1561 Commits

Author SHA1 Message Date
Danil Chapovalov
d06588a758 Change Av1 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I322115263f60439bee36277157a0acef9bd28e3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165343
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30260}
2020-01-15 10:16:03 +00:00
Sebastian Jansson
f1173f46e5 Revert "Using simulated rtc::Thread for peer connection scenario tests."
This reverts commit b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453.

Reason for revert: Interferes with other tests in same binary.

Original change's description:
> Using simulated rtc::Thread for peer connection scenario tests.
> 
> Bug: webrtc:11255
> Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30258}

TBR=steveanton@webrtc.org,srte@webrtc.org

Change-Id: If2e60edae264a4bb0dee3abf66ba2078fd85f493
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166045
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30259}
2020-01-15 10:10:07 +00:00
Sebastian Jansson
b70c5c5ce9 Using simulated rtc::Thread for peer connection scenario tests.
Bug: webrtc:11255
Change-Id: I5d29e997a7209ffc64595082358cca9b2115d07a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165689
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30258}
2020-01-15 09:35:40 +00:00
Sebastian Jansson
3d4d94a832 Adds scenario test for transport wide feedback based retransmission.
This ensures more end to end test coverage of the feature and captures
a wider class of regression then the existing unit test.

Bug: webrtc:9883
Change-Id: I6e74e571500c5c5d74caf8f661cac08bee8934f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164461
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30252}
2020-01-14 16:00:14 +00:00
Jonas Olsson
b2b2031457 Concatenate string literals at compile time.
This CL was generated by running:
git ls-files | grep ".cc" | xargs perl -i -ne 'BEGIN {undef $/}; s/("[\s\n]*<<[\s\n]*")/" "/g; print;'; git cl format

After that I manually edited modules/audio_processing/gain_controller2.cc to preserve its original
formatting.

This primary benefit of this change is a small reduction in binary size.

Bug: None
Change-Id: I689fa7ba9c717c314bb167e5d592c3c4e0871e29
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165961
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30251}
2020-01-14 14:47:48 +00:00
Sebastian Jansson
3e66a498c3 Use RTX SSRCs in scenario test framework.
Using RTX SSRCs and payload type for retransmission of video. This
corresponds to the behavior when using the peer connection API.

Bug: webrtc:9883
Change-Id: Ic0e3964d097f42219ca225513a4bc771d70ccaa4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164460
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30248}
2020-01-14 12:04:56 +00:00
Sebastian Jansson
2a92d2b461 Cleanup: Prepares for simulated time peer connection tests.
This CL contains some preparatory cleanup that can be done
outside the main CL.

Bug: webrtc:11255
Change-Id: Ib0dcd81d352bafc446dcd2f7f82ba81f5e82e210
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165766
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30247}
2020-01-14 09:55:42 +00:00
Patrik Höglund
2ea27968d3 Extract an interface from the perf results logger.
The new interface is called PerfTestResultWriter and is currently
implemented by PerfResultsLogger (renamed PerfTestGraphJsonWriter).

I plan to introduce a second implementation of the perf logger that
uses the new Histogram C++ API. I add a flag that chooses
between the two implementations so I can try it out (perhaps by
setting up a second, limited run of webrtc_perf_tests on the perf
bots that uses the new implementation). The histogram C++
implementation will come in the next patch.

As a side effect, I disentangled the plottable counter printer from
the perf result printer so it will work for both implementations.
The only thing they had in common was that both wrote JSON anyway.

See the bug for details on the new API.

Bug: chromium:1029452
Change-Id: Icb21b25ced08ea73aeecd221e9d51f2adf3dab1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165389
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30243}
2020-01-14 06:05:02 +00:00
Danil Chapovalov
7d43801a07 Delete RtpGenericDepacketizer as no longer used
Bug: webrtc:11152
Change-Id: I275765e1aa013d8188d43e2911e8ab022563d1d8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165394
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30234}
2020-01-13 13:45:37 +00:00
Sebastian Jansson
53cd9e2645 Separates simulated TaskQueue and simulated ProcessThread.
The overlap in functionality is quite limited and separating the
functionality makes it a bit easier to follow each. This prepares
for adding a SimulatedThread class in a follow up CL.

