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Author SHA1 Message Date
Tommi
891d393b80 Call Call::GetStats() from the correct thread in ProbingEndToEndTest. 
Also removing the stop_event_ from the RampUpTester class, which I missed in review 148067.

Bug: webrtc:10847
Change-Id: I102cc75287503915b51e37ea4ee01dfcc2437699
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148062
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28801}
 2019-08-08 06:40:26 +00:00
Tommi
aaaf804187 Call Call::GetStats() from the correct thread in VideoSendStreamTest. 
Bug: webrtc:10847
Change-Id: Ia471e132f35385d702f5d279a023ac7aa554069b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148060
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28800}
 2019-08-08 06:36:16 +00:00
chromium-webrtc-autoroll
efffd0a5fa Roll chromium_revision 3d0c04364f..5744654b26 (684897:685023) 
Change log: 3d0c04364f..5744654b26
Full diff: 3d0c04364f..5744654b26

Changed dependencies
* src/build: abdb0d3b19..899021679f
* src/ios: 000ce55673..01cac292ff
* src/testing: 8263776c61..15f9a2baf2
* src/third_party: 0a27208fc3..86ae88eee8
* src/third_party/googletest/src: a45c24ac18..90a443f9c2
* src/tools: 44859dd08c..654682c917
DEPS diff: 3d0c04364f..5744654b26/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I4debc0b1dfb8d4c0650ae5c2de6853e2a9a1482e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148500
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28799}
 2019-08-08 00:38:22 +00:00
chromium-webrtc-autoroll
307448fb7a Roll chromium_revision 006302cd2e..3d0c04364f (684781:684897) 
Change log: 006302cd2e..3d0c04364f
Full diff: 006302cd2e..3d0c04364f

Changed dependencies
* src/base: 77385a2080..613886a7f0
* src/build: a60cb97d12..abdb0d3b19
* src/ios: c602f600a7..000ce55673
* src/testing: 6bcc48bd0c..8263776c61
* src/third_party: c4a4be5d0d..0a27208fc3
* src/third_party/depot_tools: aa2db565b3..a110bf60c0
* src/tools: a157125f0d..44859dd08c
DEPS diff: 006302cd2e..3d0c04364f/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I0960b03dd054398c3bfbf4cccef9622021d5166e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148481
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28798}
 2019-08-07 20:32:47 +00:00
Alex Narest
5b5d97c938 Reland of "Reporting of decoding_codec_plc events"" 
This is a reland of 0a88ea050cda58de81d624cf2764d46929447ed5.

The new stat will not be reported unless it is GT 0.

Reporting of decoding_codec_plc events

Bug: webrtc:10838
Change-Id: Ic8585b4eeae9a2643374f15bc2578d1141e59683
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148448
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@google.com>
Cr-Commit-Position: refs/heads/master@{#28797}
 2019-08-07 18:41:46 +00:00
Steve Anton
2d2bbb16e5 Filter out duplicate receive codecs in the media engine 
A malformed session description can assign the same codec to
different payload types which would hit a DCHECK in the
WebRtcVideoEngine. This changes the video engine to just ignore
the duplicate payload type instead of failing.

Bug: chromium:987598
Change-Id: I2034dd11d315ef05448630c860c7ca3f69ef700b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147943
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Amit Hilbuch <amithi@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28796}
 2019-08-07 17:29:12 +00:00
chromium-webrtc-autoroll
3cc2f70bc3 Roll chromium_revision 192da69226..006302cd2e (684664:684781) 
Change log: 192da69226..006302cd2e
Full diff: 192da69226..006302cd2e

Changed dependencies
* src/base: 6c823a49bc..77385a2080
* src/build: fcd9f647c2..a60cb97d12
* src/ios: db54f52dd6..c602f600a7
* src/testing: 687322b2e9..6bcc48bd0c
* src/third_party: 45e877675e..c4a4be5d0d
* src/tools: f68f930ebc..a157125f0d
DEPS diff: 192da69226..006302cd2e/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I3311b7d6739be193c32b6a4e4d81f502dcfa9059
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148460
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28795}
 2019-08-07 16:32:42 +00:00
Oleh Prypin
b1686786e8 Add RTC_ prefix to non-standard format specifier macro "PRIdNS" 
Some of the macros in format_macros.h follow the C standard and try to fill holes in it (on Windows). But this one has no direct equivalent in the standard and is just mimicking the naming convention. That's not nice.

