admin/webrtc_m130
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity
42,909 Commits 1 Branch 0 Tags
Commit Graph

701 Commits

Author SHA1 Message Date
Andrey Logvin
d154e36d96 Remove unused QualityMetricsReporter::Start method from pc test api 
Bug: webrtc:11381
Change-Id: I254aef4f0c218fe1ee70499f405449f8a226b44b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178567
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31618}
 2020-07-03 07:11:36 +00:00
Artem Titov
1ff3c584cd Add TimeController to the CreatePeerConnectionE2EQualityTestFixture API 
Add TimeController to the CreatePeerConnectionE2EQualityTestFixture
method as a first step to make PC level framework compatible with
TimeController abstraction.

Bug: webrtc:11743
Change-Id: I69305abc880059bf9fe1d4f2e3b7c10cf35417db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178485
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31607}
 2020-07-01 15:18:34 +00:00
Andrey Logvin
20f45823e3 Add sync group mapping to TrackIdStreamLabelMap 
Bug: webrtc:11381
Change-Id: I0f4c590d5474d1aa84c8a6e7a8b3fab252b0b3fc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178362
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31601}
 2020-07-01 11:17:21 +00:00
Andrey Logvin
9d841fb1f5 Add Start method with TrackIdStreamLabelMap to PeerConnectionE2EQualityTestFixture::QualityMetricsReporter 
Bug: webrtc:11381
Change-Id: I55b671e9a2928da3d204030654d4eee2a5893448
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178360
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31598}
 2020-07-01 07:43:12 +00:00
Per Kjellander
4c50e70569 Ensure CreateTimeControllerBasedCallFactory use simulated time in Call::SharedModuleThread 
Also removes unnecessary Call::SharedModulesThread ctor.

Bug: webrtc:11598
Change-Id: I6d6a7ca4359598fbbfd4ae1aa550be6227ea27e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178394
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31594}
 2020-06-30 15:38:35 +00:00
Andrey Logvin
739cfb2f58 Add sync group validation in pc level test framework 
Bug: webrtc:11381
Change-Id: I4ef62675c0cb688abccc130fb91a69c3c78bf837
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178383
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31587}
 2020-06-30 09:53:19 +00:00
Niels Möller
29d59a1402 Add method PeerConfigurer::SetBitrateSettings 
It replaces the method SetBitrateParameters, which uses the
deprecated type PeerConnectionInterface::BitrateParameters.

Bug: None
No-try: True
Change-Id: I3690d391d679c3ff5b79e088f6c7f79bc3571064
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177667
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31557}
 2020-06-24 12:07:06 +00:00
Niels Möller
2a70703eb8 Delete MediaTransportInterface and DatagramTransportInterface 
Bug: webrtc:9719
Change-Id: Ic9936a78ab42f4a9bb4cc3265f0a2cf36946558f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176500
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31536}
 2020-06-17 08:41:14 +00:00
Andrey Logvin
9b526180c9 Migrate pc level test metrics to new getStart API 
Bug: webrtc:10430
Change-Id: I7555cb967f2e341da43338cb0f8652490992bd31
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176857
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31526}
 2020-06-15 18:28:52 +00:00
Mirko Bonadei
2dcf348011 Use absl_deps in order to preapre to the Abseil component build release. 
Bug: webrtc:1046390
Change-Id: Ia35545599de23b1a2c2d8be2d53469af7ac16f1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176502
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31463}
 2020-06-08 12:59:40 +00:00
Steve Anton
f84ab8e2dd Add publicly visible mock for PeerConnectionFactoryInterface 
Bug: webrtc:11642
Change-Id: I71c99034a70780d4d1ffc8330836c76c6105bc96
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176421
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31430}
 2020-06-03 17:46:11 +00:00
Artem Titov
3b641675de Add list of participants to the start method of video analyzer. 
To support multiple participants video quality analyzer may need to know
peer names in advance to simplify internal structures and metrics
reporting.