Bug: webrtc:11255
Change-Id: I83c754bd570113dfb582098bb4d39e27bb4f4d87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165688
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30229}
2020-01-13 11:26:42 +00:00
Sebastian Jansson
9d4bbc216b Using tasks to process packets in FakeNetworkSocketServer.
This way we can rely on existing task scheduling and execution
functionality, reducing the required functionality to support the
fake socket server.

This prepares for support simulated time execution of peer
connection level tests.

Bug: webrtc:11255
Change-Id: I7de64a099c2e355c70929ecff79b8ea3b98b70b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165398
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30221}
2020-01-12 12:53:30 +00:00
Sebastian Jansson
290de82b2a Cleanup: Replace MessageQueue pointers with Thread pointers.
This is part of a CL series merging rtc::MessageQueue into rtc::Thread.

Bug: webrtc:9883
Change-Id: I4a1bcd44c9523b6402b3f05b50597bdc2e6615e3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165345
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30216}
2020-01-10 19:03:12 +00:00
Patrik Höglund
6ddbe2c5b0 Extract results line plotting.
This will make RESULT lines still come out after we add a second JSON
writer implementation.

Bug: chromium:1029452
Change-Id: I5cba3151c21df2901f19305e9b71bc5c9638a0ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165399
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30208}
2020-01-10 11:29:55 +00:00
Patrik Höglund
e1cbb9c20e Extract plottable counter from perf results logger.
Split out of https://webrtc-review.googlesource.com/c/src/+/165389.

I disentangled the plottable counter printer from the perf result
printer so it will work for both future implementations of the perf
test JSON writers. The only thing plottable counters and the
results writer had in common was that both wrote JSON anyway.

Bug: chromium:1029452
Change-Id: I041c3096641eda42542e8d994b246eb313940b4b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165397
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30198}
2020-01-09 15:40:26 +00:00
Danil Chapovalov
5c35f2fb1b Delete RtpDepacketizerVp9 in favor of VideoRtpDepacketizerVp9
Bug: webrtc:11152
Change-Id: Ic50f2dc49ca420b3406d4dea11ed20328aa59136
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165382
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30195}
2020-01-09 13:07:44 +00:00
Danil Chapovalov
26e1b7ac01 Delete RtpDepacketizerVp8 in favor of VideoRtpDepacketizerVp8
Bug: webrtc:11152
Change-Id: I1a6225701ecd6f7a34c946d7296f0ab0cbb5eaef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165342
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30190}
2020-01-09 12:10:19 +00:00
Sebastian Jansson
a406ee1ca2 Moving FakeNetworkSocket to fake_network_socket_server.h
This means that we avoid exposing FakeNetworkSocket and
moves related code closer together.

It's done in preparation for future work on simulated time testing.

Bug: webrtc:9883
Change-Id: Id6d1b0a6055f30da8e6646bd5347024fbd9c9dfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164537
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30181}
2020-01-08 14:23:36 +00:00
Danil Chapovalov
1b4e4bf42e Migrate several call tests from legacy RtpHeaderParser to RtpPacket parsing.
Bug: None
Change-Id: I96f11922b3cd66eb02093fa7e6e4d21774c19575
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161323
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30166}
2020-01-07 13:02:52 +00:00
Yves Gerey
eb3beb8504 Revert "Replace the ExperimentalAgc config with the new config format"
This reverts commit f3aa6326b8e21f627b9fba72040122723251999b.

Reason for revert: Breaks downstream project.

