1974 Commits

Author SHA1 Message Date
Danil Chapovalov
fc961357a7 Delete TestAudioDeviceModule factory which uses GlobalTaskQueueFactory
Bug: webrtc:10284
Change-Id: Ic92f6ff31b40c48a3362745a0a81179af0595fe0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141409
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28227}
2019-06-11 12:15:44 +00:00
Philip Eliasson
49d661a7d3 Revert "Inform VideoEncoder of negotiated capabilities"
This reverts commit 11dfff0878c949f2e19d95a0ddc209cdad94b3b4.

Reason for revert: Downstream import failure.

Original change's description:
> Inform VideoEncoder of negotiated capabilities
> 
> After this CL lands, an announcement will be made to
> discuss-webrtc about the deprecation of one version
> of InitEncode().
> 
> Bug: webrtc:10720
> Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Elad Alon <eladalon@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28224}

TBR=sakal@webrtc.org,kwiberg@webrtc.org,eladalon@webrtc.org,kthelgason@webrtc.org,sprang@webrtc.org

Change-Id: I7f833055c67f1f879b01dd8c156ba7b8840e8747
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10720
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141411
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28225}
2019-06-11 11:56:04 +00:00
Elad Alon
11dfff0878 Inform VideoEncoder of negotiated capabilities
After this CL lands, an announcement will be made to
discuss-webrtc about the deprecation of one version
of InitEncode().

Bug: webrtc:10720
Change-Id: Ib992af0272bbb16ae16ef7e69491f365702d179e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140884
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28224}
2019-06-11 11:32:13 +00:00
Sebastian Jansson
a7d70ab0fe Adds network update state cache to scenario tests.
Bug: webrtc:9510
Change-Id: I1ffbc21770f0442086491a07ac7970eaf594a05f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/141400
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28223}
2019-06-11 11:16:04 +00:00
Danil Chapovalov
1a5fc9035b in test/pc/e2e pass TaskQueueFactory explicitly
instead of relying on factories that use GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: Icc32ae1c159c39a6594d2aaec79c68dcc826fea8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139894
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28220}
2019-06-11 08:48:56 +00:00
Chen Xing
5d24b16c77 Prepare for splitting the api/video:video_frames build rule.
This change is part of a change to break the dependency between "api:rtp_headers" and "api/video:video_frame". It does so by first creating an empty "api/video:video_rtp_headers" build rule so that downstream projects can be fixed before moving the source files.

Bug: webrtc:10668
Change-Id: I81aa6edfef3639b457a40aa93de048e62cbfd8ef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140291
Commit-Queue: Chen Xing <chxg@google.com>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28209}
2019-06-10 11:50:51 +00:00
Sebastian Jansson
24cf2606e4 Adds TCP fairness test to receive side congestion controller.
Bug: webrtc:9883
Change-Id: I3697491285e4f70b8f7857198e4e1ccb0097da5b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140883
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28196}
2019-06-07 13:49:37 +00:00
Sebastian Jansson
b13ccc5288 Adds TCP fairness test to GoogCC.
Bug: webrtc:9883
Change-Id: Ie78e51edb08f6c22dbf02168b1d3b067b2c0c55e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140293
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28193}
2019-06-07 11:52:03 +00:00
Mirko Bonadei
71061bcca8 Replace calls to deprecated googletest APIs.
SetUpTestCase/TearDownTestCase -> SetUpTestSuite/TearDownTestSuite.

TBR=kwiberg@webrtc.org

Bug: None
Change-Id: I6d873c62d6b5c9d7100624d00e1c4894d686a9f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/140041
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28189}
2019-06-07 06:41:20 +00:00
Ilya Nikolaevskiy
ab62b2ee51 Don't copy video frame metadata in each encoder/decoder
As this is handled higher up the pipeline in a single
place for all encoders/decoders

Bug: webrtc:10460
Change-Id: I95b0a69aecaf07283c8776ac0d7e85d097e3576b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139882
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28172}
2019-06-05 15:33:49 +00:00
Danil Chapovalov
f4c7ab1bb2 in test/scenario pass TaskQueueFactory explicitly
instead of relying on factories that use GlobalTaskQueueFactory

Bug: webrtc:10284
Change-Id: Iafece5e83ccfd33499e9a473ea7e2e99d5c824c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139522
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28150}
2019-06-04 09:21:17 +00:00
Elad Alon
e86af2c75f Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
Loss notifications may either be sent immediately, or wait until another
RTCP feedback message is sent.

