1633 Commits

Author SHA1 Message Date
Sebastian Jansson
d7fade5738 Makes all units and operations constexpr
Since RTC_DCHECK was made constexpr compatible, we can now
make the unit classes fully constexpr.

Bug: webrtc:9883
Change-Id: I18973c2f318449869cf0bd45699c41be53fba806
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167722
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30403}
2020-01-29 10:57:54 +00:00
Yves Gerey
071d025929 Activate event tracing for unit tests. For good!
The --trace_event=file.json option allows to log events,
for further inspection in chromium event viewer.

Previous handling of this option was broken,
closing the logger before the tests were even run.

Bug: webrtc:10926
Change-Id: I9123d12666b5f254feeaef685def96eb8ba1c7f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167720
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30401}
2020-01-29 10:11:34 +00:00
philipel
52c62df2ed Don't condition the time_controller target on rtc_include_tests.
Bug: none
Change-Id: Ifb3f811c71a778a447c41593902c417614ae9824
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167723
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30400}
2020-01-29 09:59:34 +00:00
Sebastian Jansson
17a6381c1c Adds fake video codec mode to PeerScenarioClient
This improves execution speed by skipping the encoding step.

Bug: webrtc:10365
Change-Id: I6aef1376c157d859f05f4a44f881d1c60f353067
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167082
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30385}
2020-01-27 18:07:45 +00:00
Artem Titov
1e02339ea6 Add ability to set custom adapter type on emulated endpoint
Bug: webrtc:10138
Change-Id: I2f53b42a2c377c9c0c9d36b61eb1c6ce96da480a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167209
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30371}
2020-01-24 12:53:07 +00:00
Sebastian Jansson
7aa2edf936 Adds CreateTimeControllerBasedCallFactory.
Bug: webrtc:11255
Change-Id: I9614823761ff5d2eb4fe03342f255a81087b6449
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166960
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30358}
2020-01-23 10:29:30 +00:00
Sebastian Jansson
094ce2ef83 Adds CreateTaskQueueFactory to TimeController
Bug: webrtc:11255
Change-Id: I02bdc944c7081590f40a77b315f64c63adbc6ff8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166921
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30349}
2020-01-22 14:19:15 +00:00
Artem Titov
ee558dcca8 Propagate multicodec support to other places of PC level framework
Bug: webrtc:10138
Change-Id: I9258db991053abfa40f2a5112eddfa7f3e0d41a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167062
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30346}
2020-01-22 13:34:18 +00:00
Sebastian Jansson
6ce033a863 Moves ownership of time controller into NetworkEmulationManager.
This makes it easier to maintain consistency between real time
and simulated time modes.

The RealTimeController is updated to use an explicit main thread,
this ensures that pending destruction tasks are run as the network
emulator goes out of scope.

Bug: webrtc:11255
Change-Id: Ie73ab778c78a68d7c58c0f857f14a8d8ac027c67
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166164
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30342}
2020-01-22 11:12:27 +00:00
Mirko Bonadei
ccbe95fd8a Reformat GN files.
`gn format` recently [1] changed its formatting behavior
for deps, source, and a few other elements when they
are assigned (with =) single-element lists to be consistent
with the formatting of updates (with +=) with single-element.

Now that we've rolled in a GN binary with the change,
reformat all files so that people don't get presubmit
warnings due to this.

CL generated with:
$ git ls-files | grep BUILD.gn | xargs gn format
$ gn format build_overrides/build.gni
$ gn format build_overrides/gtest.gni
$ gn format modules/audio_coding/audio_coding.gni
$ gn format webrtc.gni
$ gn format .gn

Plus a few manual changes to add exceptions for
"public_deps" (after changing these lines the presubmit
started to complain).

[1] - https://gn-review.googlesource.com/c/gn/+/6860

Bug: webrtc:11302
Change-Id: Iac29d23c1618ebef925c972e2891cd9f4e8cd613
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166882
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30334}
2020-01-21 12:13:11 +00:00
Ivo Creusen
cee751abff Reland "Enable using a custom NetEqFactory in simulations"
This is a reland of 2a11b2451a4068746fa0c55fa210efd4a15e4423
There are no changes compared to the first attempt.

