Commit Graph

1648 Commits

Author SHA1 Message Date
206ccd0b36 Merge remote tracking branch 'upstream-master'
Bug: 153469641
Test: run cuttlefish locally
Change-Id: Ida3bfe62ef5c6549278f4c155a1f690b008e9b9d
2020-07-23 13:08:51 -07:00
1062cfee8d Add list of local_addresses for network stats object
local_addresses is a list of IPs that were used to send data, which was
used during stats calculation.

Bug: webrtc:11756
Change-Id: Ie6307eaa69c73ebe9f69e44503752151be9e9ef6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179841
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31774}
2020-07-21 16:17:02 +00:00
a5d9c1a45c In DependencyDescriptor rtp header extension drop partial chain support
i.e. when chain are used,
require each decode target to be protected by some chain.
where previously it was allowed to mark decode target as unprotected.

See https://github.com/AOMediaCodec/av1-rtp-spec/pull/125

Bug: webrtc:10342
Change-Id: Ia2800036e890db44bb1162abfa1a497ff68f3b24
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178807
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31772}
2020-07-21 14:01:27 +00:00
c1a0737855 Add per source network statistics
Add ability to obtain incoming data network statistic splitted by data
source IP address.

Bug: webrtc:11756
Change-Id: I023c99f6bd19363a52a358dba52d25cd507097ae
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179368
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31770}
2020-07-21 09:03:34 +00:00
889b21a2e1 Generate new Android.bp file and correct build errors
The following are not yet available in their respective libraries so
attempts to use it in webrtc result in a call to abort():
* libvpx's CONSTRAINED_FROM_ABOVE_DROP constant
* libyuv's I010 buffers

The original webrtc project expects to have third party libraries
checked out in third_party/ and base/third_party/. Added some headers
in those libraries with a single line including the right header from
external/<library>. Updated .gitignore to keep track of said headers.

Bug: 153469641
Test: mm, also built cuttlefish using this library and ran it locally
Change-Id: I2d596942e34093dccc65d4b7b8249b6afc14d31f
2020-07-17 14:29:21 -07:00
85770cd25b Merge branch 'upstream-master'
Bug: 153469641
Test: none
Change-Id: Ic33e363deb0d1ac4bb4d57c3c239eb2e45370802
2020-07-17 14:25:15 -07:00
e6542f2112 Removed unused include from encoded_image.h.
Bug: webrtc:9378
Change-Id: Ie26ab4d30d62ec109a8be638661789399821c162
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179525
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31758}
2020-07-17 14:14:03 +00:00
3d2210876e Remove unused critical section includes.
Bug: webrtc:11567
Change-Id: Ic5e43c51ce06c0619adc265d12ad4bef73a9df76
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179521
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31745}
2020-07-16 13:52:28 +00:00
3592839896 Add default values for VideoEncoderFactory::CodecInfo
So that applications can construct a default struct without naming the
members.

Bug: None
Change-Id: Idd9028bee9016670e776f17a62077eb9c34d6f2a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179485
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirta Dvornicic <mirtad@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31742}
2020-07-16 11:12:35 +00:00
9ad1f6feca Reland "Delete PeerConnectionInterface::BitrateParameters"
This is a reland of e2dfe74b0e29558ddea6473d0272bc38c838370c
Downstream breakage has been fixed.

Original change's description:
> Delete PeerConnectionInterface::BitrateParameters
>
> Replaced by the api struct BitrateSettings, added in
> https://webrtc-review.googlesource.com/74020
>
> Bug: None
> Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31714}

Tbr: kwiberg@webrtc.org
Bug: None
Change-Id: Ic039e51f9f842329525887a28d1cb9819addc74b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179282
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31728}
2020-07-15 07:35:16 +00:00
f60d4c2dfe Revert "Delete PeerConnectionInterface::BitrateParameters"
This reverts commit e2dfe74b0e29558ddea6473d0272bc38c838370c.

Reason for revert: Breaks downstream project

Original change's description:
> Delete PeerConnectionInterface::BitrateParameters
> 
> Replaced by the api struct BitrateSettings, added in
> https://webrtc-review.googlesource.com/74020
> 
> Bug: None
> Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Taylor <deadbeef@webrtc.org>
> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31714}

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

Change-Id: Ia62b3c43996e95668d7972882baf06a186a539d3
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179221
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31719}
2020-07-13 15:41:39 +00:00
e2dfe74b0e Delete PeerConnectionInterface::BitrateParameters
Replaced by the api struct BitrateSettings, added in
https://webrtc-review.googlesource.com/74020

