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Author SHA1 Message Date
Sam Zackrisson
308bc646e0 Remove one acquisition of capture lock in APM AudioFrame API 
This brings the two ProcessStream functions closer in implementation.
Additionally, the error checking that is currently done in the period of not holding the lock seems cheaper than releasing and reacquiring the capture lock.

Bug: webrtc:11235
Change-Id: Ib4afc68afb419fcabbb8cf08a3a2e61d2c12acda
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163021
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30140}
 2020-01-03 10:56:24 +00:00
Per Åhgren
2bd85ab039 Avoid AGC2 runtime allocation and activate it on demand 
This CL ensures that the AGC2 is created and initialized only when
needed.

Apart from that, the CL also avoids a runtime-reallocation that happens
each time the setting is applied.

Bug: webrtc:5298
Change-Id: Iad9eaa05a3d0baa0788cd11b2aa17ddd8e0c509b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163987
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30139}
 2020-01-03 10:47:14 +00:00
Niels Möller
83ee98292f Delete p2p/base/packet_transport_interface.h 
This file only defined an unused alias.

Bug: None
Change-Id: I0c731401295814e8f5dd91f41350973021efd5d5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/155173
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30138}
 2020-01-03 08:58:04 +00:00
Per Åhgren
c0734715d1 APM: Move the TransientSuppression activation to the apm_config 
This CL moves the activation of the transient suppression to the APM
config.

Bug: webrtc:5298
Change-Id: Iba7975bec4654c3df8834fd5b7d1082ff53641dd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163985
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30137}
 2020-01-03 08:48:54 +00:00
chromium-webrtc-autoroll
045c36d17c Roll chromium_revision 81693dc9aa..d4992c6f92 (727940:728071) 
Change log: 81693dc9aa..d4992c6f92
Full diff: 81693dc9aa..d4992c6f92

Changed dependencies
* src/build: 19d4aa51af..41f432e3bf
* src/ios: 7befbd01d7..71eba10063
* src/testing: a090bf66d7..5e994c4758
* src/third_party: 291f5c9f97..1f1d75d8c1
* src/third_party/freetype/src: 7e1b39f6cd..10d8de7541
* src/tools: 1bf5fec63c..006ad6a9d6
DEPS diff: 81693dc9aa..d4992c6f92/DEPS

No update to Clang.

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

Change-Id: I450276bbf9e2113d9951663e24c5b827b0029ea0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164457
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@{#30136}
 2020-01-03 00:44:07 +00:00
Natalie Chouinard
65bbcabe2f [Android] Replace java_files with sources 
Replace all usages of java_files with sources in gn files, and
automatically format.

This is in preparation for java_files being completely removed upstream
in favor of sources.

NOPRESUBMIT=true

Bug: chromium:1035074
Change-Id: Ib9a698740b7ad26a127031d90321c7ae2feb59bf
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163161
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Natalie Chouinard <chouinard@google.com>
Cr-Commit-Position: refs/heads/master@{#30135}
 2020-01-02 20:08:20 +00:00
Per Åhgren
29fec66c77 AEC3: Remove metrics that are not used for analysis 
Bug: webrtc:8671
Change-Id: I12a6584a70e2b56e0926c07999c919272499c255
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163981
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30134}
 2020-01-02 16:23:43 +00:00
Alex Narest
d2fb5f510f Fixes WebRtcAudioTrack crash while stopping 
TBR=henrika@webrtc.org

Bug: webrtc:11248
Change-Id: I5b829b5193d2accdfbf1e06c5317a5cd441c48c3
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163990
Commit-Queue: Alex Narest <alexnarest@google.com>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30133}
 2020-01-02 16:03:54 +00:00
Yves Gerey
a688d11d96 Return unavailable rate rather than garbage value. 
This CL quiets UBSan when value doesn't fit uint32_t.

Bug: webrtc:11182
Change-Id: I8a45867be9aaceeb490db1a3747eb0efc6eb6a8f
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163983
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30132}
 2020-01-02 15:55:24 +00:00
Per Åhgren
cf4c872dbd APM: Make the GetStatistics call independent of the locks in APM 
This CL changes the GetStatistics call in the audio processing module
(APM) to not aquire the render or capture locks in APM, while still
being thread-safe.
This change eliminates the risk of thread-priority inversion due to the
GetStatistics call.