Bug: webrtc:11631
Change-Id: I4ffb1554ab7f0e015b8e937b7ffddd55aba9826f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176364
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31415}
 2020-06-03 08:08:47 +00:00
Artem Titov
8a0284e2a8 Add peer name to video quality analyzer interface. 
Add peer name to video quality analyzer interface to make it possible to
add multipeer support.

Change-Id: I2570cd4481503c8634bdd91208b3dd2fa1d62029
Bug: webrtc:11631
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176329
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31395}
 2020-06-01 11:48:50 +00:00
Artem Titov
65dd291c8d Fix documentation for some public API methods 
Bug: None
Change-Id: I3f6d6d7934fbe08a1b475da3d3d430b47aa27dd0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176141
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31366}
 2020-05-27 13:56:33 +00:00
Tommi
25c77c1aea Add SharedModuleThread class to share a module thread across Call instances. 
This reduces the number of threads allocated per PeerConnection when
more than one PC is needed.

Bug: webrtc:11598
Change-Id: I3c1fd71705f90c4b4bbb1bc3f0f659c94016e69a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175904
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31347}
 2020-05-25 17:21:56 +00:00
Per Åhgren
e9cd6177eb Add ability for audioproc_f to operate on any AudioProcessing object. 
This CL extends the WebRTC testing API to allow audioproc_f -based
testing using a pre-created AudioProcessing object. This is an
important feature to allow testing any AudioProcessing objects
that are injected into WebRTC.

Beyond adding this, the CL also changes the simulation code to
operate on a scoped_refptr<AudioProcessing> object instead of a
std::unique<AudioProcessing> object

Bug: webrtc:5298
Change-Id: I70179f19518fc583ad0101bd59c038478a3cc23d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175568
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31319}
 2020-05-19 11:37:18 +00:00
Marina Ciocea
bb13f38a9e Add MockTransformableVideoFrame to api/test/. 
Define MockTransformableVideoFrame in api/test to be use in Blink
tests.

Bug: chromium:1069295
Change-Id: I960d8228eddbdf8263b731814a00fd02d0a6c54e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175136
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Marina Ciocea <marinaciocea@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31272}
 2020-05-15 09:50:19 +00:00
Tommi
9b7232a68c Set up a new rtc::Thread instance per test. 
Several tests leave pending tasks behind after executing, which may
affect the state of subsequent tests. This CL isolates each test in
the sense that a dedicated Thread instance is created per test and
then pending tasks are flushed and the Thread instance deleted.

Down the line we may want to improve on this and flag those tests
that leave pending tasks/timers etc.

Change-Id: Ibaf3719a9974c57ac2169edca0e2a06a9ea6c78f
Bug: webrtc:11574
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175132
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31268}
 2020-05-15 09:13:02 +00:00
Danil Chapovalov
42748d8e22 In rtc_base/ and api/ replace mock macros with unified MOCK_METHOD macro 
Bug: webrtc:11564
Change-Id: I8f4171490df39c683276f9d401a4083962ee409a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/175130
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31266}
 2020-05-14 19:33:26 +00:00
Andrey Logvin
f3319816ad Separate capturing device index from VideoConfig 
The last step of the pc framework tests migration.

Bug: webrtc:11534
Change-Id: I344c443b6d21422ef418315b7e5a6cb26ae3473d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174741
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31232}
 2020-05-13 09:16:40 +00:00
Andrey Logvin
b856dc1556 Remove VideoGeneratorType from pc framework test api. 
VideoGeneratorType wasn't deleted in https://webrtc-review.googlesource.com/c/src/+/174541

Bug: webrtc:11534
Change-Id: I3e631240dc0b28a53e62b65e3dd094b5773fac2c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174721
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31228}
 2020-05-12 22:24:36 +00:00
Artem Titov
f9ed56b656 Add ability to set custom RtpEncodingParameters for each simulcast stream in PC framework 
Bug: webrtc:11557
Change-Id: I9f44728ff9178cd9c7dbe4cbcd639d610a981015
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174754
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31218}
 2020-05-11 20:46:30 +00:00
Artem Titov
cc57b935cd Make video quality analyzer compatible with real SFU in the network 
Bug: webrtc:11557
Change-Id: I8ab1fb0896e267f30856a45df6099bd9aae9bc03
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174801
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@{#31216}
 2020-05-11 18:54:33 +00:00
Artem Titov
baa2c836ba Introduce ability to set peer name for PC level tests 
Add peer's name to params and use it for logging and metrics naming
for whole peer related metrics.