Bug: None
Change-Id: I8b50b32f5c7a8918fad675904d913a21fd905274
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177665
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Taylor <deadbeef@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31714}
2020-07-13 10:06:42 +00:00
edacbd53de Reland "Implement packets_(sent | received) for RTCTransportStats"
This is a reland of fb6f975401972635a644c0db06c135b4c0aaef4a. Related
issue in chromium is fixed here:
https://chromium-review.googlesource.com/c/chromium/src/+/2287294

Original change's description:
> Implement packets_(sent | received) for RTCTransportStats
>
> Bug: webrtc:11756
> Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31643}

Bug: webrtc:11756
Change-Id: I1e310e3d23248500eb7dabd23d0ce6c4ec4cb8c6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178871
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31700}
2020-07-10 11:50:59 +00:00
4e5bc9f081 Reland "Complete migration from "track" to "inbound-rtp" stats"
This is a reland of 94fe0d3de5e8162d1a105fd1a3ec4bd2da97f43b with a fix.

Original change's description:
> Complete migration from "track" to "inbound-rtp" stats
>
> Bug: webrtc:11683
> Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Cr-Commit-Position: refs/heads/master@{#31683}

Bug: webrtc:11683
Change-Id: I173b91625174051c02ff34127aaf6c086d3c5c66
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179060
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31696}
2020-07-10 10:17:50 +00:00
e6f3897945 Revert "Complete migration from "track" to "inbound-rtp" stats"
This reverts commit 94fe0d3de5e8162d1a105fd1a3ec4bd2da97f43b.

Reason for revert:
Causes an assert in this line during a call:
https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/sdk/objc/api/peerconnection/RTCStatisticsReport.mm;drc=87a6e5ab4d8f0baf4e2a9b7752b43d825f9c0ce1;l=122?originalUrl=https:%2F%2Fcs.chromium.org%2F

where frameWidth appears more than once

Original change's description:
> Complete migration from "track" to "inbound-rtp" stats
> 
> Bug: webrtc:11683
> Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Eldar Rello <elrello@microsoft.com>
> Cr-Commit-Position: refs/heads/master@{#31683}

TBR=hbos@webrtc.org,hta@webrtc.org,elrello@microsoft.com

Change-Id: I0ded36a40c8808dac5a777ed41815e52ab9a2573
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179020
Reviewed-by: Zeke Chin <tkchin@webrtc.org>
Commit-Queue: Zeke Chin <tkchin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31692}
2020-07-10 00:06:53 +00:00
94fe0d3de5 Complete migration from "track" to "inbound-rtp" stats
Bug: webrtc:11683
Change-Id: I4c4a4fa0a7d6a20976922aca41d57540aa27fd1d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178611
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Cr-Commit-Position: refs/heads/master@{#31683}
2020-07-09 10:02:26 +00:00
059f4f74df Cleanup setting VideoQualityTestFixtureInterface::Param struct
put default values next to the (named) members.
Remove constructor
declared as VideoQualityTestFixtureInterface::Param::Param,
but defined as VideoQualityTest::Param::Param

avoid braced initialisation for structures with lot's of fields that change over time

Bug: None
Change-Id: I647e3ace47ae981c515d6939274cb2744dd1b866
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178811
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@{#31674}
2020-07-08 13:53:57 +00:00
9b35da880b Revert "Implement packets_(sent | received) for RTCTransportStats"
This reverts commit fb6f975401972635a644c0db06c135b4c0aaef4a.

Reason for revert: Looks like this breaks chromium.webrtc.fyi:
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Mac%20Tester/6000
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/6209
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win7%20Tester/6177
https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win8%20Tester/6299

Original change's description:
> Implement packets_(sent | received) for RTCTransportStats
> 
> Bug: webrtc:11756
> Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741
> Reviewed-by: Tommi <tommi@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Commit-Queue: Artem Titov <titovartem@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31643}

TBR=hbos@webrtc.org,tommi@webrtc.org,titovartem@webrtc.org

Change-Id: Icbb0974ba29cbddb614f1f37f8a2de1a7c56b571
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:11756
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178868
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31665}
2020-07-08 09:42:41 +00:00
656efbee6f Fix thread usage in PC level tests for getting to the IceConnected state
Bug: webrtc:11743
Change-Id: I18a6318c35b350b3d729bbd5ac1d25f035e6ad9d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178809
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31658}
2020-07-08 08:15:32 +00:00
fb6f975401 Implement packets_(sent | received) for RTCTransportStats
Bug: webrtc:11756
Change-Id: Ic0caad6d4675969ef3ae886f50326e4a2e1cbfe7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178741
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31643}
2020-07-07 10:45:05 +00:00
0ef4a2488a Add simulated time support for PC level test.
Bug: webrtc:11743
Change-Id: If69ab07618a30ec1a66dd5f36b3198486bee55fa
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178608
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Andrey Logvin <landrey@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31639}
2020-07-06 21:18:00 +00:00
a018919bf6 Remove old method for creating PC test fixture
Bug: webrtc:11743
Change-Id: Ia57681c8ddf1b352be275fa22198c4d671f90efe
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178604
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31624}
2020-07-03 13:38:47 +00:00
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
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
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
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
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
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
39adce1498 Add RtpEncodingParameters.adaptive_ptime.
When enabled:
- Creates an audio network adapter config that is passed to audio send
stream.
- Configures a lower default min bitrate.