Apart from the above the CL:
-Corrects the GetStatistics to not be const (it was const even though it
 aquired locks).
-Slightly changes the statistics reporting, so that the stats received
may be older than the most recent stats reported.

Bug: webrtc:11241
Change-Id: I00deb5507e004cbe6e4a19a8bad357491f86f4ab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163982
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30131}
 2020-01-02 15:45:14 +00:00
chromium-webrtc-autoroll
a43777dead Roll chromium_revision 2d48822491..81693dc9aa (727839:727940) 
Change log: 2d48822491..81693dc9aa
Full diff: 2d48822491..81693dc9aa

Changed dependencies
* src/base: de91707f3e..77a8fe72cc
* src/build: cac0fb467a..19d4aa51af
* src/buildtools: fa02977a1a..8d21328415
* src/buildtools/linux64: git_revision:6feb55993083dfd27b93da195c8a82a3a9529848..git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5
* src/buildtools/mac: git_revision:6feb55993083dfd27b93da195c8a82a3a9529848..git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5
* src/buildtools/win: git_revision:6feb55993083dfd27b93da195c8a82a3a9529848..git_revision:a5bcbd726ac7bd342ca6ee3e3a006478fd1f00b5
* src/ios: 9bfe3aa33d..7befbd01d7
* src/testing: f66f73a419..a090bf66d7
* src/third_party: 10bd75beb4..291f5c9f97
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c9e75ab1ff..7c4300cb19
* src/tools: 938fc63fed..1bf5fec63c
DEPS diff: 2d48822491..81693dc9aa/DEPS

No update to Clang.

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

Change-Id: I10574b49ad9f27b11b1bb8955f3b2c60bbd69703
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164449
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@{#30130}
 2020-01-02 04:18:59 +00:00
chromium-webrtc-autoroll
1c34ca7676 Roll chromium_revision f19d6cb823..2d48822491 (727734:727839) 
Change log: f19d6cb823..2d48822491
Full diff: f19d6cb823..2d48822491

Changed dependencies
* src/build: 54b7873dba..cac0fb467a
* src/ios: 59c7a48c58..9bfe3aa33d
* src/testing: 766cc08c49..f66f73a419
* src/third_party: a946678e1a..10bd75beb4
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6043069708..c9e75ab1ff
* src/tools: 786d92bae9..938fc63fed
DEPS diff: f19d6cb823..2d48822491/DEPS

No update to Clang.

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

Change-Id: Ia6453450478cd37200a94d204c2d568277fd6cde
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164281
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@{#30129}
 2019-12-31 02:43:18 +00:00
chromium-webrtc-autoroll
9d2c2dba28 Roll chromium_revision 9986f2241a..f19d6cb823 (727633:727734) 
Change log: 9986f2241a..f19d6cb823
Full diff: 9986f2241a..f19d6cb823

Changed dependencies
* src/build: 8339882a89..54b7873dba
* src/ios: 10c77ec97a..59c7a48c58
* src/testing: 240d660c68..766cc08c49
* src/third_party: 53c0cac961..a946678e1a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/263d57d376..6043069708
* src/tools: 3275bdd803..786d92bae9
DEPS diff: 9986f2241a..f19d6cb823/DEPS

No update to Clang.

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

Change-Id: I672bc7be941b1bb60c5d4e71d46ed39e698f097a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/164268
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@{#30128}
 2019-12-30 10:18:30 +00:00
chromium-webrtc-autoroll
8ac79125c0 Roll chromium_revision d814fc7ea9..9986f2241a (727531:727633) 
Change log: d814fc7ea9..9986f2241a
Full diff: d814fc7ea9..9986f2241a

Changed dependencies
* src/build: 69e421a3a6..8339882a89
* src/ios: f230393662..10c77ec97a
* src/third_party: 533ae73635..53c0cac961
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/189074525f..263d57d376
* src/tools: c83a8ed9c4..3275bdd803
DEPS diff: d814fc7ea9..9986f2241a/DEPS

No update to Clang.