Bug: webrtc:10336
Change-Id: I40601d9fa1dec6c17b2ce905cb0c8cd2dcff7893
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139242
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28142}
2019-06-03 16:28:34 +00:00
Artem Titov
85a9d91cd4 Add ability to set min/start/max bitrate on peer's PC in PC quality tests
Bug: webrtc:10138, webrtc:10692
Change-Id: I4d7ae84dc2945fef6451a6671786b3b19cd9abd8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/139108
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28107}
2019-05-29 13:25:26 +00:00
Artem Titov
232b6a16cc Propagate screenshare info into video track and it's source.
If screen share is set, then we need to tell video source, that it
is screen share source. Also video track should be aware, that it is
screen share track. It is required to choose proper video encoding
settings.

Bug: webrtc:10138
Change-Id: I5c82584ae0325a303a495554d87962a98b676694
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138278
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@{#28098}
2019-05-29 10:14:22 +00:00
Artem Titov
2370242acf Enable flex fec support in PC quality test framework
Bug: webrtc:10138, webrtc:10683
Change-Id: I9235fef99d3ea857f10234fdd82e8468480f71a1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138822
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28078}
2019-05-27 14:48:48 +00:00
Anton Sukhanov
8b096a03b4 LogToSderr by default in WebRTC tests
Printing logs to stderr helps debugging and investigating CQ failures.

Bug: webrtc:5996
Change-Id: I365ee0a0b3ff3d999f1a9a293d3c05bd75e5b999
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138187
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28065}
2019-05-24 19:39:18 +00:00
Elad Alon
fadb1811a8 Negotiate use of RTCP loss notification feedback (LNTF)
When the LossNotifications field trial is in effect, LNTF should
be offered/accepted in the SDP message, not assumed to be configured
on both sides equally.

Bug: webrtc:10662
Change-Id: Ibd827779bd301821cbb4196857f6baebfc9e7dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/138079
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28056}
2019-05-24 12:44:14 +00:00
Sebastian Jansson
58c71db1b3 Fix for crash in event log when using scenario tests.
Scenario tests runs all its activities on task queues. This is not
allowed by the default event log writer, causing a DCHECK failure.
This CL makes it possible to stop the event asynchronously,
thereby avoiding the need for the DCHECK.

Bug: webrtc:10365
Change-Id: I1206982b29fd609ac85b4ce30ae9291cbec52041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136685
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28027}
2019-05-22 15:22:49 +00:00
Danil Chapovalov
b32f2c7f57 Publish rtc event log api and default factory for it in api/
Bug: webrtc:10206
Change-Id: I34194ddb6fd2b0a3d7c553fadc9ddc1ea9740da0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137500
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28023}
2019-05-22 13:38:25 +00:00
Anton Sukhanov
4f08faae82 Introduce MediaTransportConfig
Currently we pass media_transport from PeerConnection to media layers. The goal of this change is to replace media_transport with struct MediaTransportCondif, which will enable adding different transports (i.e. we plan to add DatagramTransport) as well as other media-transport related settings without changing 100s of files.

TODO: In the future we should consider also adding rtp_transport in the same config, but it will require a bit more work, so I did not include it in the same change.


Bug: webrtc:9719
Change-Id: Ie31e1faa3ed9e6beefe30a3da208130509ce00cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137181
Commit-Queue: Anton Sukhanov <sukhanov@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28016}
2019-05-21 18:58:33 +00:00
Niels Möller
04a3cc1ad9 Delete rtc_base/unittest_main.cc
Usage replaced with test/test_main.cc.

Bug: webrtc:5996
Change-Id: I65e7539f2072fb45255a3c1af0b10dd06e1701ca
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137805
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28010}
2019-05-21 14:44:11 +00:00
Artem Titov
4d29ef063c Add periodic alive message logging to prevent test infra think, that test is dead
Bug: webrtc:10138
Change-Id: Ib39ff6df81776a7784687be2dc16ab81c500cc3e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137428
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28005}
2019-05-21 11:09:18 +00:00
Sebastian Jansson
8abcf83b4f Adds IsEmpty to SampleStats.
Bug: webrtc:9883
Change-Id: Ie8ef801cb60fd74c0354ff9fbbdbc33b7d105317
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137514
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28003}
2019-05-21 09:41:41 +00:00
Niels Moller
9d1840c3df Revert "Delete NO_MAIN_THREAD_WRAPPING preprocessor define."
This reverts commit 0f78c6b28dbc0c9caa555ce89ce91b0f08c510ea.

Reason for revert: Breaks downstream tests.

