1561 Commits

Author SHA1 Message Date
Niels Möller
1a29a5da84 Delete rtc::Bind
Bug: webrtc:11339
Change-Id: Id53d17bbf37a15f482e9eb9f8762d2000c772dcc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202250
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33099}
2021-01-29 08:24:43 +00:00
Artem Titov
cc8a1f8450 Add API to get current time mode from NetworkEmulationManager
Bug: None
Change-Id: I1aeca7484bab2b9bc28684b055b8f6bb86135327
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203888
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33074}
2021-01-26 14:01:17 +00:00
Artem Titov
08f46909a8 Protect DefaultVideoQualityAnalyzer::peers_ with lock
Protect DefaultVideoQualityAnalyzer::peers_ with lock, because it's now
accessed from multiple threads.

Bug: webrtc:12247
Change-Id: I41932afe678979f6da9e8d0d5fe2e1ffef0fb513
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203880
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33073}
2021-01-26 10:35:47 +00:00
Artem Titov
c57089a97a Add new peer to injector when adding it to analyzer. Removed unused injector
Bug: webrtc:12247
Change-Id: I735f2b69a8239633bfddca48efd45fe4886c1598
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203820
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33072}
2021-01-26 10:08:16 +00:00
Artem Titov
d2dd732d83 Introduce network emulated endpoint optional name for better logging
Change-Id: Iedce88400c6f1e91c30249fb49c7914723da2a8d
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/203141
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33054}
2021-01-21 18:48:04 +00:00
Lahiru Ginnaliya Gamathige
5eb527cf7f Replace sigslot usages with callback list library.
- Replace few sigslot usages in jsep_transport_controller.
- There is still one sigslot usages in this file so keeping the inheritance
and that is the reason for not having a binary size gain in this CL.
- Remaining sigslot will be removed in a separate CL.

Bug: webrtc:11943
Change-Id: Idb8fa1090b037c48eeb62f54cffd3c485cebfcda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190146
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33034}
2021-01-19 12:03:50 +00:00
Mirko Bonadei
9c8dd87636 Fixing WebRTC/Chromium FYI build.
After [2] has landed, some code started to be built on the iOS FYI
bots. Previous attempts to fix are [3] and [4].

Error:
  error: no member named 'SimpleStringBuilder' in namespace 'rtc'

[1] - https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/8155/overview
[2] - https://webrtc-review.googlesource.com/c/src/+/200981
[3] - https://webrtc-review.googlesource.com/c/src/+/202037
[4] - https://webrtc-review.googlesource.com/c/src/+/202038

TBR=landrey@webrtc.org

Bug: None
Change-Id: Ibee99b274f742acf41a837492d215ef45e5d9de4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202240
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33012}
2021-01-16 18:55:09 +00:00
Mirko Bonadei
be93b788c6 Move iOS bundle data for tests inside rtc_include_test (take 2).
In https://webrtc-review.googlesource.com/c/src/+/202037, there was an
unused variable issue, after testing locally with
rtc_include_tests=false, this CL should fix it.

TBR=landrey@webrtc.org

Bug: None
Change-Id: I9bbc5a124a752ce4b520af29ec5bd0f52b6e3d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202038
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33011}
2021-01-16 13:36:19 +00:00
Mirko Bonadei
8ed61858c3 Move iOS bundle data for tests inside rtc_include_test.
This should fix issues like [1] which surfaced after landing [2] which
is not the direct cause though.

[1] - https://ci.chromium.org/ui/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20ios-device/8133/overview
[2] - https://webrtc-review.googlesource.com/c/src/+/200981

TBR=landrey@webrtc.org

Bug: None
Change-Id: Id222bf0df109711a4aaba708786b8f623e50b9f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202037
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33008}
2021-01-16 11:19:09 +00:00
Mirko Bonadei
e5f4c6b8d2 Reland "Refactor rtc_base build targets."
This is a reland of 69241a93fb14f6527a26d5c94dde879013012d2a

Fix: The problem was related to NO_MAIN_THREAD_WRAPPING, which
affects https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/rtc_base/thread.cc;l=257-263;drc=7acc2d9fe3a6e3c4d8881d2bdfc9b8968a724cd5.
The original CL didn't attach the definition of the macro
NO_MAIN_THREAD_WRAPPING when building for Chromium (which doesn't have
to be related to //rtc_base anymore but to //rtc_base:threading).