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

Change-Id: Icadfbe9e5de439a93fa8e7e45399c342d71c0e6a
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163969
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@{#30127}
 2019-12-27 22:41:49 +00:00
chromium-webrtc-autoroll
f2dc05978f Roll chromium_revision 6f7e5e79ce..d814fc7ea9 (727038:727531) 
Change log: 6f7e5e79ce..d814fc7ea9
Full diff: 6f7e5e79ce..d814fc7ea9

Changed dependencies
* src/base: 96567c8f45..de91707f3e
* src/build: 8b4f17ac5f..69e421a3a6
* src/ios: 108bcc40ba..f230393662
* src/testing: 5a36a5534b..240d660c68
* src/third_party: 9056dff38d..533ae73635
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/efb804d60c..189074525f
* src/tools: 41fb035d80..c83a8ed9c4
DEPS diff: 6f7e5e79ce..d814fc7ea9/DEPS

No update to Clang.

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

Change-Id: I23a84892cc04d70b24a7906c8e1c61f5fb5667c5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/163960
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@{#30126}
 2019-12-27 04:39:05 +00:00
Danil Chapovalov
26762d0425 Add video codec AV1 to the deprecated android decoder/encoder wrappers 
modifying java enum with new value was overlooked in
https://webrtc-review.googlesource.com/c/src/+/159282

Bug: b/146586166
Change-Id: I2c9d2a7a807a8ddabc2704bf1de7b697c6977a7d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162903
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30125}
 2019-12-23 09:29:46 +00:00
chromium-webrtc-autoroll
38a55a0487 Roll chromium_revision 937a78378f..6f7e5e79ce (726883:727038) 
Change log: 937a78378f..6f7e5e79ce
Full diff: 937a78378f..6f7e5e79ce

Changed dependencies
* src/base: 4732007ec7..96567c8f45
* src/build: 60770dc780..8b4f17ac5f
* src/ios: accdfb9cce..108bcc40ba
* src/testing: 49838f15f5..5a36a5534b
* src/third_party: 0cb550d337..9056dff38d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/47e7bed708..efb804d60c
* src/third_party/depot_tools: 8b876bd407..44134341fa
* src/tools: 0797f716b7..41fb035d80
DEPS diff: 937a78378f..6f7e5e79ce/DEPS

No update to Clang.

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

Change-Id: Ib0c2c5ba3b43d85947c2a6ff7b6865203d29c734
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162950
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@{#30124}
 2019-12-21 10:41:45 +00:00
Yves Gerey
a79fc591df Roll chromium_revision e4c6d7fe53..937a78378f (726742:726883) 
Manual tweak: do not roll src/third_party

Change log: e4c6d7fe53..937a78378f
Full diff: e4c6d7fe53..937a78378f

Changed dependencies
* src/base: f7fb459610..4732007ec7
* src/build: 032b1bd069..60770dc780
* src/ios: d292120bff..accdfb9cce
* src/testing: c5705ae7a3..49838f15f5
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8c77d963b8..47e7bed708
* src/third_party/depot_tools: 05934953bf..8b876bd407
* src/third_party/libvpx/source/libvpx: b7e03724b3..50d1a4aa72
* src/tools: 78adf6b3a0..0797f716b7
DEPS diff: e4c6d7fe53..937a78378f/DEPS

No update to Clang.

TBR=yvesg@webrtc.org,marpan@webrtc.org, jianj@chromium.org,
BUG=None

Change-Id: I45ffbcae85147c43ab9230f415fff5acfe2a7ba8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162905
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30123}
 2019-12-20 19:55:21 +00:00
Olga Sharonova
b5159fe4a7 Revert "Reland "Reland "Distinguish between send and receive video codecs""" 
This reverts commit 4e64e605894df287178c5a1b537fbe859b7d420c.