Original change's description:
> Delete NO_MAIN_THREAD_WRAPPING preprocessor define.
> 
> Since many tests rely on rtc::Thread::Current(), add an
> explicit rtc::AutoThread in the main() function used by tests.
> 
> Bug: webrtc:9714
> Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28000}

TBR=kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org

Change-Id: Iff939bb0d5ad0ea01b953321993733bb56c9070b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9714
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137512
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28001}
2019-05-21 07:26:54 +00:00
Niels Möller
0f78c6b28d Delete NO_MAIN_THREAD_WRAPPING preprocessor define.
Since many tests rely on rtc::Thread::Current(), add an
explicit rtc::AutoThread in the main() function used by tests.

Bug: webrtc:9714
Change-Id: Id82121967c9621fe1c2945846009c48139fa57da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/39680
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28000}
2019-05-21 06:53:54 +00:00
Sebastian Jansson
871ac42597 Refactor of GoogCC debug printer.
Simplifying the code to better fit with how it is used.

Bug: webrtc:9883
Change-Id: I2bd52f26b829413e516dee4f551cf36574275019
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136681
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27994}
2019-05-20 18:40:26 +00:00
Mirko Bonadei
39f46810ff Remove unused dependency.
Bug: None
Change-Id: I13ef76d9f8410bda3591c5fc8a9607c768c92b65
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137432
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27987}
2019-05-20 12:57:44 +00:00
Sebastian Jansson
3525f86c42 Adds feedback generator.
This is a useful tool to use for unittests of code that uses
TransportFeedback as input.

Bug: webrtc:10498
Change-Id: I171b22841eb9e16a5d5b785ff45ae9df5a6ccd7f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137423
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27975}
2019-05-17 16:14:32 +00:00
Artem Titov
519d74a5fc Drop data for disabled endpoints.
Drop packets received from disabled endpoint and return socket error
when trying to send data from disabled endpoint.

Bug: webrtc:10138
Change-Id: I55259d2ac47adea78b47aeb25842e63a98a405c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134643
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27973}
2019-05-17 10:31:22 +00:00
Mirta Dvornicic
a24d934ee4 Add the option to use raw RTP packetization without the generic header.
Bug: webrtc:10625
Change-Id: I198031154dbb706ae1e7c15bd34a3bdf93d1a51a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136923
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27964}
2019-05-16 14:41:42 +00:00
Niels Möller
198cf00532 Reland "Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN"
This is a reland of e779847fb6499ac2dc4757de8c625ac377e9d0d4

Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
>
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
>
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}

Tbr: kwiberg@webrtc.org
Bug: webrtc:6424
Change-Id: Ic08d5d7fbc25ff89e4182d7c9cb3b0e8e356339a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135946
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27957}
2019-05-16 08:21:04 +00:00
Artem Titov
7581ff7375 Add screen share support to PC level test framework
Bug: webrtc:10138
Change-Id: I1a8ac683e91f8061387f407610d7db2a6d0d4fe9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136805
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27950}
2019-05-15 14:07:00 +00:00
Artem Titov
a768345596 Reduce flakiness of NetworkEmulationManagerTest.ThroughputStats
Bug: webrtc:10138, webrtc:10553
Change-Id: Id7027b94b3a6c072ee48c1291b638f2dc8b8b095
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133164
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27949}
2019-05-15 11:32:46 +00:00
Ilya Nikolaevskiy
2ebf523978 Reland "Copy video frames metadata between encoded and plain frames in one place"
Reland with fixes.

Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346

Bug: webrtc:10460
Change-Id: I98629589fa55ca1d74056033cf86faccfdf848cd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136582
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27930}
2019-05-13 14:51:11 +00:00
Sebastian Jansson
dab21c6a91 Fix for crash in frame matcher on short runs.
The capture time stamp was not set when finalizing a simulation where
no frames were delivered, this triggered a DCHECK.

Also adding a unit test that would have caught this.

Bug: webrtc:10365
Change-Id: I839d1c01dbf260723ed30d3e846efff280d7744f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136585
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27929}
2019-05-13 13:43:06 +00:00
Artem Titov
0379d8cfea Change a way, how receive stream is determined in DefaultAudioQualityAnalyzer.
Bug: webrtc:10138
Change-Id: I8955c30f0a5d98abeca029323396e469a2fb243b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/136683
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27927}
2019-05-13 13:07:01 +00:00
Niels Moller
fb8c856afa Revert "Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN"
This reverts commit e779847fb6499ac2dc4757de8c625ac377e9d0d4.

Reason for revert: Breaks downstream projects, depending on indirect include.

