31 Commits

Author SHA1 Message Date
Markus Handell
f70fbc8411 Introduces rtc_base/synchronization/mutex.h.
This change introduces a new non-reentrant mutex to WebRTC. It
enables eventual migration to Abseil's mutex.

The mutex types supportable by webrtc::Mutex are

- absl::Mutex
- CriticalSection (Windows only)
- pthread_mutex (POSIX only)

In addition to introducing the mutexes, the CL also changes
PacketBuffer to use the new mutex instead of rtc::CriticalSection.

The method of yielding from critical_section.cc was given a
mini-cleanup and YieldCurrentThread() was added to
rtc_base/synchronization/yield.h/cc.

Additionally, google_benchmark benchmarks for the mutexes were added
(test courtesy of danilchap@), and some results from a pthread/Abseil
shootout were added showing Abseil has the advantage in higher
contention.

Bug: webrtc:11567, webrtc:11634
Change-Id: Iaec324ccb32ec3851bf6db3fd290f5ea5dee4c81
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176230
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31443}
2020-06-04 09:55:12 +00:00
Artem Titov
68063a25de Move media configuration for PC level tests into separate class
Bug: webrtc:11479
Change-Id: I325e5c6f5d571dde0fdb5d579bf85cf32a81e174
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172783
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30985}
2020-04-02 15:51:50 +00:00
Artem Titov
b907f1f9f8 Extract test peer creation into separate file
Extract test peer creation into separate file to simplify code and
increase readability. Also it is 1st step in bigger refactoring of PC
level test fixture implementation to make it more granular and reusable.

Change-Id: I687a17bda33a8eebc1ef0ddc0d54572e095fd709
Bug: webrtc:11479
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/172628
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30980}
2020-04-02 12:44:05 +00:00
Artem Titov
8041b651a3 Add YUV to IVF video converter util
Bug: webrtc:10138
Change-Id: I79ca08c45a664c66b15a1ed0c1322719c9f5574d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161449
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30559}
2020-02-19 14:44:21 +00:00
Artem Titov
89e7fcb726 Revert "Enable capturing from camera in PC framework"
This reverts commit 482d26ce9d2b676ca277ca3f44a5d89105627cce.

Reason for revert: Reduced amount of captured frames on some devices. Will require deeper look on it.

Original change's description:
> Enable capturing from camera in PC framework
> 
> Bug: webrtc:10138
> Change-Id: I6b2eaddf4975ddc7237932511de06744ef962489
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154357
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#29318}

TBR=ilnik@webrtc.org,kwiberg@webrtc.org,titovartem@webrtc.org

Change-Id: Ie9db3b1a13fa6ebfd8e277b68b5d808533a84620
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154560
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29320}
2019-09-26 12:00:01 +00:00
Artem Titov
482d26ce9d Enable capturing from camera in PC framework
Bug: webrtc:10138
Change-Id: I6b2eaddf4975ddc7237932511de06744ef962489
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/154357
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#29318}
2019-09-26 11:42:29 +00:00
Danil Chapovalov
40dc98a117 Print stack trace when a test crash
That should simplify debugging failures caused by RTC_DCHECK/RTC_CHECK

TBR=kwiberg@webrtc.org [to pass DEPS change check]

Bug: None
Change-Id: Idb8be3b79309b4ad0baf5ae480356aa5c8cfe361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148802
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28832}
2019-08-12 15:36:20 +00:00
Artem Titov
ef3fd9c8ad Add support for simulcast with Vp8 from caller into PC level quality tests.
Add support of negotiating simulcast offer/answer. Also fix some minor
issues around to make it finally work.

Bug: webrtc:10138
Change-Id: I382f5df04ca6ac04d8ed1e030e7b2ae5706dd10c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137425
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28274}
2019-06-13 17:27:09 +00:00
Mirko Bonadei
60f14ce217 Do not use absl::flat_hash_map in DefaultVideoQualityAnalyzer.
This CL removes the usage of absl::flat_hash_map because it transitively
depends on CCTZ which fails to link with lld-link after the switch to
libc++.

Since std::map doesn't support heterogeneous lookup until C++14, this
CL also stops using absl::string_view and switches to
`const std::string&`.

