Commit Graph

29161 Commits

Author SHA1 Message Date
c4f865413a Add TimeController to api/test/ and add a CreateTimeController API.
Creates an abstraction for an "alarm clock" which can schedule
time-controller callbacks and exposes a time controller driven by
an external alarm.

Bug: webrtc:9719
Change-Id: I08c2aa9dba25603043bfba48f55c925716a55bae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158969
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29879}
2019-11-22 17:07:23 +00:00
3daedb6c88 Making the Analog AGC properly support multi-channel
This CL adds proper multi-channel support to the analog AGC.

Beyond that, it prepares adding multi-channel support to the digital
AGC by removing the tight dependency between the analog and digital
AGC codes.

Bug: webrtc:10859
Change-Id: I4414ccbc3db5dbb5ae069fdf426cbd038375ca7b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159480
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29878}
2019-11-22 15:32:55 +00:00
5b82ba37cc Adding VoIP specific channel adjustments
This CL introduces a channel remapping for the mono input case that
is more tailored to the VoIP usecase.

The CL contains a kill-switch that can be used to fall back to the old
mapping behavior in case a need for this is perceived.

Bug: chromium:1027117
Change-Id: Idaaba6eac952e6436beaaf5a1a697cfab8f63286
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160280
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29877}
2019-11-22 15:25:05 +00:00
6031716473 Move IceFieldTrials into own .h-file
Several patches for webrtc:10647 has split the
P2PTransportChannel class/file. This has had the
side effect of it being hard to share the IceFieldTrials-struct.

This patch moves that struct into own file so that can be included
from other components. This patch is a behavioral NOP.

BUG=webrtc:10647

Change-Id: If49cd4d919684a48dde3188a26baf20e4ff2cd8e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160301
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29876}
2019-11-22 15:01:15 +00:00
2148e9a931 Make webrtc_fuzzer_main depend on webrtc_component in Chromium.
This is needed in order to land [1] and restrict visibility of some
//third_party/webrtc_overrides targets.

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1930801

Bug: chromium:896154
Change-Id: Ie71c44ee9a0203a85d77a1199acdcb8581dfb71b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160308
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29875}
2019-11-22 14:31:00 +00:00
af070d0299 Improves device enumeration in ADM2 for Windows.
Summary of changes/improvements and fixes:

Changes container for list of devices from std::vector to std:deque to
allow fast insertion and deletion at both its beginning and its end. This
approach makes it easier to first build a list of all available devices
and then check the size of the list. If size > 0 => two more devices are
added at the front (Default and Default Communication). The old solution
contained a risk of adding invalid Default and Default Communication
devices in cases where not physical device could be found.

Adds usage of |device_index_| in CoreAudioBase to ensure that the selected
device is unique. The previous version used only an ID but that ID is not
unique when e.g. only one device exists since it can have up to three
different roles.

Improves logging and comments.

No-Try: True
Tbr: thaloun@chromium.org
Bug: webrtc:11107
Change-Id: I9a09f7716ed8d8858dcc6a5354b038fc06496166
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160050
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29874}
2019-11-22 14:27:10 +00:00
e43b531266 Nuke p2p/base/stun.h
All downstream users have been moved to the new one.

Bug: webrtc:11091
Change-Id: Ia18d0df94a7b95b1a58b4a53cfb195c61ef59ffd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160201
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29873}
2019-11-22 14:02:10 +00:00
9281436650 Add field trial to cap trendline slope in delay-based BWE.
Bug: webrtc:10932
Change-Id: I34a36a8cad16d65143eff9c675ee98bdbf176ace
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160014
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29872}
2019-11-22 13:14:53 +00:00
efbda8d90a Don't perform DataCallback if the input object has been stopped.
Fix signed/unsigned mismatch.


Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results.

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.

Bug: b/144382120
Change-Id: Iab008c28f03023c830011d229b1f1c7e3e7bb5ee
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160226
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29871}
2019-11-22 10:18:39 +00:00
15f2ff4610 VideoRtpReceiver: factor out VideoRtpTrackSource.
This change factors out VideoRtpTrackSource in preparation
of building the class out.

