Commit Graph

29258 Commits

Author SHA1 Message Date
8fe22fad4e StreamSynchronizationTest: Replace class Time with SimulatedClock.
Remove unused constants and variables.

Bug: none
Change-Id: I7336bbe5bfecbaaf646c9704e4f75532629754d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159944
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29826}
2019-11-19 08:48:45 +00:00
db82cd7e4f Make android_compile_x64_rel actually compile in x64.
Due to a copy/paste error, the bot was compiling in x86.

Bug: webrtc:11097
Change-Id: I55b013f20707915886fa04956a37fb3fec0477b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159931
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29825}
2019-11-18 20:21:53 +00:00
351173c88c Tests that all available audio devices can be selected and used by the ADM.
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}
2019-11-18 17:47:31 +00:00
8ae70f6a30 Enable WebRTC-Bwe-MaxRttLimit by default.
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}
2019-11-18 16:53:11 +00:00
56d945233d Move stun.h to api/.
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}
2019-11-18 16:11:27 +00:00
aa3f5da8dc Fork VCMPacket for PacketBuffer into own struct
it is easier to reduce and eliminate it when it is not bound to legacy video code

Bug: webrtc:10979
Change-Id: I517e298501b3358a914a23ddce40fcb3075d672d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159707
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29821}
2019-11-18 15:48:07 +00:00
acdc22d784 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}
2019-11-18 15:37:58 +00:00
ccf12c6e97 Reland "Add AV1 RtpDepacketizer class"
This is a reland of 49470c2ac460ed8cce250942e8525c5f14e32778
Tentative reland to rule-out bot flakiness.

Original change's description:
> Add AV1 RtpDepacketizer class
>
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
>
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=saza@webrtc.org,philipel@webrtc.org

Bug: webrtc:11042
Change-Id: Ibd672ce685bcab86960500740465539ed70fcdf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159941
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29819}
2019-11-18 15:23:08 +00:00
cb0b87473a Add source-side perf upload script for WebRTC.
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}
2019-11-18 14:37:01 +00:00
4186603902 Always record timestamp of keyframe request.
Bug: chromium:1013590
Change-Id: I85b20f06cb0bec15dae199cf96512173f0faad42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159884
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29817}
2019-11-18 12:17:48 +00:00
92dd35d035 Reland "Force Chromium deps on the WebRTC component."
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}
2019-11-18 12:16:43 +00:00
9f99175710 Revert "Add AV1 RtpDepacketizer class"
This reverts commit 49470c2ac460ed8cce250942e8525c5f14e32778.

Reason for revert: Seems to trigger linker error on iOS64. See:
https://ci.chromium.org/p/webrtc/builders/ci/iOS64%20Debug/17733

Original change's description:
> Add AV1 RtpDepacketizer class
> 
> Implement Parse function that extracts is_first_packet_in_frame,
> is_last_packet_in_frame, and frame_type fields.
> 
> Bug: webrtc:11042
> Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
> Reviewed-by: Philip Eliasson <philipel@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29814}

TBR=danilchap@webrtc.org,saza@webrtc.org,philipel@webrtc.org

Change-Id: I2eb5994d8e31e12d6cb6e9f792b691ed10d9df81
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11042
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159940
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#29815}
2019-11-18 12:14:56 +00:00
49470c2ac4 Add AV1 RtpDepacketizer class
Implement Parse function that extracts is_first_packet_in_frame,
is_last_packet_in_frame, and frame_type fields.

Bug: webrtc:11042
Change-Id: I9360ea52ef274281b5c5e4c31955100b92155bfe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159180
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29814}
2019-11-18 09:39:34 +00:00
2419dc9cfc Revert "Force Chromium deps on the WebRTC component."
This reverts commit 53e157d25ce78ba6cd8625b0b655b46f8e1b0a91.