All parameters can be configured via a field trial that can also force
enable the audio network adaptor (this is mainly intended for testing).

Bug: chromium:1086942
Change-Id: I48dfcca1ee2948084199352abed6212a6c78eb6c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177840
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Jakob Ivarsson <jakobi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31565}
2020-06-25 14:51:13 +00:00
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
755c65d8b5 Reland RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
This change adds exposure of a new transceiver method for
modifying the extensions offered in the next SDP negotiation,
following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.

Features:
- The interface allows to control the negotiated direction as
  per https://tools.ietf.org/html/rfc5285#page-7.
- The interface allows to remove an extension from SDP
  negotiation by modifying the direction to
  RtpTransceiverDirection::kStopped.

Note: support for signalling directionality of header extensions
in the SDP isn't implemented yet.

https://chromestatus.com/feature/5680189201711104.
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk

Tested: new unit tests in CL and manual tests with downstream project.
Bug: chromium:1051821
Change-Id: I7a4c2f979a5e50e88d49598eacb76d24e81c7c7a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177348
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31554}
2020-06-24 10:38:30 +00:00
24762f207f Fix missing dependencies.
Setting gtest_enable_absl_printers to false in .gn uncovers some missing
dependencies that were pulled in by gtest.

Bug: None
Change-Id: Ibd7772f6e2af9c798c97161c24f70b1658e3723c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177843
Reviewed-by: Jeremy Leconte <jleconte@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31551}
2020-06-23 15:46:34 +00:00
09867d37ed Share constants for dependency descriptor rtp header extension
Bug: webrtc:10342
Change-Id: I9c81215569bd1bd96b953faa359f5a3d32c7d0c4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177521
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31549}
2020-06-22 11:58:29 +00:00
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
4db954eec1 Add RTC_EXPORT to webrtc::Resource so that it can be used in Chrome.
This is needed because chromium build targets need to be exported for
its component builds.

// TBR because this is a purely building related change and it has been
// reviewed by mbonadei@.
TBR=stefan@webrtc.org

Bug: webrtc:11525
Change-Id: I97f0c814b11e7fad86eeff319e644ae51204c3b8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177341
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31534}
2020-06-16 18:02:00 +00:00
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
1f0f59f02f Add ValueOrDefault to RTCStatsMember
Bug: webrtc:10430
Change-Id: I0d0d7e3664f7b027c7efe3166559f7b35bef0194
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177245
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Andrey Logvin <landrey@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31525}
2020-06-15 16:06:43 +00:00
9465978a3b Remove framemarking RTP extension.
BUG=webrtc:11637

Change-Id: I47f8e22473429c9762956444e27cfbafb201b208
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176442
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31522}
2020-06-15 11:18:00 +00:00
6f727da62b Revert "RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions."
This reverts commit 71db9acc4019b8c9c13b14e6a022cbb3b4255b09.

Reason for revert: breaks downstream project.
Reason for force push: win bot broken.

Original change's description:
> RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
>
> This change adds exposure of a new transceiver method for
> modifying the extensions offered in the next SDP negotiation,
> following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.
>
> Features:
> - The interface allows to control the negotiated direction as
>   per https://tools.ietf.org/html/rfc5285#page-7.
> - The interface allows to remove an extension from SDP
>   negotiation by modifying the direction to
>   RtpTransceiverDirection::kStopped.
>
> Note: support for signalling directionality of header extensions
> in the SDP isn't implemented yet.
>
> https://chromestatus.com/feature/5680189201711104.
> Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk
>
> Bug: chromium:1051821
> Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Commit-Queue: Markus Handell <handellm@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#31487}

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

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

No-Try: true
Bug: chromium:1051821
Change-Id: I70e1a07225d7eeec7480fa5577d8ff647eba6902
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177103
Commit-Queue: Markus Handell <handellm@webrtc.org>
Reviewed-by: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31516}
2020-06-12 16:26:49 +00:00
4c1e7cc19b [Adaptation] Add ability to inject resources on the PeerConnection.
This unblocks injecting platform-specific resources, such as power
usage signals in Chrome.