Original change's description:
> Enable using a custom NetEqFactory in simulations
>
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}

TBR=kwiberg

Bug: webrtc:11005
Change-Id: I4aa377e05916bd23f8f63aece9d0e27731c80d3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166465
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30319}
2020-01-20 12:46:34 +00:00
Artem Titov
9fbe9ae1c1 Add support of negotiating multiple codecs in PC framework
Bug: webrtc:10138
Change-Id: Iec7df60a4185a039bd81de200c0691747e92c10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166601
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30318}
2020-01-20 12:13:04 +00:00
Sebastian Jansson
274cc7fadf Adds current thread to yielders in SimulatedThread::SendTask.
Bug: webrtc:11255
Change-Id: Ib65b902b60b15f402fac51269c74ac46b56cabc5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166462
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30304}
2020-01-17 15:08:53 +00:00
Ilya Nikolaevskiy
db6ca7f2d7 Add safety checks in RtpPacket::ZeroMutableExtensions and fuzz it
Bug: chromium:1042535
Change-Id: I0f7ef1086631b5beb2e0c89d57534d2551289117
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166441
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30303}
2020-01-17 14:22:04 +00:00
Sebastian Jansson
77bd385b55 Using EmulatedEndpoint in Scenario tests.
Bug: webrtc:9883
Change-Id: I7d1dc9d8efbdddc14e1fbe08d7b6a71c4bbe24ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166341
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30300}
2020-01-17 12:50:20 +00:00
Sebastian Jansson
c9f42ad909 Simplifies transport overhead mechanism in Scenario test framework.
This changes the behavior for adding virtual transport overhead so it
doesn't change the size of the actual payload buffer, only the
calculated packet size.

Bug: webrtc:9883
Change-Id: I6e24598378c4dd6a591d36ca3b162e933ff4ef7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164523
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30298}
2020-01-17 11:30:02 +00:00
Per Åhgren
0695df1a59 Reland "Replace the ExperimentalAgc config with the new config format"
This is a reland of f3aa6326b8e21f627b9fba72040122723251999b

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}

Bug: webrtc:5298
Change-Id: I6db03628ed3fa2ecd36544fe9181dd8244d7e2df
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165760
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30295}
2020-01-17 10:09:09 +00:00
Sebastian Jansson
fc8279d66c Reland "Using simulated rtc::Thread for peer connection scenario tests."
This is a reland of b70c5c5ce97e7dcf2e1d8453f5ea0639d4b60453

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}

Bug: webrtc:11255
Change-Id: If65cd56b59158cebec5609407a721fbdb47cfd1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166046
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30294}
2020-01-17 09:22:18 +00:00
Sandeep Siddhartha
3f0bc2c176 Revert "Enable using a custom NetEqFactory in simulations"
This reverts commit 2a11b2451a4068746fa0c55fa210efd4a15e4423.

Reason for revert: Causes b/147826709

Original change's description:
> Enable using a custom NetEqFactory in simulations
> 
> Bug: webrtc:11005
> Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
> Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30286}

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

Change-Id: I14a0bd6ad2a90f1686b8b1a78f18aea9325871fe
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11005
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166403
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Sandeep Siddhartha <sansid@google.com>
Cr-Commit-Position: refs/heads/master@{#30288}
2020-01-16 22:56:21 +00:00
Ivo Creusen
2a11b2451a Enable using a custom NetEqFactory in simulations
Bug: webrtc:11005
Change-Id: I8a15f77953cbd3c29a75c7cfc77f926b138994b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165580
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30286}
2020-01-16 18:26:44 +00:00
Jonas Oreland
c7bce99540 Make it possible to inject IceTransport in pc quality test fixture
Bug: chromium:1024965
Change-Id: I55296a31e1638c8c00bd6c53151fc4898202b033
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166168
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30279}
2020-01-16 11:56:50 +00:00
Danil Chapovalov
61d6471912 Change H264 depacketizer to implement VideoRtpDepacketizer interface
Bug: webrtc:11152
Change-Id: If5169f47d85918356fa66e2bf3422d722044aa1f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165581
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30264}
2020-01-15 12:26:55 +00:00
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