Bug: chromium:1013590
Change-Id: I015e285b9fcc10b39428dea9f74e0e8648385f62
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159925
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29870}
2019-11-22 10:13:59 +00:00
4995f872ca Cleans up the round robin packet queue.
Usage of this class has now been simplified so that we can do some
cleanup:

* Removes dead code: Push() with 9 args, CancelPop()
* Replaces BeginPop()/CancelPop() with a single Pop() method
* Makes QueuePacket a private class
* Replaces rtp_packets_ with direct ownership from QueuePacket

Bug: webrtc:10809
Change-Id: Iea131ee87d5d920360c71fb180b2af0ea4fc6c7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160007
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29869}
2019-11-22 08:07:15 +00:00
0174ffe700 Can list UWP applications
Before the CL [1] https://webrtc-review.googlesource.com/c/src/+/144960
the UWP applications could not be captured so they were filtered out.

Another reason of this previous filter was because otherwise some
'ghost windows' are listed too. These 'ghost windows' are prelaunched
UWP apps whose windows are created in a hidden/cloaked state to improve
perceived performance of launching these apps later, see:
[2] https://docs.microsoft.com/en-us/windows/uwp/launch-resume/handle-app-prelaunch

They can be filtered out using the new API merged recently
'webrtc::WindowCaptureHelperWin::IsWindowCloaked, see:
[3] https://webrtc-review.googlesource.com/c/src/+/143980

This patch allows non-cloaked UWP apps to be listed by taking advantage
of CL [3]. So that user can select them with the app window picker and can
then share them thanks to [1].

Bug: chromium:700037
Change-Id: I4b41bb764ebbd6e2f164f036a63a4b1cd06c3f2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160021
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Commit-Queue: Julien Isorce <julien.isorce@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29868}
2019-11-21 23:35:09 +00:00
b0df593e56 Reland "Prepares PacingController for simplified packet queue."
This is a reland of acdc22d7845c5dde7c23366110e54e5d26127c85

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=philipel@webrtc.org

Bug: webrtc:10809
Change-Id: Id8196d9348d7fa69a5e410367b8a88e6039ef1b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160205
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29867}
2019-11-21 17:19:10 +00:00
6e9d0d38ef Make base classes for native video encoder/decoder public
Implementers of Java wrappers for native encoders need to have the same
implementation of all the unsupported methods, as mentioned in the
documentation of VideoEncoder.createNativeVideoEncoder (and its decoder
equivalent).

This simplifies implementation of such encoders/decoders, and also make sure
they don’t override unsupported methods, as they are guaranteed not to be
called.

Bug: None
Change-Id: Iaa8499eda1b52cc14b04622bea2766cd09ba43e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160186
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Xavier Lepaul <xalep@google.com>
Cr-Commit-Position: refs/heads/master@{#29866}
2019-11-21 17:04:50 +00:00
58a3210823 Add config to reduce weight on small samples in BitrateEstimator.
Change #159711 adds the option to filter out small packets on the
input to the delay-based BWE. This change adds similar functionality
to BitrateEstimator by reducing the weight of small observations.

Bug: webrtc:10932
Change-Id: I0a673a067f7ef86769cabd30443e60e9de70053c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160009
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29865}
2019-11-21 15:52:25 +00:00
63737a918b Add new GOOG_PING and GOOG_MESSAGE_INTEGRITY_32
This patch adds
- Attribute: STUN_ATTR_GOOG_MESSAGE_INTEGRITY_32
  which is a ordinary message integrity but truncated to 32-bit
- Method: GOOG_PING,
  which will be used for webrtc:11100

Both the attribute and the method has been registered at iana,
https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-4

BUG=webrtc:11100

Change-Id: Iddd5614473fd6f18fbbe76e72d047c617df7123f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160180
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29864}
2019-11-21 15:29:50 +00:00
026f64fc54 Revert "VideoFrame: Store a reference to an encoded frame"
This reverts commit e6eded31e642b3b986fef478315603b5f398c227.

Reason for revert: A better method for communicating encoded frames in VideoTrackSourceInterface surfaced.

Original change's description:
> VideoFrame: Store a reference to an encoded frame
> 
> Enable webrtc::VideoFrame to store a reference to an encoded frame.
> 
> Bug: chromium:1013590
> Change-Id: Id5a06f1c7249f104dfd328f08677cf8001958f0d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158788
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29809}

TBR=ilnik@webrtc.org,nisse@webrtc.org,stefan@webrtc.org,philipel@webrtc.org,handellm@webrtc.org