References:
https://devblogs.microsoft.com/cppblog/c99-library-support-in-visual-studio-2013/
https://stackoverflow.com/a/2524673

Change-Id: I53f3faca2976a5b5d4b04a67ffb56ae0f4e930b2
Bug: webrtc:10852
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147862
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28794}
 2019-08-07 13:36:05 +00:00
Niels Möller
12ebfa69ba Delete RtcpStatisticsCallback from ReceiveStatistics 
Update VideoReceiveStream::GetStats to use
StreamStatistician::GetStatistics instead, similarly to the audio
receiver.

Bug: webrtc:10679
Change-Id: I8a701e8a8e921c87895424362dc83500737c916d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/142233
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28793}
 2019-08-07 13:33:55 +00:00
Oleh Prypin
b6685420c4 Delete unused format specifier macros for NSInteger and NSUInteger 
Bug: webrtc:10852
Change-Id: I0f8ba32b3f22ebd1bb85602daeebd34fb3d8c544
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148441
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28792}
 2019-08-07 12:25:16 +00:00
Danil Chapovalov
83bbe91398 Delete deprecated rtc_event_log header 
Bug: webrtc:10206
Change-Id: I9ed3148843c647372993729b87c0e74741ab540b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147870
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28791}
 2019-08-07 10:58:17 +00:00
Chen Xing
e08648dc70 Add AbsoluteCaptureTime to RtpPacketInfo. 
This change stores the optional `AbsoluteCaptureTime` header extension in `RtpPacketInfo` so that we later can consume it in `SourceTracker`.

Bug: webrtc:10739
Change-Id: I975e8863117fcda134535cd49ad71079a7ff38ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148068
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28790}
 2019-08-07 10:12:56 +00:00
Per Åhgren
f40a340756 Remove deprecated code related to AEC2 
This CL removes code related to the usage of the delay agnostic and
extended filter modes in AEC2.

Bug: webrtc:8671
Change-Id: I1a2c7a9eba54b03f5a015df3adb617785f52a939
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/133912
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28789}
 2019-08-07 10:09:36 +00:00
Niels Möller
75caef7a4b Delete unused ACM members isac_decoder_16k_ and isac_decoder_32k_ 
Unused since https://webrtc-review.googlesource.com/c/103821.

Bug: webrtc:8396
Change-Id: Ia83f02f16d6ea8c260ea765b41506f2691e035bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148072
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28788}
 2019-08-07 09:53:22 +00:00
Sebastian Jansson
d2845f8246 Removes unused AudioAllocationSettings from voice engine. 
Bug: webrtc:9883
Change-Id: Ie322a1cae1f9682f64a05767f3933cba13b70ae0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148281
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28787}
 2019-08-07 09:36:42 +00:00
Tommi
d23f67e6be Call Call::GetStats() from the correct thread in StatsEndToEndTest. 
Bug: webrtc:10847
Change-Id: I8a82709073827f0eb901e20600f4e8bcf86d96a3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148061
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28786}
 2019-08-07 08:50:09 +00:00
Tommi
c24a5b18ba Fix CallPerfTests to call Call::GetStats() from the right thread. 
Bug: webrtc:10847
Change-Id: Ic44d9b644aff981ec570c98c2486bb8be787ac68
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148066
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28785}
 2019-08-07 08:49:04 +00:00
Niels Möller
c653172e74 Delete obsolete method AudioCodingModule::SetBitRate 
Bug: None
Change-Id: I2291f7b4b46d269592eacad67a126010b750fac2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148079
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28784}
 2019-08-07 08:37:25 +00:00
chromium-webrtc-autoroll
e71edc5e0d Roll chromium_revision 838e9d2793..192da69226 (684401:684664) 
Change log: 838e9d2793..192da69226
Full diff: 838e9d2793..192da69226

Changed dependencies
* src/base: f9fbebdb46..6c823a49bc
* src/build: c5d5866502..fcd9f647c2
* src/ios: 0e10d4dc90..db54f52dd6
* src/testing: a2ba4c1956..687322b2e9
* src/third_party: bd6ca9b364..45e877675e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c802efc06a..cad35e22dc
* src/third_party/depot_tools: 36756e4590..aa2db565b3
* src/third_party/freetype/src: ad3443c931..734d60f63c
* src/tools: 40f65277bd..f68f930ebc
DEPS diff: 838e9d2793..192da69226/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I1a587f06ca59ad489fa594ea97ccfc66893e8fa9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148381
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28783}
 2019-08-07 08:36:14 +00:00
Artem Titov
1e49ab2d40 Migrate part of Vp9 SVC tests on PC framework. Add temporal layers support. 
Bug: webrtc:10138
Change-Id: I3f0fc38cbe8c31a2aea2f231fed4428b39e3125a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147260
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28782}
 2019-08-07 04:18:46 +00:00
Artem Titov
8dcaed97a9 Split VideoFrameWriter into yuv and y4m writers 
Bug: webrtc:10138
Change-Id: I5eae094a1a4b426281d291273f7feb9555497139
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147645
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28781}
 2019-08-07 04:06:46 +00:00
chromium-webrtc-autoroll
9d62a5675e Roll chromium_revision 9d357a520c..838e9d2793 (684300:684401) 
Change log: 9d357a520c..838e9d2793
Full diff: 9d357a520c..838e9d2793