Reason for revert: breaks a bunch of WebRtcBrowserTests on Win: https://ci.chromium.org/p/chromium/builders/webrtc.fyi/WebRTC%20Chromium%20FYI%20Win10%20Tester/4843


Original change's description:
> Reland "Reland "Distinguish between send and receive video codecs""
> 
> This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228
> 
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> >
> > This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> >
> > Reason for revert: Downstream test updated.
> >
> > Original change's description:
> > > Revert "Reland "Distinguish between send and receive video codecs""
> > >
> > > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > >
> > > Reason for revert: Breaks another downstream test.
> > >
> > > Original change's description:
> > > > Reland "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > >
> > > > Reason for revert: Downstream tests have been updated.
> > > >
> > > > Original change's description:
> > > > > Revert "Distinguish between send and receive video codecs"
> > > > >
> > > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > >
> > > > > Reason for revert: Breaks downstream test.
> > > > >
> > > > > Original change's description:
> > > > > > Distinguish between send and receive video codecs
> > > > > >
> > > > > > Even though send and receive codecs are 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: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > >
> > > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > >
> > > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > > No-Presubmit: true
> > > > > No-Tree-Checks: true
> > > > > No-Try: true
> > > > > Bug: chromium:1029737
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > >
> > > > Bug: chromium:1029737
> > > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30078}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30079}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> >
> > Bug: chromium:1029737
> > Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30097}
> 
> Bug: chromium:1029737
> Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30120}

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

Change-Id: I709ee0eb6246aa79dde3aacfc4c47e070c4e90ea
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162904
Reviewed-by: Olga Sharonova <olka@webrtc.org>
Commit-Queue: Olga Sharonova <olka@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30122}
 2019-12-20 13:57:12 +00:00
Andrey Efremov
f6b875c8a7 Fixed crash on iOS13, methods beginGeneratingDeviceOrientationNotifications and endGeneratingDeviceOrientationNotifications. 
1. On iOS13 the implementation of methods begin- and endGeneratingDeviceOrientationNotifications changed and now are looks like "not threadsafe" (in specific sence) - they should be called only on the main thread. This fact is not documented. And may be a mistake.

2. By the Apple official documentation methods begin- and endGeneratingDeviceOrientationNotifications should be balanced. (Each begin- method should be balanced with end- method.)

By the reason two above facts they consequences merged and produced the "floating" NSInternalInconsistencyException crash.

Bug: webrtc:11216
Change-Id: Ibedd5bba7476cc687de3b9b04be49e3cceac1d27
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162205
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30121}
 2019-12-20 12:24:46 +00:00
Johannes Kron
4e64e60589 Reland "Reland "Distinguish between send and receive video codecs"" 
This is a reland of 77eb338ae48acb0cb1437da05d86941bb4063228

Original change's description:
> Reland "Distinguish between send and receive video codecs"
>
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
>
> Reason for revert: Downstream test updated.
>
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> >
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> >
> > Reason for revert: Breaks another downstream test.
> >
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > >
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > >
> > > Reason for revert: Downstream tests have been updated.
> > >
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > >
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > >
> > > > Reason for revert: Breaks downstream test.
> > > >
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > >
> > > > > Even though send and receive codecs are 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: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > >
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > >
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > >
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > >
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > >
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> >
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> >
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30079}
>
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
>
> # Not skipping CQ checks because original CL landed > 1 day ago.
>
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30097}

Bug: chromium:1029737
Change-Id: I5912822df8169fbb3097c0f440f7924527fa950b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162483
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30120}
 2019-12-20 11:44:42 +00:00
Yves Gerey
23df143850 Roll chromium_revision 26cf7e7d6c..e4c6d7fe53 (725941:726742) 
Manual tweak:  Don't roll src/third_party (harfbuzz-ng/BUILD.gn broken in WebRTC).

Change log: 26cf7e7d6c..e4c6d7fe53
Full diff: 26cf7e7d6c..e4c6d7fe53

Changed dependencies
* src/base: 6109a80975..f7fb459610
* src/build: 4abd203d72..032b1bd069
* src/ios: 4bc3bea248..d292120bff
* src/testing: 873e02ab19..c5705ae7a3
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d04ef219dd..8c77d963b8
* src/third_party/depot_tools: 6037820448..05934953bf
* src/tools: 3c85ad5a73..78adf6b3a0
DEPS diff: 26cf7e7d6c..e4c6d7fe53/DEPS

No update to Clang.