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

Bug: webrtc:9987
Change-Id: I7cdf49d2aac8357f1f50f90010bf2c2f62fa19f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/202021
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33001}
2021-01-15 17:00:05 +00:00
Andrey Logvin
f9ee0e08ec Add cross trafic emulation api
Bug: webrtc:12344
Change-Id: I958dc4deda4af4576818600c31aecdf48285172f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200981
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32989}
2021-01-15 07:38:17 +00:00
Mirko Bonadei
7acc2d9fe3 Revert "Refactor rtc_base build targets."
This reverts commit 69241a93fb14f6527a26d5c94dde879013012d2a.

Reason for revert: Breaks WebRTC roll into Chromium.

Original change's description:
> Refactor rtc_base build targets.
>
> The "//rtc_base:rtc_base" build target has historically been one of the
> biggest targets in the WebRTC build. Big targets are the main source of
> circular dependencies and non-API types leakage.
>
> This CL is a step forward into splitting "//rtc_base:rtc_base" into
> smaller targets (as originally started in 2018).
>
> The only non-automated changes are (like re-wiring the build system):
> * The creation of //rtc_base/async_resolver.{h,cc} which allows to
>   break a circular dependency (is has been extracted from
>   //rtc_base/net_helpers.{h,cc}).
> * The creation of //rtc_base/internal/default_socket_server.{h,cc} to
>   break another circular dependency.
>
> Bug: webrtc:9987
> Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32941}

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

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

No-Try: True
Bug: webrtc:9987
Change-Id: I1e36ad64cc60092f38d6886153a94f1a58339256
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201840
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32986}
2021-01-14 21:27:38 +00:00
Erik Språng
c12f625938 Adds VideoDecoder::GetDecoderInfo()
This adds a new way to poll decoder metadata.
A default implementation still delegates to the old methods.
Root call site is updates to not use the olds methods.

Follow-ups will dismantle usage of the olds methods in wrappers.

Bug: webrtc:12271
Change-Id: Id0fa6863c96ff9e3b849da452d6540e7c5da4512
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196520
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32976}
2021-01-14 13:33:22 +00:00
Andrey Logvin
b24e720907 Fix inconsistencies in network BUILD.gn file
Bug: webrtc:12344
Change-Id: I885d5a8aa598d6986a8551eb97debb76c20da34f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201720
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32970}
2021-01-14 09:43:26 +00:00
Niels Möller
db7920424c Change PeerConnectionE2EQualityTest to use lambdas instead of rtc::Bind
Bug: webrtc:11339
Change-Id: I17ff9f01ca4039165227ad5c98195baa81a14d79
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201206
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32969}
2021-01-14 09:25:27 +00:00
Andrey Logvin
0e8dd039be Fix cpu usage dvqa test on windows
Windows cpu clock has low accuracy. We need to fake some load to be sure that the clock ticks.

Bug: webrtc:12249
Change-Id: I6c3b2b0e51badd9b7a58391755a37f4d1c28af40
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201540
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32961}
2021-01-13 11:35:25 +00:00
Danil Chapovalov
a782022d6c Use rtc::CopyOnWriteBuffer::MutableData instead of data in fuzzers
Bug: webrtc:12334
Change-Id: I3df42998f5cf5c3b09ad3f6253cab34170d725bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201380
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32947}
2021-01-12 12:36:43 +00:00
Niels Möller
a68bfc5537 Delete KeepBufferRefs helpers, and use of rtc::Bind.
The rtc::Bind usages are replaced with lambdas with copy-capture
of the ref pointers.

Bug: webrtc:11339
Change-Id: I2fb544fcd2780feac3d725993c360df91899b532
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/201201
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32946}
2021-01-12 12:35:38 +00:00
Mirko Bonadei
69241a93fb Refactor rtc_base build targets.
The "//rtc_base:rtc_base" build target has historically been one of the
biggest targets in the WebRTC build. Big targets are the main source of
circular dependencies and non-API types leakage.

This CL is a step forward into splitting "//rtc_base:rtc_base" into
smaller targets (as originally started in 2018).

The only non-automated changes are (like re-wiring the build system):
* The creation of //rtc_base/async_resolver.{h,cc} which allows to
  break a circular dependency (is has been extracted from
  //rtc_base/net_helpers.{h,cc}).
* The creation of //rtc_base/internal/default_socket_server.{h,cc} to
  break another circular dependency.