Changed dependencies
* src/base: 60deb682db..f9fbebdb46
* src/build: 1f2944270d..c5d5866502
* src/ios: 51fc3e12da..0e10d4dc90
* src/testing: 4531161454..a2ba4c1956
* src/third_party: e97bb11690..bd6ca9b364
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/131b9f3245..c802efc06a
* src/third_party/r8: 1ceH-5pheRe3OUeFz4TzVc2xcu6sm45yeFQBTT4rd6IC..6mmxCrRGY15GNu1r-XmGVxohZfruF6_XjiKSCZakkN4C
* src/tools: f6f1296e62..40f65277bd
DEPS diff: 9d357a520c..838e9d2793/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I09e549e66600e22b7862593af5fb0ff1e719a7b3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148300
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28780}
 2019-08-06 18:31:57 +00:00
Alex Drake
00c7ecf625 Surface CandidatePairChange event 
In order to be able to detect and measure context around candidate pair changes.

Bug: webrtc:10419
Change-Id: Iab0d7e7c80d925d1aa44617fc35975fdc6bbc6b9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147340
Commit-Queue: Alex Drake <alexdrake@google.com>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28779}
 2019-08-06 18:25:57 +00:00
Sebastian Jansson
63c38e21da Fix for incorrect transport sequence number config for audio in scenario tests. 
Bug: webrtc:9883
Change-Id: Iafe1db4b4dbfa81c7901640114057806821de760
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148280
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28778}
 2019-08-06 16:26:22 +00:00
Sebastian Jansson
7cbee84610 Reland "Adds PeerConnection scenario test framework." 
This is a reland of ad5c4accad00e04de08e2b62d366cc1f8e0320a5

It was flaky due to starting ICE signaling before SDP negotiation
finished. This was solved by adding an helper for adding ice candidates
which will wait until the peer connection is ready if needed.

Original change's description:
> Adds PeerConnection scenario test framework.
>
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}

Bug: webrtc:10839
Change-Id: I6eb8f482561c87e7b0f20d2431d21a41b26c91d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147877
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28777}
 2019-08-06 16:12:12 +00:00
Erik Språng
c64881925f DegradedCall: fake network using TaskQueue instead of ProcessThread 
Tested: Manually in Chrome webrtc calls.
Bug: webrtc:10851
Change-Id: I6bc6b7625101b39e4dd8b0efa5db213ab57980a8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148077
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28776}
 2019-08-06 15:05:30 +00:00
Mirko Bonadei
bb1f2454cd Disable RunPythonTests on rtc_tools. 
The only python test present in that folder has been removed by
https://webrtc-review.googlesource.com/c/src/+/146941.

TBR=tommi@webrtc.org

No-Try: True
Bug: None
Change-Id: Idb03c3b8acceae04ebbf5bb0d8db7cd3288b929d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148074
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28775}
 2019-08-06 12:48:33 +00:00
chromium-webrtc-autoroll
61b15905b0 Roll chromium_revision 8776a3887d..9d357a520c (684182:684300) 
Change log: 8776a3887d..9d357a520c
Full diff: 8776a3887d..9d357a520c