Bug: webrtc:10605
Change-Id: I4fc93969c6fc0cc7e7e62b4d2f801bdd27cff0f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135566
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27877}
2019-05-08 10:23:59 +00:00
Mirko Bonadei
4487ac4a53 Reland "Add Video Bwe stats collection to DefaultVideoQualityAnalyzer."
This is a reland of 8848229234aae01ec19582ece7b748d557119d66

Original change's description:
> Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
>
> This CL adds the possibility to collect the following Video BWE stats:
> - available_send_bandwidth
> - transmission_bitrate
> - retransmission_bitrate
> - actual_encode_bitrate
> - target_encode_bitrate
>
> Example of the output:
>
> RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
> RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond
>
> Bug: webrtc:10138
> Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27760}

TBR=tommi@webrtc.org

Bug: webrtc:10138
Change-Id: Ib76dfeca741134d6f18ae0eb436920ead42a1d42
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134543
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27856}
2019-05-06 06:32:48 +00:00
Artem Titov
e680c83a41 Revert "Add Video Bwe stats collection to DefaultVideoQualityAnalyzer."
This reverts commit 8848229234aae01ec19582ece7b748d557119d66.

Reason for revert: break chromium compilation on iOS
https://logs.chromium.org/logs/chromium/buildbucket/cr-buildbucket.appspot.com/8915214519549611184/+/steps/compile/0/stdout

Original change's description:
> Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
> 
> This CL adds the possibility to collect the following Video BWE stats:
> - available_send_bandwidth
> - transmission_bitrate
> - retransmission_bitrate
> - actual_encode_bitrate
> - target_encode_bitrate
> 
> Example of the output:
> 
> RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
> RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
> RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
> RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
> RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
> RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond
> 
> Bug: webrtc:10138
> Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Artem Titov <titovartem@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@google.com>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27760}

TBR=mbonadei@webrtc.org,mbonadei@google.com,ilnik@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org

Change-Id: Ib0ef94331410d9b22b6425e4da412b75360fa2d9
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10138
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134210
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27771}
2019-04-25 13:39:04 +00:00
Mirko Bonadei
8848229234 Add Video Bwe stats collection to DefaultVideoQualityAnalyzer.
This CL adds the possibility to collect the following Video BWE stats:
- available_send_bandwidth
- transmission_bitrate
- retransmission_bitrate
- actual_encode_bitrate
- target_encode_bitrate

Example of the output:

RESULT available_send_bandwidth: smoke_test/alice= {487754.33,87583.093} bytesPerSecond
RESULT transmission_bitrate: smoke_test/alice= {465779.17,212075.5} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/alice= {20036,26326.751} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/alice= {418779.33,200486.03} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/alice= {469491.17,77866.909} bytesPerSecond
RESULT available_send_bandwidth: smoke_test/bob= {642924.83,168842.34} bytesPerSecond
RESULT transmission_bitrate: smoke_test/bob= {626115.5,294783.56} bytesPerSecond
RESULT retransmission_bitrate: smoke_test/bob= {0,0} bytesPerSecond
RESULT actual_encode_bitrate: smoke_test/bob= {594235.33,297289.54} bytesPerSecond
RESULT target_encode_bitrate: smoke_test/bob= {640463.5,167676.66} bytesPerSecond

Bug: webrtc:10138
Change-Id: I0414055af0010b8fb4d909297e6da86d398157c2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/132703
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@google.com>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27760}
2019-04-25 09:37:54 +00:00
Sebastian Jansson
f7f9845d9e Adds modules/utility to test/DEPS.
This will be used in an upcoming CL.

Bug: webrtc:10365
Change-Id: Ic5f44fdb7579de994dd0896116573de6a46dfc00
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/128401
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27211}
2019-03-20 14:48:15 +00:00
Artem Titov
bf9e01ab4e Add support of fast media sending in peer connection e2e test
Start sending media from the peer when it's ICE connection state is
connected.

Bug: webrtc:10138
Change-Id: I9f5a1cd917317a3ebadd7c156563035b0bbecf2a
Reviewed-on: https://webrtc-review.googlesource.com/c/121956
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26698}
2019-02-15 00:51:03 +00:00
Artem Titov
a6a273db11 Introduce PeerConnectionE2EQualityTestFixture implementation.
Introduce PeerConnectionE2EQualityTestFixture implementation with
example test.

Bug: webrtc:10138
Change-Id: Iec1d135f1b43863a3fa6f0723b579d2b7ff44807
Reviewed-on: https://webrtc-review.googlesource.com/c/120810
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26589}
2019-02-07 17:07:39 +00:00
Artem Titov
0774bd9583 Introduce network layer.
This CL contains network emulation layer and is a first part of landing
CL https://webrtc-review.googlesource.com/c/src/+/116663

Bug: webrtc:10138
Change-Id: If664b21e9df847aef8144d622d08fc7e9f6608da
Reviewed-on: https://webrtc-review.googlesource.com/c/120406
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26470}
2019-01-30 15:45:10 +00:00
Artem Titov
840b05587f Introduce TestPeer.
TestPeer represent single participant in the call and will own most
required for call objects.