Reason for revert: Breaks Chromium iOS FYI bots.
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/5088

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=mbonadei@webrtc.org,kwiberg@webrtc.org,dpranke@chromium.org

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

TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Id4d906910d569a3e5db3afef8c03672fba6dad81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159921
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29813}
2019-11-18 08:11:30 +00:00
313a10ecef p2p_transport_channel_unittest : put frequently used checks into functions.
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}
2019-11-18 07:31:24 +00:00
39bab5afb5 Add missing assert.h for win no-test build
Add some missing `#include <assert.h>` for Windows build when compiling
without RTC tests (rtc_include_tests = false) with the MSVC compiler
(is_clang=false, use_lld=false).

Bug: None
Change-Id: Ie9861100efeae87f4c4e29303d62293ad541125a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158533
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29811}
2019-11-17 07:52:32 +00:00
408cb4bf30 Make SCTPtransport enter "closed" state when DTLStransport does.
Bug: webrtc:11090
Change-Id: I30e0b70387746d6c544ed1818f276569d4258cf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159888
Reviewed-by: Emad Omara <emadomara@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29810}
2019-11-16 14:56:01 +00:00
e6eded31e6 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}
2019-11-16 10:15:38 +00:00
ef6fe0cf2b Use GetDefaultAudioEndpoint for the default communications device as well as the vanilla default device
Bug: b/144524502
Change-Id: I3349010a2f2d67cde29a61740496c38712f0f391
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159900
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29808}
2019-11-15 23:26:07 +00:00
25ec8882f7 Make ICE transports injectable.
Bug: chromium:1024965
Change-Id: I4961f50aee34c82701299f59a95cb90d231db6f5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158820
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Honghai Zhang <honghaiz@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29807}
2019-11-15 21:31:19 +00:00
53e157d25c 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}
2019-11-15 16:36:28 +00:00
bb55c5e2ec Correct the upmixing of mono to stereo in ACM2
This CL is a correction to the former CL that changed the remixing for
surround. A bug in that CL caused the upmixing from mono to stereo to
place zeros in the right channel.

The unittest CL is present in https://webrtc-review.googlesource.com/c/src/+/155740

Bug: b/144458371
Change-Id: I192e587a1b083a7bb55dcac2343f8b6d3942b9ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159864
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29805}
2019-11-15 16:07:30 +00:00
fd0e32a87a Fix filtering of small packets in delay-based BWE
crodbro@ found that the previous field trial, which filtered the deltas
in the trendline estimator, can increase the noise caused by varying
packet sizes. Moving the filtering to the DelayBasedBwe class fixes the
issue.

To avoid confusion, we've updated the field trial name, so e.g.
WebRTC-BweIgnoreSmallPacketsFix/small:200bytes,large:200bytes,
                                fraction_large:0.25,smoothing:0.1/
should be used to enable the feature.

Bug: webrtc:10932
Change-Id: If77e83043c37fff909038405f634e541ce41abb8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159711
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29804}
2019-11-15 14:53:59 +00:00
7b46e17c31 In rtc::ByteBuffer drop support for ORDER_HOST as unused
Bug: None
Change-Id: Ideab428b13d981cddf9784cfd07fb7dfb2e914fe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159698
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29803}
2019-11-15 11:48:42 +00:00
74f35e48d5 Add support for dynamic processing mode in PacedSender.
Behind a default-disabled field trial.

Bug: webrtc:10809
Change-Id: If5d9b69721bd67e59e68b1026e3797e9a1b0a760
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159783
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29802}
2019-11-15 08:56:00 +00:00
02fac7d86e Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

Chromium CL that introduces the component build support:
https://chromium-review.googlesource.com/c/chromium/src/+/1874722

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Tree-Checks: true
No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: Iff8e35c6f9a53a0d08979bc873b6488dd7164ba5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159860
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29801}
2019-11-14 19:51:04 +00:00
eb48799ec5 Prepares PacingController for scheduled send tasks.
This CL is in preparation for a dynamic (possible TaskQueue-driven)
pacer that instead of processing blindly every 5ms, posts delayed
tasks to be executed when it is actually time to send packs.