Bug: webrtc:9987
Change-Id: I0c8f5e7efe2c8fd8e6bffa0d6dd2dd494cf3df02
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196903
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32941}
2021-01-11 18:32:30 +00:00
philipel
360da05ed1 Remove webrtc::VideoDecoder::PrefersLateDecoding.
This is just general cleanup.

The assumed behavior is late decoding, and this function is not used to make any decision (except in the deprecated jitter buffer).

Bug: webrtc:12271
Change-Id: Ifb48186d55903f068f25e44c5f73e7a724f6f456
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200804
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32940}
2021-01-11 18:02:25 +00:00
Artem Titov
1c5e63e545 Add module documentation for emulated network
Bug: webrtc:12208
Change-Id: I28d5f349706751d1762b90527601eaa86906e42d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200803
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32938}
2021-01-11 16:57:24 +00:00
Danil Chapovalov
e15dc58f32 Use rtc::CopyOnWriteBuffer::MutableData through webrtc
where mutable access is required.

Bug: webrtc:12334
Change-Id: I4b2b74f836aaf7f12278c3569d0d49936297716b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198846
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32936}
2021-01-11 11:31:33 +00:00
Niels Möller
6afa794b6e Delete deprecated H264BitstreamParser methods
Bug: webrtc:10439
Change-Id: I1513907f03f9adfcf5657298e69d60519af764ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/198121
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32934}
2021-01-11 09:31:54 +00:00
Artem Titov
ec9b281bbc Add ability to specify random seed when creating built it network emulation
Bug: webrtc:12340
Change-Id: Iffd054928249099866ef4527b911b1e358e26f5a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/200805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32920}
2021-01-07 16:40:50 +00:00
Christoffer Rodbro
8649e49d10 Add a field trial to skip REMB modification of BWE internal state.
By enabling the field trial, REMB caps the output target bitrate, but
does not change any internal BWE state variables.

Bug: webrtc:12306
Change-Id: I43e9ac1d1b7dff292d7aa5800c01d874bc91aaff
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197809
Reviewed-by: Jakob Ivarsson <jakobi@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32867}
2020-12-21 10:49:06 +00:00
Mirko Bonadei
5686e3457e Optimize calls to std::string::find() and friends for a single char.
The character literal overload is more efficient.

No-Presubmit: True
No-Try: True
Bug: None
Change-Id: Ice0b8478accd8a252ab81a0496d46c0f71db3db6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197810
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32841}
2020-12-16 09:01:44 +00:00
Erik Språng
ebe5acb27a VideoCodecTextFixture and YuvFrameReader improvements.
Adds ability to specify desired frame size separate from actual clip
resolution, as well as clip and desired fps.
This allows e.g. reading an HD clip but running benchmarks in VGA, and
to specify e.g. 60fps for the clip but 30for encoding where frame
dropping kicks in so that motion is actually correct rather than just
plaing the clip slowly.

Bug: webrtc:12229
Change-Id: I4ad4fcc335611a449dc2723ffafbec6731e89f55
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/195324
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32839}
2020-12-15 23:18:06 +00:00
Mirko Bonadei
9ff23bad06 Fix order-dependent tests.
webrtc::test::GetPerfResults() relies on a singleton and this makes
some tests be order dependent (running in a different order makes them
fail).

A good fix is to remove the singleton but this CL at least makes the
fragile test set up the environment correctly.

No-Try: True
Bug: None
Change-Id: I7ad25f685f0bc5d246beeadedfa9f5a39f3547e5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/197425
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32823}
2020-12-14 12:59:52 +00:00
Andrey Logvin
b95d90b78a Rename UNIT_TEST to WEBRTC_UNIT_TEST
Current name conflicts with upstream project code.

Bug: webrtc:12247
Change-Id: Ibd78273a75262772fc18fca688c29b9ba9525ce5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196653
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32813}
2020-12-10 11:04:58 +00:00
Andrey Logvin
d7808f1c46 Add DVQA support for scenarios with new participants joining
Bug: webrtc:12247
Change-Id: Id51a2ab34e0b802e11931cad13f48ce8eefddcae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196361
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32804}
2020-12-08 18:24:08 +00:00
Andrey Logvin
6efb0310ec Set default max_threads_count to DVQA injection helper
Bug: webrtc:12247
Change-Id: I608cffad7ee5397c306fb03a36d89f31882c112c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196092
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32748}
2020-12-02 18:49:22 +00:00
Andrey Logvin
9c296e5b9e Fix DVQA cpu usage when Stop is called multiple times
Bug: webrtc:12247
Change-Id: I946338e0ecf58f91c87c8638977a8bc52e648fd2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196083
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32747}
2020-12-02 18:48:17 +00:00
Andrey Logvin
8dbbd648e7 Revert "Ignore frames that are comming to DVQA after Stop is called"
This reverts commit 8d4cdd11d8d4ce3e6ddbe9c729c7cfbd8f495880.