Bug: webrtc:11479
Change-Id: Ia7e3fc4839c90a958d66910614515ac02a96e389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174752
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31215}
 2020-05-11 18:47:03 +00:00
Danil Chapovalov
fc11519c92 Cleanup mocks in api/test 
Modernise functions to unified MOCK_METHOD macro,
delete few deprecated functions on the way.
add one missing function (in MockEncodedImageCallback)
Remove proxy mock function (in MockVideoBitrateAllocatorFactory)

Remove default constructors and destructors

Bug: None
Change-Id: Ibebb0d9e3c9be5877649af7bde8b87222ddf04fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174751
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31195}
 2020-05-08 20:01:03 +00:00
Andrey Logvin
435fb9ad06 Remove screen_share_config from the VideoConfig. 
After the migration of the pc framework tests (https://webrtc-review.googlesource.com/c/src/+/174023), having "absl::optional<ScreenShareConfig> screen_share_config" field in VideoConfig became redundant. Replaced it with VideoTrackInterface::ContentHint content_hint field.

Bug: webrtc:11534
Change-Id: Ibf4b1c8daed95ef02111fe952171f11e290905d3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174702
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31187}
 2020-05-08 08:56:13 +00:00
Danil Chapovalov
b63331bb8f Cleanup mocks for Video (en|de)coder factories 
In particular remove proxy mocks in favor of lambdas and Return(ByMove(...))

Bug: None
Change-Id: If6b79601437e82a7116479d128d538e965622fab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174701
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31179}
 2020-05-07 11:58:50 +00:00
Andrey Logvin
1e83d34fc1 Remove pc level test framework redundant code. 
After the migration to passing frame video source implementation directly, part of the peer connection framework code became redundant. Removing screen_share_config and capturing_device_index from the VideoConfig is to be done in later reviews.

Bug: webrtc:11534
Change-Id: I7a8ea85d26d00fb5bfe7ec0d2facef9c44a0f749
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174541
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31178}
 2020-05-07 09:23:29 +00:00
Andrey Logvin
42c59525b1 Create default frame generator in the AddVideoConfig method. 
Bug: webrtc:11534
Change-Id: I5f8e6009ac48be99180574ab3ac835005f67cf58
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174540
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31176}
 2020-05-06 21:01:29 +00:00
Andrey Logvin
c064467b32 Pass frame generator to the AddVideoConfig method in the pc framework tests. 
Bug: webrtc:11534
Change-Id: Id68feca50611f412897ddef3d43b811a224b200f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174023
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31167}
 2020-05-05 17:20:25 +00:00
Andrey Logvin
dad6a940e1 Add helper frame generator factories for the pc framework tests. 
Bug: webrtc:11534
Change-Id: I569fb9e78aa38f0a17f4e4a261dd93c4b8ba9ca0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174340
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31162}
 2020-05-04 18:56:22 +00:00
Artem Titov
b5a013815f Rename done() into condition(), because it is actually condition in TimeController API 
Bug: None
Change-Id: Ia3a742d1d2ad1238223f4da7ae843a8d22108ec5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/174060
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31144}
 2020-04-29 10:29:09 +00:00
Johannes Kron
3e98368ec5 Reland "Distinguish between send and receive codecs" 
This reverts commit 8e8b36a94a7a7a1fd0f8093979a406afa56e18c1.