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

Change-Id: Iaf9e165f0447555b3b811b8cb1df682fc93664e6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162925
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30119}
 2019-12-20 11:23:39 +00:00
Jiwon Kim
077ee35774 Remove unused parameter in RtpFragmentize 
Bug: None
Change-Id: Ic110e3561bc93cb2156240193bc2077e2646ed87
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161560
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30118}
 2019-12-20 11:22:33 +00:00
CZ Theng
41875aa686 add rotationOverride for RTCEAGLVideoView 
Bug: webrtc:11221
Change-Id: I105b93de21fd2faeaf072c947c08006857c7a654
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162460
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30117}
 2019-12-20 10:57:33 +00:00
Per Åhgren
2e8e1c699e Open up for do the noise suppressor analysis on the linear AEC output 
This CL allows the noise suppressor to use the linear AEC output
for analysis whenever that is available. This will potentially
lower the risk that the noise suppressor estimates the noise
based on echo.
The feature is off by default.

Bug: webrtc:5298,b:132164318
Change-Id: Idc6c8e197d96209d213819d87a8fb2533b7303ec
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162900
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30116}
 2019-12-20 09:28:01 +00:00
Per Åhgren
9136abb45a AEC3: Ensure that the data size in the reverb computer is not fixed 
This CL ensures that the no data vectors in the reverb computer code
are fixed. This allows arbitrary long filters to be used, and ensures
that a minimum required heap size is used.

Bug: webrtc:8671
Change-Id: I7085ed262a3f5965d796270434b6578f4030606e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162661
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30115}
 2019-12-19 16:35:56 +00:00
Johannes Kron
c8f3134b29 Parse max-fr and max-fs from SDP FMTP line 
max-fr and max-fs are mandatory fields for VP8 and VP9.

Add parsing as a first step to enable use of these fields.

Bug: chromium:1032518
Change-Id: I4fd8f7f84f6303d646fb3f5313a02d6cf4160346
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162801
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30114}
 2019-12-19 15:00:41 +00:00
Rasmus Brandt
5cad55b240 Signal requested resolution alignment requirements from sinks to sources. 
Bug: webrtc:11218
Change-Id: I593b0515ea389bece472234a3c4082ccc5321ea5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162400
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30113}
 2019-12-19 10:39:04 +00:00
Per Åhgren
c04242548c Make the high-pass filter operate in full-band 
This CL moves the high-pass filter to run in the full-band domain
instead of the split-band domain.

Bug: webrtc:11193
Change-Id: Ie61f4a80afda11236ecbb1ad544bbd0350c7bbfd
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161453
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30112}
 2019-12-18 16:01:24 +00:00
Yves Gerey
26335a94de Roll chromium_revision 98124fd660..26cf7e7d6c (725465:725941) 
Change log: 98124fd660..26cf7e7d6c
Full diff: 98124fd660..26cf7e7d6c

Changed dependencies
* src/base: 9238aaece0..6109a80975
* src/build: 2da4a4ab32..4abd203d72
* src/ios: 11ba078b59..4bc3bea248
* src/testing: b1a11372f7..873e02ab19
* src/third_party: 64f463c5bb..b0673e3c20
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/ba22253ee9..d04ef219dd
* src/third_party/depot_tools: ba4699fef5..6037820448
* src/tools: 4a950d6680..3c85ad5a73
DEPS diff: 98124fd660..26cf7e7d6c/DEPS

No update to Clang.

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

Change-Id: I44e028ff47f369604659b6105eace1695659d48d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162561
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Yves Gerey <yvesg@google.com>
Cr-Commit-Position: refs/heads/master@{#30111}
 2019-12-18 15:46:24 +00:00
Rasmus Brandt
7ab41e59ba Fix typo in abseil-in-webrtc.md. 
NOTRY=True  # ios bots failing

Bug: None
Change-Id: Id3f3ea98be9cfef22c3431ae30cc15e282423ee2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162521
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30110}
 2019-12-18 14:27:34 +00:00
Erik Språng
ae10029bff Prevents probing while paused. 
The pacing controller allowed sending bitrate probes, despite it being
paused. This CL adresses that, and makes sure the task-queue based mode
also properly repsects pausing.