Original change's description:
> Replace the ExperimentalAgc config with the new config format
> 
> This CL replaces the use of the ExperimentalAgc config with
> using the new config format.
> 
> Beyond that, some further changes were made to how the analog
> and digital AGCs are initialized/called. While these can be
> made in a separate CL, I believe the code changes becomes more
> clear by bundling those with the replacement of the
> ExperimentalAgc config.
> 
> TBR: saza@webrtc.org
> Bug: webrtc:5298
> Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
> Reviewed-by: Per Åhgren <peah@webrtc.org>
> Commit-Queue: Per Åhgren <peah@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30149}

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

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

Bug: webrtc:5298
Change-Id: I794d2ab4b8caa5330c5ad490ba604646a249a1c1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164530
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30153}
2020-01-07 05:22:01 +00:00
Per Åhgren
f3aa6326b8 Replace the ExperimentalAgc config with the new config format
This CL replaces the use of the ExperimentalAgc config with
using the new config format.

Beyond that, some further changes were made to how the analog
and digital AGCs are initialized/called. While these can be
made in a separate CL, I believe the code changes becomes more
clear by bundling those with the replacement of the
ExperimentalAgc config.

TBR: saza@webrtc.org
Bug: webrtc:5298
Change-Id: Ia19940f3abae048541e6716d0184b4caafc7d53e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163986
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30149}
2020-01-03 23:14:13 +00:00
Yves Gerey
2257c087b1 [Cleanup/Optim] Pass IPAddress by const reference.
The IPAddress class (32 bytes) was copied for each invocation.
This CL also saves some bytes in generated binary.

Bug: webrtc:9855
Change-Id: I40f2fe8570ee30d1d2251fddd56131ca4c3e7155
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164521
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30147}
2020-01-03 18:42:32 +00:00
philipel
d572748885 Run delay tasks on time when using GlobalSimulatedTimeController.
This change means tasks scheduled at the end time reached when making a call to GlobalSimulatedTimeController::AdvanceTime will also be executed.

In other words, with this change, if you schedule a task in X milliseconds and then call AdvanceTime(TimeDelta::ms(X)) the scheduled task will be executed.

Bug: none
Change-Id: I337e574a88b235639e82ffcacf1484daa6cf3172
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164522
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30146}
2020-01-03 16:37:42 +00:00
Natalie Chouinard
65bbcabe2f [Android] Replace java_files with sources
Replace all usages of java_files with sources in gn files, and
automatically format.

This is in preparation for java_files being completely removed upstream
in favor of sources.

NOPRESUBMIT=true

Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
2020-01-02 20:08:20 +00:00
Sebastian Jansson
f4cf4c789a Don't allow creation of sockets for wild card IPs in emulated networks.
The network emulation framework does not support creation sockets that
receive from all addresses (e.g. 0.0.0.0) but would instead crash at
runtime. This CL explicitly ensures that we don't provide such networks.

Bug: webrtc:9883
Change-Id: I1d77df0f2c68f878eace30e4b037ebc7eb9f1aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162482
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30104}
2019-12-17 10:16:59 +00:00
Artem Titov
8525a8028a Add ability to resize buffers pool in decoder and use it in IVF generator
Bug: webrtc:10138
Change-Id: I452f08f1d9af57de789bd947a1fcb95536845f80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162183
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30098}
2019-12-16 14:51:16 +00:00
Artem Titov
9d06bc2e6d Replace sequence checker with lock in IvfFrameGemerator.
It was found that generator can be destroyed on another thread
comparing to the one, from which frame were generated. It can happen
because generator injected into PC though scoped_ref object and the
last pointer to that object can be destroyed on different thread
depending on machine load.

To fix this sequence checker is replaced with lock. It is required
to ensure that generator won't be destroyed while it is reading frame,
because otherwise it can catch SIGSEGV.

Bug: webrtc:10138
Change-Id: Ia3488bd8ae396c209b90977469593784bb82114b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162182
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30086}
2019-12-13 14:06:30 +00:00
Danil Chapovalov
a3ecb7a656 Migrate tests from RtpDepacketizer to VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: I1b1c5183d35b791c09c14c9d1f0ca240c1749d9a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161455
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30055}
2019-12-10 17:37:46 +00:00
Per Åhgren
62ea0aaea0 Remove deprecated legacy AEC code
This CL removes the deprecated legacy AEC code.

Note that this CL should not be landed before the M80 release has been cut.