Reason for revert: The CL has been improved with the following changes,
  - Fixed negotiation of send/receive only clients.
  - Handles the implicit assumption that any H264 decoder also can
    decode H264 constraint baseline.

Original change's description:
> Distinguish between send and receive codecs
>
> Even though send and receive codecs may be the same, they might have
> different support in HW. Distinguish between send and receive codecs
> to be able to keep track of which codecs have HW support.
>
> Bug: chromium:1029737
> Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30284}

Change-Id: I834ed48ee78d04922c73e2836165e476925e1cc5
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168605
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30932}
 2020-03-29 21:03:27 +00:00
Artem Titov
d19513f3ff Move calculation of target_encode_bitrate to DefaultVideoQualityAnalyzer 
To migrate on new GetStats API and properly support target encode bitrate
for regular, simulcast and svc cases we need to calculate it inside video
quality analyzer getting values from SetRates in VideoEncoder.

Bug: webrtc:11381
Change-Id: Ia37acac764ed3c30f64cdbfda8906d543fa03ae2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171501
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30881}
 2020-03-25 11:38:47 +00:00
Ivo Creusen
26d52e1ba0 Add optional output audio file to NetEq simulation API 
Bug: webrtc:10337
Change-Id: I2e9071d4d2bd4b181d198031cf459965c9682775
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171518
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30873}
 2020-03-24 16:31:08 +00:00
Sebastian Jansson
89eb0bba0c Adds UpdateConfig to SimulatedNetwork 
Bug: webrtc:9510
Change-Id: Ied0e5ff291021ba4f539eee9820b8490a7004882
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/170462
Reviewed-by: Ali Tofigh <alito@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30803}
 2020-03-16 15:58:43 +00:00
Sebastian Jansson
db5d7e470f Cleanup: Use common IP overhead definitions in test and prod code 
This avoid duplication. As part of this moving the overhead calculation
to the IP address class so it's easier to find and more natural to use.

Bug: webrtc:9883
Change-Id: If4d865f445bc1a302572896932966ce30294e339
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169445
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30657}
 2020-03-02 11:36:58 +00:00
Harald Alvestrand
61f74d91f8 Reland "Expose can_trickle_ice_candidates on PeerConnection" 
This reverts commit cb8c40138ca170f841bc45fa6771cdfc4b966e5f.

Reason for revert: Added missing default.

Original change's description:
> Revert "Expose can_trickle_ice_candidates on PeerConnection"
>
> This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.
>
> Reason for revert: Breaks downstream due to missing default
>
> Original change's description:
> > Expose can_trickle_ice_candidates on PeerConnection
> >
> > Bug: chromium:708484
> > Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> > Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> > Reviewed-by: Taylor <deadbeef@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30653}
>
> TBR=deadbeef@webrtc.org,hta@webrtc.org
>
> Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:708484
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30655}

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

Change-Id: I608da7781f158b4b02dd226d4dcd5615c4935fa8
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169541
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30656}
 2020-03-02 11:35:53 +00:00
Harald Alvestrand
cb8c40138c Revert "Expose can_trickle_ice_candidates on PeerConnection" 
This reverts commit c6a65c8866487c6adc0a7bb472d3bad9389501f9.

Reason for revert: Breaks downstream due to missing default

Original change's description:
> Expose can_trickle_ice_candidates on PeerConnection
> 
> Bug: chromium:708484
> Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
> Commit-Queue: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30653}