Reason for revert: Upstream project needs have changed

Original change's description:
> Ignore frames that are comming to DVQA after Stop is called
>
> Bug: webrtc:12247
> Change-Id: Ie3e773bdff66c900956019ac3131bbdb9ee874cd
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196084
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Andrey Logvin <landrey@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32738}

TBR=mbonadei@webrtc.org,srte@webrtc.org,landrey@webrtc.org

Change-Id: Ie7483435eae9b0344f875673ca9651ff4d591bd3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:12247
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196280
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32746}
2020-12-02 18:42:58 +00:00
Andrey Logvin
e301c18eb7 Add landrey as an owner to PC framework code
While Artem is OOO there is no owner over PC framework changes that can appove CLs.

Bug: webrtc:12247
Change-Id: I70aa5e1263efa9c0971a077ecbb247a7c41991cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196091
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32743}
2020-12-02 16:34:41 +00:00
Andrey Logvin
8d4cdd11d8 Ignore frames that are comming to DVQA after Stop is called
Bug: webrtc:12247
Change-Id: Ie3e773bdff66c900956019ac3131bbdb9ee874cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/196084
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32738}
2020-12-02 09:22:14 +00:00
Jakob Ivarsson
47a03e8743 Default enable sending transport sequence numbers on audio packets.
This enables send side bandwidth estimation for audio and removes field
trial "WebRTC-Audio-SendSideBwe" which this was controlled through.

Transport-cc extension still needs to be negotiated.

Bug: webrtc:12222
Change-Id: Ie2268fad13703eeb0f0d38fcf484baaa29715b7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/194142
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32681}
2020-11-24 09:19:54 +00:00
Mirko Bonadei
01719fbeb5 Reland "Rename FATAL() into RTC_FATAL()."
This is a reland of 9653d26f8e83bb685477e7ba5c2adf2863187743

Original change's description:
> Rename FATAL() into RTC_FATAL().
>
> No-Try: True
> Bug: webrtc:8454
> Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32620}

No-Try: True
Bug: webrtc:8454
Change-Id: Idb80125ac31ea307d1434bc9a65f148ac2017a3c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193864
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32635}
2020-11-18 20:49:08 +00:00
Mirko Bonadei
a4fd641f51 Revert "Rename FATAL() into RTC_FATAL()."
This reverts commit 9653d26f8e83bb685477e7ba5c2adf2863187743.

Reason for revert: Breaks downstream project.

Original change's description:
> Rename FATAL() into RTC_FATAL().
>
> No-Try: True
> Bug: webrtc:8454
> Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32620}

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

Change-Id: I0ad01bcac60c87b30bd4575a9d631e7dd8f34992
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8454
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193863
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32627}
2020-11-18 07:03:54 +00:00
Jonas Oreland
97050115f0 Add TURN server to Emulated Network infrastructure
This can be used to test ICE behavior.

Bug: chromium:1024965
Change-Id: Ie4ba9cd5c3cf3c2f71bab3637f925263dbc6296e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193701
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32625}
2020-11-17 21:07:56 +00:00
Mirko Bonadei
9653d26f8e Rename FATAL() into RTC_FATAL().
No-Try: True
Bug: webrtc:8454
Change-Id: I9130487a92463a2128cf1493e6c5117b2fab313a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/193703
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32620}
2020-11-17 16:12:40 +00:00
Danil Chapovalov
62a9a32937 In Av1 packetizer set marker bit with respect of end_of_picture flag
Bug: webrtc:12167
Change-Id: If14fdd7144951c7aa7e48efd390637dd66201bf7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192791
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32612}
2020-11-16 11:08:48 +00:00
Artem Titov
5d55597932 Add support for loopback route on emulated endpoints
Bug: b/172995851
Change-Id: I70b5ec6cd84784dcc452e8f96a02f4be849fa0f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/192920
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32596}
2020-11-12 17:01:59 +00:00
Sergey Silkin
b72cc6d670 Analyze quality of dropped frames in VideoProcessor.
Calculate quality metrics for dropped frames by comparing original
frame against last decoded one.