Bug: webrtc:11165
Change-Id: I59ee94526e62f702bb9fa9fa2d38c4e48f44753c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161238
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30036}
2019-12-09 10:37:49 +00:00
Artem Titov
33f9d2b383 Migrate WebRTC on FrameGeneratorInterface and remove FrameGenerator class
Bug: webrtc:10138
Change-Id: If85290581a72f81cf60181de7a7134cc9db7716e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161327
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30033}
2019-12-07 00:54:26 +00:00
Artem Titov
895069045f Fix: IvfFrameGenerator won't decode frame on release build
Bug: webrtc:10138
Change-Id: Id0a6328da20bbb841ed3cb013a0d96d8d88c0152
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161446
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30029}
2019-12-06 15:10:49 +00:00
Ivo Creusen
1518fd34d8 Add support for setting a custom NetEqFactory in PeerConnection level tests.
This allows running Peerconnection level tests with a custom NetEqFactory.

Bug: webrtc:11005
Change-Id: If3063cf61a6274a137e4ab74f9ec2665425f21ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161307
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30028}
2019-12-06 12:34:02 +00:00
Sebastian Jansson
cec2433c47 Exposing more features in the network emulation manager API.
Bug: webrtc:9883
Change-Id: I2a687b46e3374db0dd08b0c02dfea1482e6fb89f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161229
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30024}
2019-12-06 08:47:19 +00:00
Minyue Li
cae277959b Introduce InbandComfortNoise RTP header extension.
BUG: webrtc:11085
Change-Id: I9b556a0d67d3c239abc247787103af9e50af4e65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159710
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30014}
2019-12-05 13:35:01 +00:00
Artem Titov
0020226e63 Replace VideoSourceInterface with FrameGeneratorInterface in AddVideoConfig
Replace VideoSourceInterface with FrameGeneratorInterface in
AddVideoConfig in PC quality test fixture.

Bug: webrtc:10138
Change-Id: I6e5fe91d286e0360bfcad1785af1fb1d8f890563
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161239
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30012}
2019-12-05 10:02:22 +00:00
Artem Titov
503d7237ce Introduce FrameGeneratorInterface
Introduce FrameGeneratorInterface to make FrameGenerator API available
for downstream projects.

Bug: webrtc:10138
Change-Id: I4216775e4b8b54c3f1c72d67ffbda31eb082fd7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161234
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30009}
2019-12-04 21:09:57 +00:00
Sebastian Jansson
340af975e9 Always enter yield policy scope using simulated TimeControllers.
This makes the class easier to use at a minor cost of making it slightly
more magic.

Bug: webrtc:9883
Change-Id: If807cfbf046615333c3bcd3b58a001813102a9f9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161231
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30008}
2019-12-04 17:16:32 +00:00
Danil Chapovalov
242a9e0ffe Fuzz RtpPacketizerAv1
Bug: webrtc:11042
Change-Id: Id44699395f6dee9cb3bde84c936573b65ad0d848
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161009
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30007}
2019-12-04 13:52:51 +00:00
Eldar Rello
0095d37137 Replace hostCandidate with address and port in RTCPeerConnectionIceErrorEvent
Bug: chromium:1013564
Change-Id: Ie1bb86ed6a2a7d73fe6ee666f973d809ed05a7ed
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161084
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#30004}
2019-12-04 13:18:22 +00:00
Ilya Nikolaevskiy
648b9d77c7 Implement automatic animation detection in VideoStreamEncoder
If WebRTC-AutomaticAnimationDetectionScreenshare experiment is enabled,
content type is screenshare and degradation preference is BALANCED,
then input resolution is restricted if update_rect of the incoming frames
is the same for considerable amount of time and is big enough.

This entails treating BALANCED degradation preference for screenshare as
MAINTAIN_RESOLUTION in adaptation logic.

Bug: webrtc:11058
Change-Id: I903dddf53fcbd7c8eac6c5b1447225b15fd8fe5f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161097
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30002}
2019-12-04 11:24:31 +00:00
Mirko Bonadei
a9ad36f322 Fix aec3_fuzzer chromium build config.
Dependencies need to use relative paths in order to work in Chromium,
see [1].