Changed dependencies
* src/base: 829a9fb70e..60deb682db
* src/build: c462f4b82d..1f2944270d
* src/ios: e2f722b278..51fc3e12da
* src/testing: aa61825872..4531161454
* src/third_party: e072174fb4..e97bb11690
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/698405a9f6..131b9f3245
* src/third_party/depot_tools: 921f6a1751..36756e4590
* src/tools: 544726b7aa..f6f1296e62
DEPS diff: 8776a3887d..9d357a520c/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Icc06b78f78fe64764ebada4d0c9e30efcb1d0cef
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148260
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28774}
 2019-08-06 12:40:23 +00:00
Artem Titov
2e6c294dba Refactor test_peer.cc to reduce amount of arguments passing around 
Bug: webrtc:10138
Change-Id: Ie89ec496bb42b0c1072d30870ffe5987ce780362
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148075
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28773}
 2019-08-06 12:07:53 +00:00
Tommi
e6b7b6678c Fix CallClient so that it calls Call::GetStats() on the right thread. 
Bug: webrtc:10847
Change-Id: Id23a389b4d5bad8f2211b5ec87b37aefc81a9292
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148065
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28772}
 2019-08-06 11:16:41 +00:00
Tommi
a22cab8a65 Calling DebugBreak() on Windows during fatal checks instead of relying on abort(). 
Bug: webrtc:10846
Change-Id: Ie701f47b9b7c9d88ea983540dce3f429903a18f7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147879
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28771}
 2019-08-06 09:19:00 +00:00
Niels Möller
7ba3b81ff5 Delete class PlatformFile. 
Add seek methods to FileWrapper, and refactor WavReader to use that
class instead.

Bug: webrtc:6463
Change-Id: Ifbb1989a072da6280ea5fc04b4beff991614dd53
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147265
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28770}
 2019-08-06 09:07:50 +00:00
Johannes Kron
10da4a0025 Fix RtpFrameReferenceFinderFuzzer to not generate invalid input 
Make sure that the packets in the packet buffer belonging to the
first and last sequence numbers are marked as first and last,
respectively.

Bug: chromium:989856
Change-Id: I57bdd7d62d585be2d2083a6b5ce67fce89ab4389
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147875
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28769}
 2019-08-06 09:03:10 +00:00
Tommi
c89468a4b7 Fix CallStatsUnittests to update the RTT on the process thread (as in production). 
Bug: webrtc:10848
Change-Id: Ib56ba887ef148188c902dee680b29809ae8cb8f6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148064
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28768}
 2019-08-06 08:46:48 +00:00
Niels Möller
4d7c405599 Split out RtcpCnameCallback from RtcpStatisticsCallback 
Cname callback is used only on receive side, and statistics (soon)
only on the send side.

Bug: webrtc:10679
Change-Id: I122e9cafaea93cd0ba75dc955a652d9d4bddc379
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147867
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28767}
 2019-08-06 08:29:57 +00:00
Niels Möller
ed44f5464a In ChannelReceive, use AcmReceiver directly, not AudioCodingModule 
Bug: webrtc:9801
Change-Id: I02d76bc89c363247c8dc782db316a9f87a2b93ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/111504
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28766}
 2019-08-06 08:23:26 +00:00
Tommi
e80885a89c Call Call::GetStats() from the correct thread in our bandwidth tests. 
Bug: webrtc:10847
Change-Id: Ief8cdd72f9d5b600d5306c00c1d249c29fb20396
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148063
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28765}
 2019-08-06 08:19:12 +00:00
Tommi
5e005f4b2d Fix RampUp tests to call Call::GetStats() from the right thread - and remove the need for a dedicated polling thread. 
Bug: webrtc:10847
Change-Id: I01492d2e385840e50d2d94f498063b5e4eea3665
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148067
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28764}
 2019-08-06 08:18:07 +00:00
chromium-webrtc-autoroll
bdc909653e Roll chromium_revision 2c4c2e2ea6..8776a3887d (684065:684182) 
Change log: 2c4c2e2ea6..8776a3887d
Full diff: 2c4c2e2ea6..8776a3887d

Changed dependencies
* src/base: 5957e8101a..829a9fb70e
* src/build: d2eb8b11ac..c462f4b82d
* src/ios: a2e9b860c7..e2f722b278
* src/testing: 82c3031d88..aa61825872
* src/third_party: 3eca64a63a..e072174fb4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a5468613fa..698405a9f6
* src/tools: d1819be25e..544726b7aa
DEPS diff: 2c4c2e2ea6..8776a3887d/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Ifc46a5e23e1ebcc0f2f25a9863c825304c816281
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148180
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28763}
 2019-08-06 00:32:17 +00:00
chromium-webrtc-autoroll
074f0d2d28 Roll chromium_revision 7c6275bdfa..2c4c2e2ea6 (683711:684065) 
Change log: 7c6275bdfa..2c4c2e2ea6
Full diff: 7c6275bdfa..2c4c2e2ea6