This feature makes comparison of encoders which do/don't drop frames
more fair.

The feature is controlled by analyze_quality_of_dropped_frames flag
and is disabled by default.

Bug: none
Change-Id: Ifab8df92d0b76e743ff3193c05d7c8dbd14921c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190660
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32518}
2020-10-29 08:23:49 +00:00
philipel
1b0d5437c9 Removed _completeFrame since we never allow incomplete frames.
In the old jitter buffer the two VCMVideoProtection modes |kProtectionNone| and |kProtectionFEC| could be set on the jitter buffer for it to not wait for NACK and instead generate incomplete frames. This has not been possible for a long time.

Bug: webrtc:9378, webrtc:7408
Change-Id: I0a2d3ec34d721126c1128306d5fad88314f8d59f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32513}
2020-10-28 16:00:27 +00:00
Åsa Persson
17b29b9121 test::CreateVideoStreams: Use default unconfigured VideoStream if layer is missing in config.
Configure framerate/temporal layers via VideoEncoderConfig in VideoStreamEncoderTest..

Bug: none
Change-Id: I1104da5e576fa25746f2f2f5eaa336cd17c0093a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187488
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32500}
2020-10-27 08:19:57 +00:00
Niels Möller
7c85d395d7 Delete unneeded includes of system_wrappers/include/sleep.h
Non-test usage is in modules/audio_device and modules/desktop_capture.

Bug: None
Change-Id: Ie7dd89aa40e6dcfa9e49e1956b87b50fd9f1c227
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/190140
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32484}
2020-10-26 09:55:26 +00:00
Erik Språng
9d69cbeabf Changes default pacing factor to 1.1x
This changes the default behavior to use pacing factor of 1.1x instead
of 2.5x, it also sets libvpx rate controler as trusted, turns on the
encoder pushback mechanism and sets spatial hysteresis to 1.2.
The unused "dynamic rate" settings in libvpx is removed.

The new settings matches what has been used in chromium since 2019.
If needed, the legacy behavior can be enabled using the field trial
WebRTC-VideoRateControl.

Bug: webrtc:10155
Change-Id: I8186b491aa5bef61e8f568e96c980ca68f0c208f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186661
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32477}
2020-10-23 13:43:32 +00:00
Danil Chapovalov
9f4859e5e3 Allow to set av1 scalability mode after encoder is constructed
Bug: webrtc:11404
Change-Id: I70b4115c8afdc4f32fd876d31d54b7d95d0a7e1b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/188582
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32437}
2020-10-19 10:42:23 +00:00
Erik Språng
279f37052c Makes WebRTC-Pacer-SmallFirstProbePacket default enabled.
This is expected to yield slightly higher bandwidth estimates when
probing is used, since it reduces a bias in how packet sizes are counted.

Bug: webrtc:11780
Change-Id: I6a4a3af0c50670d248dbe043a4d9da60915e3699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187491
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32394}
2020-10-13 21:45:42 +00:00
Guido Urdaneta
ff7913204c Revert "Reland "Replace sigslot usages with robocaller library.""
This reverts commit c5f71087589b18bb4df1b78f2c452c4083edf2d9.

Reason for revert: Causes Chromium WPT Tests to fail, preventing rolls.

Sample failed run:
https://ci.chromium.org/p/chromium/builders/try/linux-rel/511995?

Sample logs:

