dependency between the analog AGC and the digital
AGC implementation.
-Removes the complicated callback interface for reporting the analog
level and replaces it with an int.
Bug: webrtc:10859
Change-Id: I3572d60ab98edebbcffa25af64cc74c66f9868fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159039
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29838}
This reverts commit acdc22d7845c5dde7c23366110e54e5d26127c85.
Reason for revert: Field trials are not enabled in the same way, will reland after that is fixed.
Original change's description:
> Prepares PacingController for simplified packet queue.
>
> This CL removes references to RoundRobinPacketQueue::QueuedPacket,
> other than the method to release an RtpPacketToSend. It also moves
> both the BeginPop() and FinalizePop() to within a single helper
> method.
>
> A follow-up cleanup of the packet queue will stop exposing the
> QueuedPacket struct and replaces the the pop-methods with a single
> new one that just returns an RtpPacketToSend.
>
> Bug: webrtc:10809
> Change-Id: I5208a93e12e6b56714d483cc12d2a37225ea8e5e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159889
> Commit-Queue: Erik Språng <sprang@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29820}
TBR=sprang@webrtc.org,philipel@webrtc.org
# Not skipping CQ checks because original CL landed > 1 day ago.
Bug: webrtc:10809
Change-Id: I02fccbfbba6b9670b0ce2008e067df3aa9d3c5f2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160010
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29836}
Add tests for different UpdateRect methods as they are no longer trivial
This change will enable providing useful update rects after scaling
is done.
Bug: webrtc:11058
Change-Id: I2311dbbbb5eca5cfaf845306674e6890050f80c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159820
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29835}
The reported audio interruption metrics are too high. If GetAudio
calls start before the first packets are arriving, and the sample rate
of the encoded audio is different from the one used to initialize
NetEq (default 16 kHz), the initial silent period of GetAudio calls
will be reported as an interruption.
Modifying a unit test to trigger the bug, and make sure it won't come
back.
Bug: webrtc:11094, b/144567257
Change-Id: Id540422cb7f35d3bef68b9e7c03c6e7c8bdb8d97
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159980
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29831}
Those are preventive annotations to prepare for incoming android update
(coming with Chromium roll).
Currently the roll is blocked partly because errorprone complains!
Bug: webrtc:11095, chromium:1003532
Change-Id: If4e2879a522e895ce7fb1f2a9ad36d06f98f2a61
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160002
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29830}
This change makes it easier to propagate more information from the sink
to the video adapter, for example alignment requirements.
Bug: None
Change-Id: I536248d59f871c103a18a48615b6c5e61f61697b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159281
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29827}
New tests are:
- AudioDeviceTest.StartStopPlayoutWithRealDevice
- AudioDeviceTest.StartStopRecordingWithRealDevice
(the comments below only affects ADM2 on Windows):
When adding these tests it was found that we could hit the same known issue
as in https://bugs.chromium.org/p/chromium/issues/detail?id=803056 and the
same solution as in Chrome was therefore ported from Chrome to WebRTC.
Hence, this change also adds support for core_audio_utility::WaveFormatWrapper
to support devices that can return a format where only the WAVEFORMATEX parts is
initialized. The old version would only DCHECK for these devices and that could
lead to an unpredictable behavior.
Tbr: minyue
Bug: webrtc:11093
Change-Id: Icb238c5475100f251ce4e55e39a03653da04dbda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159982
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29824}
Some of the field trial default values are changed as well.
Now available bitrate estimation will be decreasing when RTT is more than 3 seconds.
Unless different parameters for the field trial are specified.
Bug: None
Change-Id: Icd1923fc2e2e7766a7f645016c5432a52537145f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158840
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Nikita Zetilov <zetilovn@google.com>
Cr-Commit-Position: refs/heads/master@{#29823}
We now have two downstream users of stun.h, so it appears to be
generally usable. I put this in a new dir networking/, but I'm open to
suggestions here (maybe some things in api/ should move in there).
I checked what our downstream users are actually using, and it's
cricket::ComputeStunCredentialHash
cricket::<constants>
cricket::TurnMessage
cricket::GetStunErrorResponseType
cricket::StunAttribute::CreateAddress
cricket::StunErrorCodeAttribute
cricket::StunByteStringAttribute
StunAttribute::CreateUnknownAttributes
cricket::TurnErrorType
cricket::StunMessage
I reckoned that was pretty much everything in stun.h, so I didn't
bother splitting it up. They don't use every function and constant
in there, but all _types_ of functions and constants, so for the
sake of coherence I don't think it makes sense to split it.
There's some old stuff in there like GTURN which could arguably
be split out, but it should likely go away soon anyway, so I don't
think it's worth the effort.
Steps:
1) land this
2) update downstream to point to the new header and target
3) remove p2p/base:stun_types.
Bug: webrtc:11091
Change-Id: I1f05bf06055475d25601197ec6fefb8d3b55e8e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159923
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29822}
This CL removes references to RoundRobinPacketQueue::QueuedPacket,
other than the method to release an RtpPacketToSend. It also moves
both the BeginPop() and FinalizePop() to within a single helper
method.
A follow-up cleanup of the packet queue will stop exposing the
QueuedPacket struct and replaces the the pop-methods with a single
new one that just returns an RtpPacketToSend.
Bug: webrtc:10809
Change-Id: I5208a93e12e6b56714d483cc12d2a37225ea8e5e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159889
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29820}
This effectively makes WebRTC upload histogram sets instead of Chart
JSON. Histogram sets is the newest format used by Chromium. I'm doing
this because it's nice to use the most modern thing, but mostly because
it's the default for PinPoint. This means I don't have to implement and
support a new read quest for Chart JSON.
This script has to be source side, because we need third_party/catapult
to write correct histograms. This script will be called from recipes.
I also considered generating histogram JSON directly in
test/testsupport/perf_test.cc, which could have avoided this conversion
from Chart JSON to histogram sets, but I can't because there is no C++
API for histogram sets.
Bug: webrtc:11084
Change-Id: If0d2315d2057112b3c2d54a9cfd12e59b5858a18
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159780
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29818}
This is a reland of 53e157d25ce78ba6cd8625b0b655b46f8e1b0a91
The issue has been fixed in
https://chromium-review.googlesource.com/c/chromium/src/+/1917204.
Original change's description:
> Force Chromium deps on the WebRTC component.
>
> This CL adds a visibility check to the rtc_* GN templates in order
> to force Chromium to depend only on publicly visible targets from
> //third_party/webrtc_overrides and not from //third_party/webrtc.
>
> This is required in order to ensure that the Chromium's component
> builds continues to work correctly without introducing direct
> dependency paths on WebRTC that would statically link it in multiple
> shared libraries.
>
> Bug: webrtc:9419
> Change-Id: Ib89f4fc571512f99678ee4f61696b316374346d9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154344
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Dirk Pranke <dpranke@chromium.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29806}
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I7123d1b44ddbc23b11d9fa25aa39aa420359e33d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159922
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29816}
this patch is puts frequently used check into a set of Check-functions.
the behavior of p2p_transport_channel_unittest is almost unchanged,
the minor change is that when waiting for connection between specific
addresses it waits and does not assume that a particular set of
local/remote addresses will be selected first.
the patch also changes a few EXPECT_ to ASSERT_ since the
tests are not useful where the first EXPECT fails.
BUG=webrtc:10647
Change-Id: Iddcc3c88114db80576e9ebc500572a00dbafdd84
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159882
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29812}