This means we need the pacing controller to be able to figure out when
those execution times shall be, and be able to correctly update budget
levels as IntervalBudget only works correctly with periodic processing.

Bug: webrtc:10809
Change-Id: Idd12acaabfb24cc2e6bcc589aac206cd04beb6e4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/158790
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29800}
2019-11-14 13:53:56 +00:00
3527a4fe55 In PacketBuffer split logic for detecting frame boundaries and assembling frame.
Bug: webrtc:11042
Change-Id: If1695067054b332569f4839aa6762af33173b769
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159283
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29799}
2019-11-14 13:32:06 +00:00
01294f0e29 Don't configure video codec switching if no video stream has been created.
Bug: none
Change-Id: I8e74fefed1e902c35064700f826b8f565e18c704
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159800
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29798}
2019-11-14 13:12:50 +00:00
bb0aac27e3 Reduce verbosity of logging around playout underrun count on iOS.
This method is called on every GetStats call and fills up log output on iOS
with three log lines per cycle at INFO+ (the not-supported one is LS_ERROR):
[181:040] [82471] (audio_device_module_ios.mm:646): GetPlayoutUnderrunCount
[181:040] [82471] (audio_device_generic.cc:48): GetPlayoutUnderrunCount: Not supported on this platform
[181:040] [82471] (audio_device_module_ios.mm:649): output: -1

Alternatively, we could remove the error logging in the base class, or (better) log it once the first time it is called, but this is the simpler change.

Bug: None
Change-Id: Ibaa1d176f10cdc92f2ba1a6bf15aaa580da6edb4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159672
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29797}
2019-11-14 09:49:39 +00:00
f11b46bc4b Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit f181137b05e4b899fa3f15afafc4f27e683d83cc.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> No-Try: True
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I8582242910bb3633b7a4675ff261b5a6a0b6954e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159712
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29795}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I6bbe0288d07e49c8a4c808c758ecb6e2ddfa2aa8
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159713
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29796}
2019-11-13 16:49:52 +00:00
f181137b05 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I8582242910bb3633b7a4675ff261b5a6a0b6954e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159712
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29795}
2019-11-13 16:45:40 +00:00
1b66890a45 Stop using DEPRECATED_SingleThreadedTaskQueueForTesting in call tests
This is practiaclly a reland of the
https://webrtc-review.googlesource.com/c/src/+/157896
except that video multi stream tests are still using the deprecated TaskQueue
(see https://webrtc-review.googlesource.com/c/src/+/159280)

Bug: webrtc:10933
Change-Id: Ie715345924f9dd2d7dd52c99de3ea595b6fad5ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159699
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29794}
2019-11-13 16:21:40 +00:00
60bd1aea3d Detach thread checker in RtcEventLogImpl destructor.
Otherwise we require that the destructor must run on the same thread as
previous calls. This is not necessary since we can assume there's no
other references to the instance when we enter the destructor.

Bug: webrtc:9883
Change-Id: Ia254bce9265979da0e25ba33598edd8f807d7e00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159704
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29793}
2019-11-13 15:42:19 +00:00
048b10a9ec Correcting the ACM upmixing from mono/stereo to surround
This CL corrects the upmixing from mono/stereo to surround in the audio
coding module.


Bug: webrtc:11083
Change-Id: Ic529107d59ff54a8e48b0424cbdf2b49b7a65c12
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159705
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29792}
2019-11-13 15:23:19 +00:00
50f8686c6d Providing EmulatedRoute instances when creating TcpMessageRoute
This CL makes it so the caller explicitly has to provide
EmulatedRoute instances when creating TcpMessageRoute.
Previously those were automatically generated.

This means that the EmulatedRoute instances can be reused.