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

Bug: chromium:1013590
Change-Id: I46384b7997e7b1cd3a2a2042cf17890fc977cca3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160204
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29863}
2019-11-21 14:55:21 +00:00
038fd99780 Add RtpDepacketizerAv1::AssembleFrame function
Bug: webrtc:11042
Change-Id: I677fc6a9affacf3b7c80adc2c3493c16806db1f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160003
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29862}
2019-11-21 14:50:41 +00:00
ad020f5a5c Delete DEPRECATED_SingleThreadedTaskQueueForTesting
Bug: webrtc:10933
Change-Id: I4d04b15a5691e586595d59ecd1c8012cba5e3dc6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160041
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29861}
2019-11-21 13:41:57 +00:00
76a7e518bd [UBSan] Add suppressions for vp8, opus, absl.
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}
2019-11-21 13:40:47 +00:00
662678dbf7 Adds injectable trials from peerconnection down to transport controller.
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}
2019-11-21 12:41:45 +00:00
ac7fd87375 Force alignment of generated JVM called functions.
This CL effectively expands the zone of influence of
https://webrtc-review.googlesource.com/64160,
forcing 16-byte stack alignment of generated JNI methods
for the Android x86 platform.

Bug: webrtc:9085
Change-Id: Idc40c00ea3fb52dbbbeac7b58ceda2a9a44733d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159928
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29858}
2019-11-21 12:34:35 +00:00
ec22183f43 Revert lock for logging to CriticalSection
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}
2019-11-21 12:09:25 +00:00
9982efa830 MediaStreamInterface: introduce encoded sinks.
This change adds a new type of sink for consuming encoded data from
a video source.

Bug: chromium:1013590
Change-Id: Ia7c4e372190c3d6bc007a0d4deb05c2d1bce58d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159927
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29856}
2019-11-21 12:03:35 +00:00
b86a1770ee Expose ABGRToI420 in YuvHelper.
Bug: None
Change-Id: I59947339a3a4bb683211ec3c00713ccfbf35bc40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160182
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29855}
2019-11-21 12:02:30 +00:00
e835fc01b1 Add UMA counter for audio interruptions
The metric is added to Chromium histograms in
https://chromium-review.googlesource.com/c/chromium/src/+/1925066.

Bug: webrtc:10549
Change-Id: I2bf98f469547aa8621832fc4f8bd29c4805ac0b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160045
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29854}
2019-11-21 11:40:21 +00:00
c4e506ebdf Fix writing into closed FrameWriter when dumping video in PC framework
Bug: webrtc:10138
Change-Id: I2b8e2cac2903c957dab1170ba098880a4f0252e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160049
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29853}
2019-11-21 10:01:57 +00:00
22ae169d17 Remove dependencies on Chromium's webrtc_overrides.
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}
2019-11-21 09:42:07 +00:00
108a2f025d Preventively fix missing braces warnings.
This CL fixes warnings that will cause issues with new versions of the
Android NDK.

Bug: None
Change-Id: Icd119efec6039d4861d89401b84f94c8da29a314
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160080
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29851}
2019-11-20 19:38:55 +00:00
09c452e7ba Split P2PTransportChannel
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}
2019-11-20 19:35:45 +00:00
d51cc7bd71 Add absolute capture time property to rtp sources.
This part of the effort to implement A/V sync metric.

Bug: webrtc:10739
Change-Id: I4adba1b99b37b31868168e37d9aa8e03f8ea6d4e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159886
Commit-Queue: Ruslan Burakov <kuddai@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ruslan Burakov <kuddai@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29849}
2019-11-20 18:50:45 +00:00
bb55e0bc72 Clarifies identification of default communication device in ADM2
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}
2019-11-20 15:02:06 +00:00
eec1d04aa9 Roll chromium_revision 8b5b046f02..022da7ca60 (715044:716991)
Change log: 8b5b046f02..022da7ca60
Full diff: 8b5b046f02..022da7ca60

Changed dependencies
* src/base: 20e7dd7480..29c24a8b46
* src/build: ed0d663913..c60d0d9d66
* src/buildtools: 140e4d7c45..6b3e658d6f
* src/buildtools/third_party/libc++/trunk: 5938e0582b..78d6a7767e
* src/ios: 899a8a4bc9..01eee3d70d
* src/testing: d4d918486a..a6c25af3b7
* src/third_party: aad8135d1e..24a1541bd5
* src/third_party/bazel: tQPvsIj1Gtw5iXssKy7OREE-S02u7zItrw42l3DHUroC..VjMsf48QUWw8n7XtJP2AuSjIGmbQeYdWdwyxVvIRLmAC
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/d709b0d892..6ba98ff601
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3992f65642..55394ddc90
* src/third_party/depot_tools: 9577daf667..639872c8ef
* src/third_party/libyuv: 53b529e362..6afd9becdf
* src/third_party/lss: https://chromium.googlesource.com/linux-syscall-support.git/+log/8048ece6c1..726d71ec08
* src/tools: 224c37eeae..f2ed5a7b38
DEPS diff: 8b5b046f02..022da7ca60/DEPS

No update to Clang.