TestPeer::CreateTestPeer is responsible for full setup of TestPeer and
allow to correctly inject media analyzers into call.

Bug: webrtc:10138
Change-Id: Ide7062004b0dc113b9c05181d8144797a3cc27a8
Reviewed-on: https://webrtc-review.googlesource.com/c/119941
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Peter Slatala <psla@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26464}
2019-01-30 10:05:54 +00:00
Steve Anton
10542f21c8 (4) Rename files to snake_case: update BUILD.gn, include paths, header guards, and DEPS entries
Mechanically generated by running this command:

tools_webrtc/do-renames.sh update all-renames.txt && git cl format

Then manually updating:

tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc

Bug: webrtc:10159
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833
Reviewed-on: https://webrtc-review.googlesource.com/c/115653
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26226}
2019-01-11 17:11:39 +00:00
Sebastian Jansson
98b07e9180 Adds scenario test framework.
Bug: webrtc:9510
Change-Id: I387aab4211f520a1c54832f82032ee724479e89e
Reviewed-on: https://webrtc-review.googlesource.com/89342
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24864}
2018-09-27 12:31:33 +00:00
Qingsi Wang
558b93b3e9 Add the multicast DNS message format.
This CL adds the utilities to generate and parse mDNS messages (RFC 1035
and RFC 6762).

TBR=phoglund@webrtc.org

Bug: webrtc:9605
Change-Id: Id6121c17926887cd3a41a2dfc829462fd15f3a4c
Reviewed-on: https://webrtc-review.googlesource.com/93241
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24505}
2018-08-31 00:02:44 +00:00
Sebastian Jansson
50eb4c44ad Adds BBR field trial to CallTest.
Bug: webrtc:8415
Change-Id: Ie0db059390fe4e079f1faa90f74f4ef53b192b6f
Reviewed-on: https://webrtc-review.googlesource.com/92383
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24189}
2018-08-03 18:07:51 +00:00
Emircan Uysaler
03e6ec9db0 Reland "Add multiplex case to webrtc_perf_tests"
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889

Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}

Bug: webrtc:7671
Change-Id: Ib6e37ce4bc0bae903dd72f49ffdc2ee583d75491
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/61120
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22376}
2018-03-10 01:21:04 +00:00
Taylor Brandstetter
081136fe53 Revert "Reland "Add multiplex case to webrtc_perf_tests""
This reverts commit 7c5bc1cbd66d2436f80a1ddafbdc4fbff5389c6e.

Reason for revert: Breaks downstream test that was relying on FrameGeneratorCapturer::Create

Original change's description:
> Reland "Add multiplex case to webrtc_perf_tests"
> 
> This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889
> 
> Original change's description:
> > Add multiplex case to webrtc_perf_tests
> >
> > This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> > codec. In order to have the correct input, it adds I420A case to
> > SquareGenerator and corresponding PSNR and SSIM calculations.
> >
> > Bug: webrtc:7671
> > Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> > Reviewed-on: https://webrtc-review.googlesource.com/52180
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> > Reviewed-by: Erik Språng <sprang@webrtc.org>
> > Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22330}
> 
> Bug: webrtc:7671
> Change-Id: Iba2e89aee73a73a0372edea26933d6a7ea2e0ec9
> TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
> Reviewed-on: https://webrtc-review.googlesource.com/60600
> Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22336}

TBR=phoglund@webrtc.org,sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: I26d32f9fe8d97ea341aac15cbbd43ed89a0b5b9d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/60680
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22337}
2018-03-08 01:54:22 +00:00
Emircan Uysaler
7c5bc1cbd6 Reland "Add multiplex case to webrtc_perf_tests"
This is a reland of d90a7e842437f5760a34bbfa283b3c4182963889

Original change's description:
> Add multiplex case to webrtc_perf_tests
>
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
>
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}

Bug: webrtc:7671
Change-Id: Iba2e89aee73a73a0372edea26933d6a7ea2e0ec9
TBR: niklas.enbom@webrtc.org, phoglund@webrtc.org, sprang@webrtc.org
Reviewed-on: https://webrtc-review.googlesource.com/60600
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22336}
2018-03-08 00:17:20 +00:00
Emircan Uysaler
5aac372db9 Revert "Add multiplex case to webrtc_perf_tests"
This reverts commit d90a7e842437f5760a34bbfa283b3c4182963889.