STDERR: # Fatal error in: ../../third_party/webrtc/pc/peer_connection.cc, line 575
STDERR: # last system error: 0
STDERR: # Check failed: (signaling_thread())->IsCurrent()
STDERR: # Received signal 6
STDERR: #0 0x7f81d39e3de9 base::debug::CollectStackTrace()
STDERR: #1 0x7f81d38f9ca3 base::debug::StackTrace::StackTrace()
STDERR: #2 0x7f81d39e393b base::debug::(anonymous namespace)::StackDumpSignalHandler()
STDERR: #3 0x7f81c9054140 (/lib/x86_64-linux-gnu/libpthread-2.31.so+0x1413f)
STDERR: #4 0x7f81c8d72db1 gsignal
STDERR: #5 0x7f81c8d5c537 abort
STDERR: #6 0x7f81c7344032 rtc::webrtc_checks_impl::FatalLog()
STDERR: #7 0x7f81c722e5c0 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #8 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #9 0x7f81c72d6e8e webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #10 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #11 0x7f81c71c6df3 webrtc::webrtc_function_impl::CallHelpers<>::CallInlineStorage<>()
STDERR: #12 0x7f81c7348d99 webrtc::robo_caller_impl::RoboCallerReceivers::Foreach()
STDERR: #13 0x7f81c73135bc rtc::OpenSSLStreamAdapter::ContinueSSL()
STDERR: #14 0x7f81c7312fd4 rtc::OpenSSLStreamAdapter::OnEvent()
STDERR: #15 0x7f81c71c30d9 cricket::StreamInterfaceChannel::OnPacketReceived()
STDERR: #16 0x7f81c71c640a cricket::DtlsTransport::OnReadPacket()
STDERR: #17 0x7f81c71cad61 cricket::P2PTransportChannel::OnReadPacket()
STDERR: #18 0x7f81c71bc90f cricket::Connection::OnReadPacket()
STDERR: #19 0x7f81c71e6255 cricket::UDPPort::HandleIncomingPacket()
STDERR: #20 0x7f81cd1f1bff blink::(anonymous namespace)::IpcPacketSocket::OnDataReceived()
STDERR: #21 0x7f81cd1f645d blink::P2PSocketClientImpl::DataReceived()
STDERR: #22 0x7f81cd50a16b network::mojom::blink::P2PSocketClientStubDispatch::Accept()
STDERR: #23 0x7f81d2b4f642 mojo::InterfaceEndpointClient::HandleValidatedMessage()
STDERR: #24 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #25 0x7f81d2b50bb1 mojo::InterfaceEndpointClient::HandleIncomingMessage()
STDERR: #26 0x7f81d2b58a3a mojo::internal::MultiplexRouter::ProcessIncomingMessage()
STDERR: #27 0x7f81d2b57f7f mojo::internal::MultiplexRouter::Accept()
STDERR: #28 0x7f81d2b5304b mojo::MessageDispatcher::Accept()
STDERR: #29 0x7f81d2b48851 mojo::Connector::DispatchMessage()
STDERR: #30 0x7f81d2b492e7 mojo::Connector::ReadAllAvailableMessages()
STDERR: #31 0x7f81d2b490a3 mojo::Connector::OnHandleReadyInternal()
STDERR: #32 0x7f81d2b498f0 mojo::SimpleWatcher::DiscardReadyState()
STDERR: #33 0x7f81d2d0e67a mojo::SimpleWatcher::OnHandleReady()
STDERR: #34 0x7f81d2d0eb2b base::internal::Invoker<>::RunOnce()
STDERR: #35 0x7f81d397f85b base::TaskAnnotator::RunTask()
STDERR: #36 0x7f81d399a04c base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWorkImpl()
STDERR: #37 0x7f81d3999c78 base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::DoWork()
STDERR: #38 0x7f81d391fe64 base::MessagePumpDefault::Run()
STDERR: #39 0x7f81d399a8dc base::sequence_manager::internal::ThreadControllerWithMessagePumpImpl::Run()
STDERR: #40 0x7f81d395ae55 base::RunLoop::Run()
STDERR: #41 0x7f81d39c87f2 base::Thread::Run()




Original change's description:
> Reland "Replace sigslot usages with robocaller library."
>
> This is a reland of 40261c3663fe316cfe40262c59cee993165ccf63
>
> Note: Instead of changing the type of JsepTransportController->SignalSSLHandshakeError
> added a new member with a different name and used it in webrtc code.
> After this change do two more follow up CLs to completely remove the old code
> from google3.
>
> Original change's description:
> > Replace sigslot usages with robocaller library.
> >
> > - Replace all the top level signals from jsep_transport_controller.
> > - There are still sigslot usages in this file so keep the inheritance
> >   and that is the reason for not having a binary size gain in this CL.
> >
> > Bug: webrtc:11943
> > Change-Id: I249d3b9710783aef70ba273e082ceeafe3056898
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/185540
> > Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#32321}
>
> Bug: webrtc:11943
> Change-Id: Ia07394ee395f94836f6b576c3a97d119a7678e1a
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/186946
> Commit-Queue: Lahiru Ginnaliya Gamathige <glahiru@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#32359}

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

Change-Id: I6bce1775d10758ac4a9d05b855f473dd70bd9815
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11943
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/187487
Reviewed-by: Guido Urdaneta <guidou@webrtc.org>
Commit-Queue: Guido Urdaneta <guidou@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#32372}
2020-10-09 18:07:56 +00:00