Bug: webrtc:9883
Change-Id: I7ad03ca6a5a3dbb91df76e3ca1190a1a75bf0cc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159703
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29791}
2019-11-13 14:55:39 +00:00
017c84f3ea Synchronize is_screencast_ state in AndroidVideoTrackSource.
Follow up to https://webrtc-review.googlesource.com/c/src/+/159689.

Bug: None
Change-Id: I3f2b481db091d405c1b00ca18c2e7ce5f3375607
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159702
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29790}
2019-11-13 14:03:09 +00:00
c20a19cc4b Allow extracting the linear AEC output
This CL enables extracting the linear AEC output,
allowing for more straightforward
testing/development.

Bug: b/140823178
Change-Id: I14f7934008d87066b35500466cb6e6d96f811688
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/153672
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29789}
2019-11-13 11:33:53 +00:00
06a394e036 Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit 57d53cf1944952ac67df2f6a3522f38cdc01d0c1.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> No-Try: True
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I4719f5b5607ea491689429ca327a3521729e4ba7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159700
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29787}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I1775bdac3ab9888d36f1552dd2eaaa000c43c9b9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29788}
2019-11-13 11:21:52 +00:00
57d53cf194 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I4719f5b5607ea491689429ca327a3521729e4ba7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159700
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29787}
2019-11-13 11:20:40 +00:00
759f161182 Use explicit template decl/def for RTCNonStandardStatsMember.
This should fix the error raised by lld-link while testing the WebRTC
component build (see [1]).

[1] - https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/436729

Bug: webrtc:9419
Change-Id: Ia239f544432c55ae69e2da474b8284143c94a003
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159697
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29786}
2019-11-13 11:12:40 +00:00
c5ec54e51b Add SetIsScreencast method to VideoSource.
Bug: None
Change-Id: Iec0bb066b8100fa1d4bd095f78a0473933d1e30d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159689
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29785}
2019-11-13 10:30:36 +00:00
c1dac38aec Revert "Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.""
This reverts commit f3dccc63f32202d2a3e3bf2d87c95c141cf0963c.

Reason for revert: This CL was just needed in order to
have a WebRTC commit to pin in Chromium to test the
component build (this CL enables symbol exports).

Original change's description:
> Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
> 
> This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed
> 
> I will revert this reland as soon as it lands because I just need
> to have a WebRTC commit to pin in Chromium in order to test the
> component build (this CL enables symbol exports).
> 
> Original change's description:
> > Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
> >
> > In order to land the component build support in Chromium, it is
> > easier to turn on symbols export every time that is_component_build=true
> > instead of setting rtc_enable_symbol_export=is_component_build in
> > Chromium (since is_component_build is not available in .gn).
> >
> > rtc_enable_symbol_export is still kept in the mix in order to turn
> > on symbol exports in any case a shared library will be added to the
> > WebRTC build.
> >
> > Bug: webrtc:9419
> > Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#29540}
> 
> No-Try: True
> TBR: kwiberg@webrtc.org
> Bug: webrtc:9419
> Change-Id: I73ae6ff93da78f9f0b2b53f0c6b12693514bc20d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159695
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29783}

TBR=mbonadei@webrtc.org,kwiberg@webrtc.org

Change-Id: I5c9c267278140d608b5063cc174eac9a2103636d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9419
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159696
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29784}
2019-11-13 08:55:54 +00:00
f3dccc63f3 Reland "Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true."
This is a reland of 03bc15c646d5b41d3169f2686316944788f640ed

I will revert this reland as soon as it lands because I just need
to have a WebRTC commit to pin in Chromium in order to test the
component build (this CL enables symbol exports).