This CL adds AddAdaptationResource to PeerConnectionInterface and
integration tests verifying that if an injected resource is overusing,
resolution will soon be reduced.

To aid testing, some testing-only classes have been updated.

Bug: webrtc:11525
Change-Id: I820099e79f18d910fd641ee1412ad064b99ebce9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177003
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31505}
2020-06-11 14:17:01 +00:00
24263f4ffb Embed FrameDependencyTemplate builder helpers directly into the struct
Bug: None
Change-Id: I4c13bdabd08dd6a6011cb534c765c1dd09f218d1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176843
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31500}
2020-06-11 13:43:51 +00:00
f4a9991cce [Adaptation] Adding adaptation resources from Call.
This CL adds AddAdaptationResource to Call and
AddAdaptationResource/GetAdaptationResources method to relevant
VideoSendStream and VideoStreamEncoder interfaces and implementations.

Unittests are added to ensure that resources can be added to the Call
both before and after the creation of a VideoSendStream and that the
resources always gets added to the streams.

In a follow-up CL, we will continue to plumb the resources all the way
to PeerConnectionInterface, and an integration test will then be added
to ensure that injected resources are capable of triggering adaptation.

Bug: webrtc:11525
Change-Id: I499e9c23c3e359df943414d420b2e0ce2e9b2d56
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177002
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Evan Shrubsole <eshr@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31499}
2020-06-11 12:43:21 +00:00
9276e2c39b Remove enable_simulcast_stats config flag as not needed anymore
Bug: webrtc:9547
Change-Id: Ie50453aa3496d16bfadfc9fdd3e7e6982278cfba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176841
Commit-Queue: Eldar Rello <elrello@microsoft.com>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31492}
2020-06-10 15:59:32 +00:00
0ca13d97d2 Introduce BYPASS_PROXY_CONSTMETHOD0.
This allows const getters that query const state to be called without
marshalling calls between threads. This must not be used to
return pointers/references etc.

I'm starting by using this macro with the data channel which has a
few of these getters, as well as changing things a bit to make more
parts of the implementation, const.

Change-Id: I6ec7a3774cd8f7be2ef122fb7c7fc5919afee600
Bug: webrtc:11547
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176846
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31489}
2020-06-10 13:52:36 +00:00
71db9acc40 RtpTransceiverInterface: introduce SetOfferedRtpHeaderExtensions.
This change adds exposure of a new transceiver method for
modifying the extensions offered in the next SDP negotiation,
following spec details in https://w3c.github.io/webrtc-extensions/#rtcrtptransceiver-interface.

Features:
- The interface allows to control the negotiated direction as
  per https://tools.ietf.org/html/rfc5285#page-7.
- The interface allows to remove an extension from SDP
  negotiation by modifying the direction to
  RtpTransceiverDirection::kStopped.

Note: support for signalling directionality of header extensions
in the SDP isn't implemented yet.

https://chromestatus.com/feature/5680189201711104.
Intent to prototype: https://groups.google.com/a/chromium.org/g/blink-dev/c/65YdUi02yZk

Bug: chromium:1051821
Change-Id: Iaabc34446f038c46d93c442e90c2a77f77d542d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176408
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Markus Handell <handellm@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31487}
2020-06-10 13:02:44 +00:00
5ad16a50fc suppress 'Paramaterized test suite never instantiated' warning
for TaskQueueTest.
This suit is instantiated in the different binary targets by design.

Bug: None
Change-Id: I99a38e2461ee9bd06dfe68758490afe75c2475ba
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176750
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31476}
2020-06-09 13:31:49 +00:00
bf3f65815f Adding brandtr as api/video owner
Bug: None
Change-Id: Idacd6442d014e610aba363519c7159564ef80a45
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176700
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Magnus Flodman <mflodman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31470}
2020-06-09 08:30:19 +00:00
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
10ef847289 Correct name of DC.dataChannelIdentifier stats member
Bug: webrtc:8787
Change-Id: Ie32b38f0671e89e94017f439de7614142328642f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176509
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31457}
2020-06-07 21:57:50 +00:00
8b7cfa16aa Introduce absl_deps in rtc_* C++ templates.
Since dependencies on Abseil need to be statically linked in case
Chromium is built with is_component_build=true, this CL introduces a new
parameter for C++ library rtc_* templates (rtc_library, rtc_source_set
and rtc_static_library). This parameter (called "absl_deps") will result
in a dependency on the Abseil component (//third_party/abseil-cpp:absl)
when is_component_build=true or on the normal granular Abseil target
when is_component_build=false.

Bug: chromium:1046390
Change-Id: Iddca886926a7874488701bc9d79affb00cca72d0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/176447
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#31441}
2020-06-04 08:41:32 +00:00