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

Change-Id: Iaa5b977c4237715a8a5127cf167cf6512a3f7059
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:708484
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169540
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30655}
 2020-03-02 10:14:14 +00:00
Harald Alvestrand
c6a65c8866 Expose can_trickle_ice_candidates on PeerConnection 
Bug: chromium:708484
Change-Id: I9a40e75066341f0d9f965bd3718bfcb3f0459533
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/169450
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30653}
 2020-03-02 05:19:16 +00:00
Artem Titov
4a6f81829b Add ability to enable AV sync in PC level tests 
Bug: webrtc:11381
Change-Id: I223ff0a2b81632ee7cbbac5b722bb6a7d5f72f7e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168959
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30629}
 2020-02-27 14:22:23 +00:00
Artem Titov
80a82f1527 PC test framework: cleanup deprecated API 
Bug: webrtc:10138
Change-Id: I116bb318d3b736f1ec60651eaab53c6e78fb9d07
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168521
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30529}
 2020-02-17 10:25:10 +00:00
Danil Chapovalov
0c626afcf3 Use newer version of TimeDelta and TimeStamp factories in webrtc 
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g"
find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g"
git cl format

Bug: None
Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30491}
 2020-02-10 12:21:17 +00:00
Johannes Kron
8e8b36a94a Revert "Reland "Reland "Reland "Distinguish between send and receive codecs"""" 
This reverts commit 184ea66aed43161f05d80fbb74183a2efccca352.

Reason for revert: Breaks downstream projects.

TBR=steveanton@webrtc.org

Original change's description:
> Reland "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.
>
> Reason for revert: Keep logic as is.
>
> Original change's description:
> > Revert "Reland "Reland "Distinguish between send and receive codecs"""
> >
> > This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
> >
> > Reason for revert: Breaks perf test on iOS.
> >
> > Original change's description:
> > > Reland "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> > >
> > > Reason for revert: Flaky test in Chromium fixed.
> > >
> > > Original change's description:
> > > > Revert "Reland "Distinguish between send and receive codecs""
> > > >
> > > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > > >
> > > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > > >
> > > > Original change's description:
> > > > > Reland "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > > >
> > > > > Reason for revert: Fixed negotiation of send-only clients.
> > > > >
> > > > > Original change's description:
> > > > > > Revert "Distinguish between send and receive codecs"
> > > > > >
> > > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > > >
> > > > > > Reason for revert: breaks negotiation with send-only clients
> > > > > >
> > > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > > >
> > > > > > Original change's description:
> > > > > > > Distinguish between send and receive codecs
> > > > > > >
> > > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > > to be able to keep track of which codecs have HW support.
> > > > > > >
> > > > > > > Bug: chromium:1029737
> > > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > > >
> > > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > > >
> > > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > > No-Presubmit: true
> > > > > > No-Tree-Checks: true
> > > > > > No-Try: true
> > > > > > Bug: chromium:1029737
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > > >
> > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > >
> > > > >
> > > > > Bug: chromium:1029737
> > > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30360}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30367}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30373}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
>
> Bug: chromium:1029737
> Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30415}

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

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

Bug: chromium:1029737
Change-Id: Ice25339e7dfb9fc75049bd207d097b0910bd4446
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168341
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30484}
 2020-02-07 15:11:08 +00:00
Sebastian Jansson
09a9f1ba72 Adds simulated time controller API. 
Bug: webrtc:11255
Change-Id: I68289a45b9441b5e612433acd96dc3cb24e47ce4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168122
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30443}
 2020-02-03 10:19:08 +00:00
Erik Språng
261f792f83 Allow software fallback on lowest simulcast stream for temporal support 
Bug: webrtc:11324
Change-Id: Ie505be0cda74c0444065d86c3727671c62bd4842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167527
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30437}
 2020-01-31 16:44:47 +00:00
Ivo Creusen
d69935c114 Remove function that takes command-line arguments directly 
This function is obsolete now that config-based functions are available.
The command-line parsing should not happen here but in the executable
that uses these functions.

Bug: webrtc:11005
Change-Id: I618d12503123e3e1fd6e572a045372c622043a75
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167860
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30421}
 2020-01-30 12:42:38 +00:00
Johannes Kron
184ea66aed Reland "Reland "Reland "Distinguish between send and receive codecs""" 
This reverts commit a104ceb0ceec0f95e199e6d6704f41ec88a51fc5.

Reason for revert: Keep logic as is.