[1] - https://ci.chromium.org/p/chromium/builders/try/linux-libfuzzer-asan-rel/334174

TBR: saza@webrtc.org
Bug: None
Change-Id: I50c401e5983fbb501d1da2ad909198261a8cb940
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161300
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30000}
2019-12-04 10:45:52 +00:00
Sebastian Jansson
41462d58b2 Always keep abs send time extension.
This makes the WebRTC-KeepAbsSendTimeExtension field trial
always enabled. This means that we no longer avoid sending the
abs-send-time extension if we have negotiated sending of transport
wide sequence numbers.

The field trial WebRTC-FilterAbsSendTimeExtension is introduced to allow
reverting to the previous behavior.

Bug: webrtc:10234
Change-Id: Ifd9761d84dd1fe79af840f98ad0882a2e5adf0b0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159181
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Konrad Hofbauer <hofbauer@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29999}
2019-12-04 09:49:04 +00:00
Markus Handell
486cc55a02 TimeController: Rename Sleep to AdvanceTime.
This change renames TimeController's Sleep method to AdvanceTime, unifying
the same name with the same semantic as for downstream projects.

Bug: webrtc:11154
Change-Id: Id79bcf0eafcd0b47a76407ba220479d84df5a736
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161092
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29989}
2019-12-03 16:08:54 +00:00
Danil Chapovalov
e3c4884b76 in PacketBuffer::Packet pass payload using smart buffer
Together with RtpDepacketizer refactoring that would reduce
number of memcpy while handling an rtp packet

Bug: webrtc:11152
Change-Id: I6f4e09c93af5e2a9314967a15eac8ced57ec712e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161087
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@{#29985}
2019-12-03 14:55:54 +00:00
Erik Språng
014dd3c9f7 Trials should always be populated in call config.
The trials are always set when a Call instead is created by a
CallFactory, but a lot of unit tests creates instances directly.
This CL updates those call site. There is still a fallback in place
in RtpTransportControllerSend, since there are down-stream usages that
need to be clean up. After that, we'll remove the fallback.

Bug: webrtc:10809
Change-Id: I0aacf0473317bcd64252dd43d93c42de730e2ffa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160408
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29978}
2019-12-03 10:34:55 +00:00
Artem Titov
5256d8bc4b Refactor FrameGenerator to return VideoFrameBuffer with VideoFrame::UpdateRect
Bug: webrtc:10138
Change-Id: I22079e2630bb1f3bb27472795fe923f9143b3401
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161010
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29973}
2019-12-02 17:11:37 +00:00
Sam Zackrisson
b0db98cf06 Fuzz AEC3
This fuzzer fuzzes AEC3 with the default configuration and variable sample rates and channel counts.

Bug: None
Change-Id: I0d178a320b75fc4cc389657fa2b99931f359b517
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160646
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29967}
2019-12-02 09:16:51 +00:00
Sebastian Jansson
efa3f76b08 Moves SampleStats and EventRateCounter to rtc_base/numerics
Bug: webrtc:9883
Change-Id: I53934c86cad3b7cd60bba6c78e5db66c10e5d56a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/159821
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29965}
2019-12-02 07:46:48 +00:00
Artem Titov
9dc209a23a Add ability to disable detailed error message in RTC_CHECKs
Bug: webrtc:11133
Change-Id: I989654f1fb97b476a17956d69ee374406439ea8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160653
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29952}
2019-11-28 17:51:00 +00:00
Danil Chapovalov
5314b13a8d Fix undefined-shift in RtpDepacketizerAv1::AssembleFrame
Bug: chromium:1028348
Change-Id: I824e84138acbf4e73fc21ee8248e29e5cc7a0ba0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160643
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29945}
2019-11-28 11:27:33 +00:00
Artem Titov
8861d02f89 Restore tests that were accidently deleted during refactoring
Bug: None
Change-Id: Ic68282d9879581c2d7d42d5d80c876e252e53c37
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/160920
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29934}
2019-11-27 13:42:58 +00:00