Original change's description:
> Define WEBRTC_ENABLE_SYMBOL_EXPORT if is_component_build=true.
>
> In order to land the component build support in Chromium, it is
> easier to turn on symbols export every time that is_component_build=true
> instead of setting rtc_enable_symbol_export=is_component_build in
> Chromium (since is_component_build is not available in .gn).
>
> rtc_enable_symbol_export is still kept in the mix in order to turn
> on symbol exports in any case a shared library will be added to the
> WebRTC build.
>
> Bug: webrtc:9419
> Change-Id: I5a7195826dea13d9a6f10a1160c35f2864bfa6c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/157108
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29540}

No-Try: True
TBR: kwiberg@webrtc.org
Bug: webrtc:9419
Change-Id: I73ae6ff93da78f9f0b2b53f0c6b12693514bc20d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159695
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29783}
2019-11-13 08:54:49 +00:00
1242d9cc48 Reland Cleanup MultiStreamTester
Instead of taking TaskQueue from outside create one internally.
Detach MultiStreamTests from test::CallTest since that inheritance
only used for constants and for task_queue object.

Unlike original cleanup
keep using DEPRECATED_SingleThreadedTaskQueueForTesting for now.

Bug: webrtc:10933
Change-Id: Ife9143bfda0ebefd56a9199622296e64b14a7b20
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159034
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Original-Commit-Position: refs/heads/master@{#29744}
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159280
Cr-Commit-Position: refs/heads/master@{#29782}
2019-11-13 08:53:22 +00:00
d4002a733d RTC_EXPORT missing symbols for Chromium's component build.
This CL adds a dependecy on rtc_base/system:rtc_export to rtc_event but
only when built as part of Chromium (since rtc::Event should not be
used outside of WebRTC).

It also adds other missing RTC_EXPORTS.

Bug: webrtc:9419
Change-Id: Ib338004a5404a6b3c7929e146c29ad42572632cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159692
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29781}
2019-11-12 21:50:01 +00:00
f47c2ab889 RTC_EXPORT webrtc::RTCNonStandardStatsMember<T>.
Since webrtc::RTCNonStandardStatsMember<T> extends some explicit
template initializations, its symbols need to be exported as well.

Example of an lld-link error:
https://ci.chromium.org/p/chromium/builders/try/win_chromium_compile_dbg_ng/435974

Bug: webrtc:9419
Change-Id: I48bd484b821b17a214e0180e78ba34c38e247947
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159693
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29780}
2019-11-12 21:19:11 +00:00
4ed17ff555 Roll chromium_revision 3b0967fe5b..0717a4dc31 (714463:714607)
Change log: 3b0967fe5b..0717a4dc31
Full diff: 3b0967fe5b..0717a4dc31

Changed dependencies
* src/base: 373c30801f..c2926851da
* src/build: a271bb921d..7e97a0cab3
* src/ios: 82ed7976ea..899a8a4bc9
* src/testing: 4a04c8503f..d4d918486a
* src/third_party: e7c00d06d6..d358c8811f
* src/third_party/depot_tools: 79d4f99506..9577daf667
* src/third_party/ffmpeg: 83304c4e52..fa04e15e1a
* src/tools: c1b8d97c4c..ca0982ef67
DEPS diff: 3b0967fe5b..0717a4dc31/DEPS

No update to Clang.

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

Change-Id: I5cd71ba1ecd4d1e2379bfa7b6401ba79ed880d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159671
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@{#29779}
2019-11-12 20:44:10 +00:00
86b33e0b7e Don't ask for the friendly name of a default device if we failed to enumerate it.
Bug: b/144233691
Change-Id: I5f80c63858ec851ab14bcc3c1ca51ca2e9507834
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159582
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Tim Haloun <thaloun@chromium.org>
Cr-Commit-Position: refs/heads/master@{#29778}
2019-11-12 18:41:24 +00:00
dcc910a209 Adds simulated TCP message route for testing.
This TCP message route allows simulation of sending a fixed lengths message
over an existing route. This can be used to simulate reliable signaling in
tests as well as simulating the cross traffic impact of TCP connection.

It is based on the existing Fake TCP cross traffic implementation.

Bug: webrtc:9510
Change-Id: Ibfc2a9a5b95593b00db16de2c09ce929077cf5c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159482
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29777}
2019-11-12 17:26:15 +00:00