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

Change-Id: I44e824d8cfee4c3f4536be4442b93ed30fefe56b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160060
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@{#29847}
2019-11-20 14:42:26 +00:00
6dd488b2e5 RTC_EXPORT RTCStatsMember's specialized members.
Without this, on some build configurations the symbols of the
specialized members don't get exported as explained at:
https://bugs.chromium.org/p/chromium/issues/detail?id=1026078#c10

Bug: chromium:1026078
Change-Id: I0c3058a82d60e6de5e401dbec5bb8501b7bbd8b5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160046
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29846}
2019-11-20 13:53:37 +00:00
dca14499be Makes RoundRobinPacketQueue use same field trials as PacingController.
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}
2019-11-20 13:36:46 +00:00
5831ddad65 Introduce IVF file reader
Bug: webrtc:10138
Change-Id: I97d332942f4e645527330159efefb1cb1d8034a0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160008
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29844}
2019-11-20 13:20:56 +00:00
1721de12bd Add STUN_ATTR_GOOG_MISC_INFO
This patch adds the new STUN attribute that has been registered at iana,
https://www.iana.org/assignments/stun-parameters/stun-parameters.xhtml#stun-parameters-4

This is part of the effort to land https://webrtc-review.googlesource.com/c/src/+/85520.
I have merged that patch with upstream, and is now doing privacy review of it.

This attribute is hence not yet used.

BUG=webrtc:9446

Change-Id: Iaf177b0c28a6aa830a9422260b67436bb05ac756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160043
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29843}
2019-11-20 13:07:25 +00:00
2f385d2ab8 Manual chromium roll: Compile using JDK 11
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}
2019-11-20 12:13:43 +00:00
3b1a8bb00c Account for rounding errors in dyanmic pacing mode.
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}
2019-11-20 11:23:43 +00:00
83b286202b Protect against NumberOfEnumeratedDevices and Get[In|Out]putDeviceNames returning inconsistent results.
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}
2019-11-20 08:51:27 +00:00
4dd56a3830 ACM: Adding unittests for the remixing functionality
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}
2019-11-20 06:20:22 +00:00
0e3198e434 Refactoring of the analog AGC functionality to add multichannel support
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}
2019-11-19 23:39:07 +00:00
f3fcde36c2 Store delay measurements as struct instead of std::pair
Bug: None
Change-Id: I60f375cda4f910550a86d2238acf39d429e2a17b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160004
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29837}
2019-11-19 17:44:11 +00:00
fa7a8ca21c Revert "Prepares PacingController for simplified packet queue."
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}
2019-11-19 16:54:32 +00:00
0660ceef0e Add scale and compare methods to VideoFrame::UpdateRect
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}
2019-11-19 15:42:42 +00:00
332274dfef Adding GetInDtx to WebRTC Opus Interface.
Bug: webrtc:11085
Change-Id: Ie9152cbe3f3c70f6febafb877852d68a831bcae9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159708
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29834}
2019-11-19 14:14:06 +00:00
67e5bd3442 [Android SDK] Update to version 'Q' (29) to unblock autoroller/roll_deps.py
Similarly to [1], this fixes the current autoroller breakage [2]
by manually pre-updating the cipd package versions for
'third_party/android_sdk/public'.

For consistency sake //src/build has been updated to [3].

[1] https://webrtc-review.googlesource.com/c/src/+/158884
[2] https://logs.chromium.org/logs/webrtc/buildbucket/cr-buildbucket.appspot.com/8896958712762421712/+/steps/autoroll_DEPS/0/stdout
[3] da9baf4669

Bug: webrtc:11095, chromium:1003532
Change-Id: Ib01143ec270ecc37eb82d061ae5d011059afd17a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159929
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29833}
2019-11-19 13:57:30 +00:00
063c7d18c0 In dependency descriptor remove extended fields indicator
to follow PR64 spec change
https://github.com/AOMediaCodec/av1-rtp-spec/pull/64

Bug: webrtc:10342
Change-Id: Ic082d5e551b5f38427d5a43be987b0d35f6ea155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160001
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29832}
2019-11-19 13:12:10 +00:00
fe047757d6 Fix a bug in interruption metrics
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}
2019-11-19 12:58:50 +00:00
29e07e5080 Add @Nullable annotations to quiet errorprone.
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}
2019-11-19 12:50:30 +00:00