Reason for revert: 
Fails on Win ASan bots.
https://logs.chromium.org/v/?s=chromium%2Fbb%2Fclient.webrtc%2FWin32_ASan%2F4002%2F%2B%2Frecipes%2Fsteps%2Fvideo_engine_tests%2F0%2Fstdout

Original change's description:
> Add multiplex case to webrtc_perf_tests
> 
> This CL adds two new tests to perf, covering I420 and I420A input to multiplex
> codec. In order to have the correct input, it adds I420A case to
> SquareGenerator and corresponding PSNR and SSIM calculations.
> 
> Bug: webrtc:7671
> Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
> Reviewed-on: https://webrtc-review.googlesource.com/52180
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22330}

TBR=phoglund@webrtc.org,sprang@webrtc.org,niklas.enbom@webrtc.org,qiangchen@chromium.org,emircan@webrtc.org

Change-Id: If6bfdd42556517db0dd6bda01f5d3d901ff56b0c
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7671
Reviewed-on: https://webrtc-review.googlesource.com/60560
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22332}
2018-03-07 19:10:22 +00:00
Emircan Uysaler
d90a7e8424 Add multiplex case to webrtc_perf_tests
This CL adds two new tests to perf, covering I420 and I420A input to multiplex
codec. In order to have the correct input, it adds I420A case to
SquareGenerator and corresponding PSNR and SSIM calculations.

Bug: webrtc:7671
Change-Id: I9735d725bbfba457e804e29907cee55406ae5c8d
Reviewed-on: https://webrtc-review.googlesource.com/52180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Niklas Enbom <niklas.enbom@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Emircan Uysaler <emircan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22330}
2018-03-07 18:40:30 +00:00
Niels Möller
e48c61fca7 Delete unused MediaFile module.
Delete the subdirectory modules/media_file, and all references to it.

Bug: none
Change-Id: I19d86420a7d1d51cb6174c914a90484918106c5a
Reviewed-on: https://webrtc-review.googlesource.com/40540
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21790}
2018-01-29 11:18:18 +00:00
Fredrik Solenberg
a8b7c7f4c6 Move remaining traces of VoiceEngine
- Move files from voice_engine/ to audio/.
- Rename voice_engine/utility.* to remix_resample.* since there are no other
  utilities in those files.
- Move test/mock_voe_channel_proxy.h to audio/.
- Removed voe_channel_id from Audio[Receive|Send]Stream::Config.
- Remove VoiceEngine* from AudioState::Config.
- Fix a few cpplint complaints which showed when moving files.

NOPRESUBMIT=true

Bug: webrtc:4690
Change-Id: Id266c822d956625c358fa5e193e6f4837164aef8
Reviewed-on: https://webrtc-review.googlesource.com/39268
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21657}
2018-01-17 13:27:47 +00:00
Mirko Bonadei
654320666d Including libyuv headers using fully qualified paths.
Using fully qualified paths to include libyuv headers allows WebRTC to
avoid to rely on the //third_party/libyuv:libyuv_config target to
set the -I compiler flag.

Today some WebRTC targets depend on //third_party/libyuv only to
include //third_party/libyuv:libyuv_config but with fully qualified
paths this should not be needed anymore.

A follow-up CL will remove //third_party/libyuv from some targets that
don't need it because they are not including libyuv headers.

Bug: webrtc:8605
Change-Id: Icec707ca761aaf2ea8088e7f7a05ddde0de2619a
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/28220
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21209}
2017-12-11 15:51:26 +00:00
Mirko Bonadei
92ea95e34a Fixing WebRTC after moving from src/webrtc to src/
In https://webrtc-review.googlesource.com/c/src/+/1560 we moved WebRTC
from src/webrtc to src/ (in order to preserve an healthy git history).
This CL takes care of fixing header guards, #include paths, etc...

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org


Bug: chromium:611808
Change-Id: Iea91618212bee0af16aa3f05071eab8f93706578
Reviewed-on: https://webrtc-review.googlesource.com/1561
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19846}
2017-09-15 05:02:56 +00:00
Mirko Bonadei
bb547203bf Moving src/webrtc into src/.
In order to eliminate the WebRTC Subtree mirror in Chromium, 
WebRTC is moving the content of the src/webrtc directory up
to the src/ directory.

NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=tommi@webrtc.org

Bug: chromium:611808
Change-Id: Iac59c5b51b950f174119565bac87955a7994bc38
Reviewed-on: https://webrtc-review.googlesource.com/1560
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Henrik Kjellander <kjellander@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19845}
2017-09-15 04:25:06 +00:00