Bug: webrtc:10809
Change-Id: I79643c9a24666110d7583fce3ed1bfd6865e9e10
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162520
Reviewed-by: Florent Castelli <orphis@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30109}
 2019-12-17 17:57:38 +00:00
Yves Gerey
768c5f438c Roll chromium_revision faed30b47a..98124fd660 (724977:725465) 
Manual tweak: don't roll src/ios and src/testing

Change log: faed30b47a..98124fd660
Full diff: faed30b47a..98124fd660

Changed dependencies
* src/base: 1a89c23360..9238aaece0
* src/build: 03d0c36c52..2da4a4ab32
* src/third_party: 23379e2aee..64f463c5bb
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/243b5cc9e3..cb3f04f584
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/69337c37de..ba22253ee9
* src/third_party/freetype/src: 0c14a3adb0..7e1b39f6cd
* src/third_party/libjpeg_turbo: bc13578529..ce0e57e8e6
* src/tools: 87947b9472..4a950d6680
DEPS diff: faed30b47a..98124fd660/DEPS

No update to Clang.

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

Change-Id: Ic6402faa70bf0ee9ad957aaa33bf7331604535e2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162393
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30108}
 2019-12-17 15:06:03 +00:00
Florent Castelli
6fd58b3388 Add maxFramerate support to SimulcastEncoderAdapter 
Bug: webrtc:11117
Change-Id: I134964e669804e1a3c5acb9b9c7ddb6c911cf610
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162203
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30107}
 2019-12-17 14:08:47 +00:00
Ilya Nikolaevskiy
9b540cb553 Correctly process disabled streams in FrameEncodeMetadataWriter 
If the first simulcast stream is disabled, but the second one is enabled,
FrameEncodeMetadataWriter would fail to store frame metadata for all
streams and later fail to restore it for encoded frames.

Bug: none
Change-Id: Ib0d257abb863716ea94e56730f7caabef6ebeb64
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162480
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30106}
 2019-12-17 12:28:04 +00:00
Ilya Nikolaevskiy
00a1bcb441 Ensure that unset capture timestamp wouldn't cause incorrect SR rtp timestamps 
If for some reason capture timestamp is unset, the default value of 0 would be
passed to RtcpSender. This will cause rtp timestamps to grow at double the rate
in Sender Reports because it has time since the last frame capture as a term.

Bug: none
Change-Id: I2fe09dabef6b0957fb504deaa06393dedc4a9e70
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162481
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30105}
 2019-12-17 12:03:24 +00:00
Sebastian Jansson
f4cf4c789a Don't allow creation of sockets for wild card IPs in emulated networks. 
The network emulation framework does not support creation sockets that
receive from all addresses (e.g. 0.0.0.0) but would instead crash at
runtime. This CL explicitly ensures that we don't provide such networks.

Bug: webrtc:9883
Change-Id: I1d77df0f2c68f878eace30e4b037ebc7eb9f1aa6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162482
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30104}
 2019-12-17 10:16:59 +00:00
Björn Terelius
3a8df884d1 Add field trial to avoid extra backoffs in AIMD rate control. 
Bug: None
Change-Id: Iaa7dd0ffd6cfabb933e8e68a002b5432d13b9aab
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161946
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30103}
 2019-12-16 18:01:20 +00:00
Danil Chapovalov
32fe4ef967 Move vp9 rtp depacketization to VideoRtpDepacketizerVp9 
Bug: webrtc:11152
Change-Id: I560d4cd62fabae093e3df592f0e7cc4001c10657
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162420
Reviewed-by: Markus Handell <handellm@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30102}
 2019-12-16 17:11:13 +00:00
Steve Anton
094396fb76 Use a fake clock for rtc::Thread::PostDelayedTask test 
The test would flake using a real clock since time may pass between
calls to PostDelayedTask which would result in the tasks running
out of the expected order.

Bug: webrtc:11208, webrtc:11219
Change-Id: Ice5fe6ec4e9bf2ce89f00c6de7ed06b89dbe88cc
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162100
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30101}
 2019-12-16 15:46:48 +00:00
Ilya Nikolaevskiy
f9d92ed2c8 Revert "Reland "Distinguish between send and receive video codecs"" 
This reverts commit 77eb338ae48acb0cb1437da05d86941bb4063228.