Changed dependencies
* src/base: 4c6ed4708b..5957e8101a
* src/build: c441c0a09d..d2eb8b11ac
* src/ios: ec642043d5..a2e9b860c7
* src/testing: 12f69d1a5b..82c3031d88
* src/third_party: 7448965c2a..3eca64a63a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/4f68ff26e4..a5468613fa
* src/third_party/depot_tools: 59bb8cce84..921f6a1751
* src/third_party/freetype/src: 12af46b649..ad3443c931
* src/tools: 494a8a0b5a..d1819be25e
DEPS diff: 7c6275bdfa..2c4c2e2ea6/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: I8f9014a0b42dfac47bb54c81752c2d30e760f8db
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148120
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#28762}
 2019-08-05 19:54:04 +00:00
Florent Castelli
9b1700cfae Enable field trial LegacySimulcastLayerLimit by default 
When using simulcast, if the source is too small, it might end up generating
layer sizes that are problematic for hardware encoders.

We can temporarily restore the old behavior that adapts the layer count to the source size until we fix the HW encoder behavior.
to fix HW encoder issues

Bug: webrtc:10849, chromium:990823
Change-Id: Ie1486c9209b408c797c92d1b319d4116fe77171b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148069
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28761}
 2019-08-05 16:11:11 +00:00
Per Åhgren
45231be79c AEC3: Removing unused code in the echo subtractor 
Bug: webrtc:8671
Change-Id: I77e9c55fe2e1030e5b74c02d4bc9222de422f6f4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137045
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28760}
 2019-08-05 14:53:59 +00:00
Per Åhgren
cdbaeeb737 Aec3:Remove unused legacy code 
Bug: webrtc:8671
Change-Id: I54f14cc304dbe1639fbb356dbdf9beeb1781ede4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/137801
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28759}
 2019-08-05 14:13:54 +00:00
Erik Språng
d7ee76cadd Wire up field trials for some experimental screenshare settings 
Bug: b/132074409
Change-Id: I83d5334255bad4fcf585b9850506bbfe1914ba57
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147868
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28758}
 2019-08-05 13:47:01 +00:00
Mirko Bonadei
8d4105850c Remove unused rtc_tools/video_analysis.py. 
Bug: None
Change-Id: I5ad3744afc809ed468dd3eb509a36332f4c9449e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/146941
Reviewed-by: Christoffer Jansson <jansson@google.com>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28757}
 2019-08-05 12:40:32 +00:00
Chen Xing
b56cca3282 Remove the old ContributingSources class. 
This change removes the old `ContributingSources` class. It has been replaced by the new `SourceTracker`.

Bug: webrtc:10793
Change-Id: Ibd481cf6584837c46b229b9fc2a071362f07d361
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147878
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Chen Xing <chxg@google.com>
Cr-Commit-Position: refs/heads/master@{#28756}
 2019-08-05 12:30:22 +00:00
Sebastian Jansson
3d351c6885 Revert "Adds PeerConnection scenario test framework." 
This reverts commit ad5c4accad00e04de08e2b62d366cc1f8e0320a5.

Reason for revert: Breaks downstream bots.

Original change's description:
> Adds PeerConnection scenario test framework.
> 
> Bug: webrtc:10839
> Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
> Commit-Queue: Sebastian Jansson <srte@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#28754}

TBR=steveanton@webrtc.org,sprang@webrtc.org,srte@webrtc.org

Change-Id: I35576b4afe100a3220c3c01a6a6d5fbdf48a258b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:10839
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147876
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28755}
 2019-08-05 10:46:25 +00:00
Sebastian Jansson
ad5c4accad Adds PeerConnection scenario test framework. 
Bug: webrtc:10839
Change-Id: If67eeb680d016d66c69d8e761a88c240e4931a5d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147276
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28754}
 2019-08-05 10:12:43 +00:00
Åsa Persson
139f4dc7ac QualityScaler: Add option to try fast adapt down at start up based on initial bw estimates. 
optional<int> initial_bitrate_interval_ms: time interval since start of call
where fast adapt down is allowed.
optional<double> initial_bitrate_factor: try fast adapt down if bw estimate is
below initial bitrate * factor.

Bug: none
Change-Id: I63e1fdaac6556d8e9a961a42e11c925f9ecb9771
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147725
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28753}
 2019-08-05 09:43:19 +00:00
Artem Titov
fedd625e0c Change 2g network pc audio test to more realistic network 
Bug: webrtc:10138
Change-Id: I6f4b23fe702d26dbbeed05d0d09b79a9a966e40c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/147728
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#28752}
 2019-08-02 22:28:02 +00:00