Original change's description:
> Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
> 
> Also add explicit includes of rtc_base/string_utils.h in files depending on it.
> 
> Bug: webrtc:6424
> Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27903}

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

Change-Id: Ib04280d401b66fe832d3fdc9293e39276710f973
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6424
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135945
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27909}
2019-05-10 10:23:01 +00:00
Niels Möller
e779847fb6 Change SimpleStringBuilder::Append to not use strcpyn and SIZE_UNKNOWN
Also add explicit includes of rtc_base/string_utils.h in files depending on it.

Bug: webrtc:6424
Change-Id: Id6b53937ab2d185d092a5d8863018fd5f1a88e27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135744
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27903}
2019-05-10 08:38:42 +00:00
Ilya Nikolaevskiy
de20b9683c Revert "Reland "Copy video frames metadata between encoded and plain frames in one place""
This reverts commit 4fb12b0caec9faa57cfbceb0f86b0e10c32a0cc2.

Reason for revert: Breaks some asan chromium bots

Original change's description:
> Reland "Copy video frames metadata between encoded and plain frames in one place"
> 
> Reland with fixes.
> 
> Currently some video frames metadata like rotation or ntp timestamps are
> copied in every encoder and decoder separately. This CL makes copying to
> happen at a single place for send or receive side. This will make it
> easier to add new metadata in the future.
> 
> Also, added some missing tests.
> 
> Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346
> 
> Bug: webrtc:10460
> Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#27828}

TBR=ilnik@webrtc.org,nisse@webrtc.org,sprang@webrtc.org,kron@webrtc.org,artit@webrtc.org

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

Bug: webrtc:10460
Change-Id: I9c87a43a716622b389974cb8377f973573fc29a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135747
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27895}
2019-05-09 17:47:51 +00:00
Mirko Bonadei
daac58290e Remove -Wno-undef and -Wno-extra-semi.
These issues have been fixed upstream in Abseil.

Bug: webrtc:10138
Change-Id: Ic0ebd22d0ad95bbd5269c08c182a76f9bf42f3a2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135571
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27883}
2019-05-08 17:42:09 +00:00
Niels Möller
d8b9ed77cf Promote RtcEventLogOutputFile to api/
Preparation for deleting PeerConnectionInterface::StartRtcEventLog
method with a PlatformFile argument.

Bug: webrtc:6463
Change-Id: Ia9fa1d99a3d87f3bf193e73382690b782ffea65c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135285
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27879}
2019-05-08 12:29:42 +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
630bd43fcf DefaultAudioQualityAnalyzer: use bytes_recv instead of packets_recv.
Bug: webrtc:10138
Change-Id: I2fa5d9da2dd7788ffc48cf6a4171eb3ce0de5423
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135461
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27866}
2019-05-07 14:59:02 +00:00
Artem Titov
f65a89b7f7 Add support of specifying concrete codec for video stream
Bug: webrtc:10138
Change-Id: I074bfccfa5c8f619ea7fa17d6ca99f9b4cbb18b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/123386
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27864}
2019-05-07 11:46:57 +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
Per Åhgren
9363c778fe Remove deprecated call to UpdateHistogramsOnCallEnd
Bug: webrtc:5298
Change-Id: I440e5972ecb69e2d90d918cc5106a16ade4a6041
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/135126
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27849}
2019-05-03 14:24:32 +00:00
Sebastian Jansson
87a92d087c Don't require call to ValidateFieldTrialsStringOrDie for ScopedFieldTrials.
Bug: webrtc:9883
Change-Id: Iae7b2d22666ad57176237241a7f895cbd47cd26d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134311
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27846}
2019-05-03 13:44:38 +00:00
Sebastian Jansson
1391ed242a Allows injection of network controller factory in test fixture.
Bug: webrtc:9155
Change-Id: I929c4cde66ad6743b4a8df2df3abfa7593992977
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134645
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27843}
2019-05-03 13:22:45 +00:00
Ilya Nikolaevskiy
4fb12b0cae Reland "Copy video frames metadata between encoded and plain frames in one place"
Reland with fixes.

Currently some video frames metadata like rotation or ntp timestamps are
copied in every encoder and decoder separately. This CL makes copying to
happen at a single place for send or receive side. This will make it
easier to add new metadata in the future.

Also, added some missing tests.

Original Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133346

Bug: webrtc:10460
Change-Id: Ia71198685de7fbd990704b575231cdce94dc0645
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134961
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27828}
2019-05-02 13:29:14 +00:00
Niels Möller
d9c2d94620 Move ownership of VCMJitterEstimator to FrameBuffer
Bug: webrtc:7408
Change-Id: I8b33ead80abff1e84ae0b223e108266f71f03e2f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134180
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#27823}
2019-05-02 10:57:04 +00:00