Reason for revert: Speculative revert, as it seems to have broken webrtc-importer

Original change's description:
> Reland "Distinguish between send and receive video codecs"
> 
> This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.
> 
> Reason for revert: Downstream test updated.
> 
> Original change's description:
> > Revert "Reland "Distinguish between send and receive video codecs""
> > 
> > This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> > 
> > Reason for revert: Breaks another downstream test.
> > 
> > Original change's description:
> > > Reland "Distinguish between send and receive video codecs"
> > > 
> > > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > > 
> > > Reason for revert: Downstream tests have been updated.
> > > 
> > > Original change's description:
> > > > Revert "Distinguish between send and receive video codecs"
> > > > 
> > > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > > 
> > > > Reason for revert: Breaks downstream test.
> > > > 
> > > > Original change's description:
> > > > > Distinguish between send and receive video codecs
> > > > > 
> > > > > Even though send and receive codecs are 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: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > > 
> > > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > > 
> > > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > > No-Presubmit: true
> > > > No-Tree-Checks: true
> > > > No-Try: true
> > > > Bug: chromium:1029737
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30042}
> > > 
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > 
> > > # Not skipping CQ checks because original CL landed > 1 day ago.
> > > 
> > > Bug: chromium:1029737
> > > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30078}
> > 
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > 
> > Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: chromium:1029737
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30079}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> # Not skipping CQ checks because original CL landed > 1 day ago.
> 
> Bug: chromium:1029737
> Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30097}

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

Change-Id: I73d4fe3bb18e40a01f1b1b0c71f9dc7b85c513b7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:1029737
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162208
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30100}
 2019-12-16 15:28:41 +00:00
Harald Alvestrand
2697ac1a1b Stop an SCTP connection when the DTLS transport closes. 
This CL propagates a "closed" signal from DTLS up to the
SCTP section of the data channel controller, where it causes
closing of all open datachannels.

Bug: chromium:1030631, webrtc:10360
Change-Id: I88bb9e1aff5c25f330edfd092ef609d4fcc3a9f8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162206
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30099}
 2019-12-16 14:54:56 +00:00
Artem Titov
8525a8028a Add ability to resize buffers pool in decoder and use it in IVF generator 
Bug: webrtc:10138
Change-Id: I452f08f1d9af57de789bd947a1fcb95536845f80
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162183
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30098}
 2019-12-16 14:51:16 +00:00
Johannes Kron
77eb338ae4 Reland "Distinguish between send and receive video codecs" 
This reverts commit f2d6fe62f23f13b974d50baa9ef60426a242af03.

Reason for revert: Downstream test updated.

Original change's description:
> Revert "Reland "Distinguish between send and receive video codecs""
> 
> This reverts commit 26e6afe93f134c844d739384784e78acc07cc145.
> 
> Reason for revert: Breaks another downstream test.
> 
> Original change's description:
> > Reland "Distinguish between send and receive video codecs"
> > 
> > This reverts commit f22af3cca7cfe517e4126db4b7083475722c3e6d.
> > 
> > Reason for revert: Downstream tests have been updated.
> > 
> > Original change's description:
> > > Revert "Distinguish between send and receive video codecs"
> > > 
> > > This reverts commit 18314bd8d2cb27fa58e4d304bbc428e3ed1736ba.
> > > 
> > > Reason for revert: Breaks downstream test.
> > > 
> > > Original change's description:
> > > > Distinguish between send and receive video codecs
> > > > 
> > > > Even though send and receive codecs are 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: I16a80da44c5061ca42f2aabda76e6bf0b879bf7b
> > > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161306
> > > > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > > > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > > Cr-Commit-Position: refs/heads/master@{#30041}
> > > 
> > > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > > 
> > > Change-Id: I7e5807460006db613e9b3b369ec6036b88f164fd
> > > No-Presubmit: true
> > > No-Tree-Checks: true
> > > No-Try: true
> > > Bug: chromium:1029737
> > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161662
> > > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#30042}
> > 
> > TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> > 
> > # Not skipping CQ checks because original CL landed > 1 day ago.
> > 
> > Bug: chromium:1029737
> > Change-Id: Ia70b11376b43888e2495ef21838c2d2e3c68d735
> > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161734
> > Reviewed-by: Johannes Kron <kron@webrtc.org>
> > Reviewed-by: Steve Anton <steveanton@webrtc.org>
> > Reviewed-by: Anders Carlsson <andersc@webrtc.org>
> > Commit-Queue: Johannes Kron <kron@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#30078}
> 
> TBR=steveanton@webrtc.org,andersc@webrtc.org,kron@webrtc.org
> 
> Change-Id: Ia4971b898c9209a3736a916a1c2c48d392dfdad6
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: chromium:1029737
> Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162140
> Reviewed-by: Johannes Kron <kron@webrtc.org>
> Commit-Queue: Johannes Kron <kron@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#30079}