Original change's description:
> Revert "Reland "Reland "Distinguish between send and receive codecs"""
>
> This reverts commit 9bac68c0cc4444b852416396f0e0f31ea66a9cfe.
>
> Reason for revert: Breaks perf test on iOS.
>
> Original change's description:
> > Reland "Reland "Distinguish between send and receive codecs""
> >
> > This reverts commit 00a30873c415d717af8dcdf21c2df7fd4b6d1ed2.
> >
> > Reason for revert: Flaky test in Chromium fixed.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive codecs""
> > >
> > > This reverts commit 133bf2bd28596aab5c7684e0ea3da99b1fece77f.
> > >
> > > Reason for revert: Breaks Chromium import due to flaky test in Chromium.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive codecs"
> > > >
> > > > This reverts commit e57b266a20334e47f105a0bd777190ec8c6562e8.
> > > >
> > > > Reason for revert: Fixed negotiation of send-only clients.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive codecs"
> > > > >
> > > > > This reverts commit c0f25cf762a6946666c812f7a3df3f0a7f98b38d.
> > > > >
> > > > > Reason for revert: breaks negotiation with send-only clients
> > > > >
> > > > > (webrtc_video_engine.cc:985): SetRecvParameters called with unsupported video codec: VideoCodec[96:H264]
> > > > > (peer_connection.cc:6043): Failed to set local video description recv parameters. (INVALID_PARAMETER)
> > > > > (peer_connection.cc:2591): Failed to set local offer sdp: Failed to set local video description recv parameters.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive codecs
> > > > > >
> > > > > > Even though send and receive codecs may be the same, they might have
> > > > > > different support in HW. Distinguish between send and receive codecs
> > > > > > to be able to keep track of which codecs have HW support.
> > > > > >
> > > > > > Bug: chromium:1029737
> > > > > > Change-Id: Id119560becadfe0aaf861c892a6485f1c2eb378d
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/165763
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30284}
> > > > >
> > > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: Iacb7059436b2313b52577b65f164ee363c4816aa
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166420
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Steve Anton <steveanton@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30292}
> > > >
> > > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > > >
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: I287efcfdcd1c9a3f2c410aeec8fe26a84204d1fd
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/166604
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30348}
> > >
> > > TBR=steveanton@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: I9f8731309749e07ce7e651e1550ecfabddb1735f
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167205
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30360}
> >
> > TBR=steveanton@webrtc.org,kron@webrtc.org
> >
> > Change-Id: I1cc2d83bd884f10685503a9c31288f96c935d6a3
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167206
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30367}
>
> TBR=steveanton@webrtc.org,kron@webrtc.org
>
> Change-Id: I0a9b0b58922ce7c558b3d31b64cc12086b2a6a55
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167364
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30373}

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


Bug: chromium:1029737
Change-Id: Id381cb6d8e03b0fca941e392978362af6fdab0b6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167531
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30415}
 2020-01-29 18:53:54 +00:00
Bjorn A Mellem
0cda7b832a Allow non-identical datagram transport parameters. 
Currently, datagram transports must report identical transport
parameters in order to negotiate use of the datagram transport.  This is
not strictly necessary, they just need parameters that fit some notion
of "compatability" (eg. both ends share some mutually-supported version
of the datagram protocol).

This change allows datagram transports to implement their own notion of
compatible transport parameters, by adding a
SetRemoteTransportParameters method to DatagramTransportInterface which
checks if the remote parameters are compatible with the local endpoint
and returns an error if they are not.

Bug: webrtc:9719
Change-Id: I166c787b468b89d9082d7e3c9995a6ed50a1650a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167741
Commit-Queue: Bjorn Mellem <mellem@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30412}
 2020-01-29 18:14:24 +00:00
Ivo Creusen
182c2b8334 Expose run function to NetEqSimulator 
Bug: webrtc:11005
Change-Id: I84f01536b40ba17e66877cdced194e05b882b5c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/167537
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30405}
 2020-01-29 11:55:05 +00:00