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

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

Bug: chromium:1029737
Change-Id: If2c3c5b5e7d86cb852a1f20f02b6ceae62b2e0c8
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162186
Reviewed-by: Johannes Kron <kron@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Johannes Kron <kron@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30097}
 2019-12-16 14:03:46 +00:00
Raman Budny
5331079132 Protect against assigning current_offset_ negative value. 
Bug: webrtc:11176
Change-Id: Ic3937da6f1ee9cd118372693cb71d70beb43159c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161329
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30096}
 2019-12-16 13:06:52 +00:00
Danil Chapovalov
cebdbf650d switch RtpVideoStreamReceiver to use VideoRtpDepacketizer interface 
instead of creating each time an object with RtpDepacketizer interface

this moves packet payload memcpy from RtpVideoStreamReceiver into
the depacketizers with possibility to remove it from there in follow ups.

Bug: webrtc:11152
Change-Id: If474207eb84d7e9d0207075bd395e60895f0d842
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162185
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30095}
 2019-12-16 12:08:11 +00:00
Danil Chapovalov
0f6bf75ab4 Make video engine tests aware of padding packets 
Specifically do not try to parse them as rtx packets.

Bug: webrtc:11213, webrtc:11188
Change-Id: I3aa5929af433b1ada9fb26516618d11207f075a5
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162204
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Yves Gerey <yvesg@google.com>
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30094}
 2019-12-16 11:43:11 +00:00
Patrik Höglund
73eb784676 Don't crash the test process when X11 isn't available. 
It's not great to use asserts in util functions like this because it
breaks the arrange-act-assert rule, but using checks is worse because
they will crash the test process on failure (= no other tests get run
after that).

Bug: b/143587130
Change-Id: If4d085311de0792b9fca1584db299fd24199e72e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162360
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30093}
 2019-12-16 09:57:59 +00:00
Yves Gerey
774fb933a3 Roll chromium_revision cd7700164d..faed30b47a (724740:724977) 
Manual tweak: Don't roll src/ios and src/testing.

Change log: cd7700164d..faed30b47a
Full diff: cd7700164d..faed30b47a

Changed dependencies
* src/base: b3e5fa8e95..1a89c23360
* src/build: ddbcd10b58..03d0c36c52
* src/third_party: 76b48eb36f..23379e2aee
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/67b9c374e8..69337c37de
* src/third_party/depot_tools: f1ad6e46ed..ba4699fef5
* src/third_party/freetype/src: 11d4ce23ac..0c14a3adb0
* src/tools: d811596acb..87947b9472
DEPS diff: cd7700164d..faed30b47a/DEPS

No update to Clang.

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

Change-Id: Id53027ecd5c582cd3efbef8d5f3483304740e6c0
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162340
Commit-Queue: Yves Gerey <yvesg@google.com>
Reviewed-by: Yves Gerey <yvesg@webrtc.org>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#30092}
 2019-12-15 22:25:37 +00:00
Harald Alvestrand
17ea068e8c Integration test that verifies that data channels open. 
This is in preparation for writing tests that verify that
they close, and that they close at the right times.

Bug: chromium:1030631, webrtc:10360
Change-Id: I8129a9fc9731c1bfe1a660e82e23c1aeff1e5087
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/162181
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30091}
 2019-12-13 23:03:34 +00:00