Commit Graph

25039 Commits

Author SHA1 Message Date
cd2e105128 Reenable test RampUpTest.AudioTransportSequenceNumber
Flakiness should be fixed with cl
https://webrtc-review.googlesource.com/96900

Bug: webrtc:8878
Change-Id: I536d670fdf3b9e52091931e2f37ff9b8d02c2f77
Reviewed-on: https://webrtc-review.googlesource.com/c/110160
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25557}
2018-11-08 12:28:19 +00:00
694ed1793c Add a style rule about not using const optional<T>& arguments
Motivated by discussions here:
https://webrtc-review.googlesource.com/c/src/+/109583

Bug: none
Change-Id: Ia0723adf9fa7c970137ffc9cb5612cb3360d7f5f
Notry: true
Reviewed-on: https://webrtc-review.googlesource.com/c/109568
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25556}
2018-11-08 11:57:35 +00:00
f0e7440a35 Add missing conditional defines to neteq test and tools targets
The .cc source files listed below #ifdef for WEBRTC_CODEC_OPUS and
WEBRTC_CODEC_ILBC but the build files don't include the defines.

modules/audio_coding/neteq/tools/neteq_test.cc
modules/audio_coding/neteq/tools/neteq_test_factory.cc

Bug: None
Change-Id: I6065021f68e58d0e5663acd006a9865bf265adc0
Reviewed-on: https://webrtc-review.googlesource.com/c/109925
Reviewed-by: Minyue Li <minyue@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25555}
2018-11-08 11:25:10 +00:00
689983f6bc Deprecate EventFactory and delete all usage.
Will be deleted as soon as downstream calls of
VideoCodingModule::Create are updated.

Tbr: sprang@webrtc.org  # Trivial change in video/
Bug: webrtc:3380
Change-Id: Iaeb6da2fb68991225fe9086ddddd4a553e1620b4
Reviewed-on: https://webrtc-review.googlesource.com/c/107890
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25554}
2018-11-08 11:00:37 +00:00
54b4924349 Update H264 encoder to use GetEncoderInfo
Bug: webrtc:9890
Change-Id: I952b979346d97c42a4f60e9e2b091da563dfffab
Reviewed-on: https://webrtc-review.googlesource.com/c/109921
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25553}
2018-11-07 17:01:50 +00:00
10608708eb Update LibVpxVp8Encoder to use GetEncoderInfo
Bug: webrtc:9890
Change-Id: I76566bc38137c81b029fa848da89c96454260895
Reviewed-on: https://webrtc-review.googlesource.com/c/109920
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25552}
2018-11-07 17:00:10 +00:00
727d1649c6 Update VP9 encoder to use GetEncoderInfo
Bug: webrtc:9890
Change-Id: I74c1e098c800a44e2e038cd8a01be6c61bec97f5
Reviewed-on: https://webrtc-review.googlesource.com/c/109922
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25551}
2018-11-07 16:51:40 +00:00
5473a45688 Remove multiple RTX codec entries in GetRtpReceiver/SenderCapabilities
The spec says there should only be a single entry with no parameters.

Bug: webrtc:9970
Change-Id: I8b55f10b8cb795021269827c6e0e9f12ab86a3c9
Reviewed-on: https://webrtc-review.googlesource.com/c/109588
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25550}
2018-11-07 15:35:56 +00:00
75de46a966 Update SimulcastEncoderAdapter merging of EncoderInfo
Determining the EncoderInfo meta data is now done during InitEncode().

This implementation assums that no dynamic wrappers are wrapped in this
simulcast encoder adapter. Ie, if supports_native_handle changes,
InitEncode() must be called again for it to be reported properly.

Bug: webrtc:9722
Change-Id: I7901effe11e89ac011659a4ea862ab2a42577eb5
Reviewed-on: https://webrtc-review.googlesource.com/c/109620
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25549}
2018-11-07 15:22:38 +00:00
e6a2d94eca Clear FrameBuffer if there were no frames received for 10 minutes
This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.

Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}
2018-11-07 15:09:11 +00:00
b768e8800f Reland "Isolating APM API build target: making :api an actual target."
This reverts commit 61c6e5643e7ea058e653956980a90e033249c055.

Reason for revert: downstream projects prepared for this change

Original change's description:
> Revert "Isolating APM API build target: making :api an actual target."
> 
> This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.
> 
> Reason for revert: breaking downstream
> 
> Original change's description:
> > Isolating APM API build target: making :api an actual target.
> > 
> > This CL is part of a refactoring work to unblock other CLs
> > that would generate a circular dependency when including
> > modules/audio_processing. It will also allow to easily move
> > the APM interface part under //api.
> > 
> > More in detail, this change moves the APM interface files from
> > the build target modules/audio_processing to
> > modules/audio_processing:api. It also adds :api as dependency
> > where needed.
> > 
> > Bug: webrtc:9535
> > Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> > Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> > Reviewed-by: Niels Moller <nisse@webrtc.org>
> > Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> > Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#25539}
> 
> TBR=saza@webrtc.org,alessiob@webrtc.org,kwiberg@webrtc.org,nisse@webrtc.org,kthelgason@webrtc.org
> 
> Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:9535
> Reviewed-on: https://webrtc-review.googlesource.com/c/109820
> Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25540}

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

Change-Id: Ic8ed4cc3baf43d639ce13cae256c007728c3ad92
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109884
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25547}
2018-11-07 14:30:06 +00:00
bdc6c40213 Add field trial for target bitrate RTCP XR message.
Bug: webrtc:9969
Change-Id: I71ce59556f43b0c0547d095793c59ab721cf1daf
Reviewed-on: https://webrtc-review.googlesource.com/c/109566
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25546}
2018-11-07 14:24:12 +00:00
d5659188a0 Delete NullEventFactory
A preparation for deleting EventFactory and EventWrapper, to instead
use rtc::Event directly.

Bug: webrtc:3380
Change-Id: I4c40daca9268e57b06d506d91e09365091c42ad6
Reviewed-on: https://webrtc-review.googlesource.com/c/109880
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25545}
2018-11-07 14:19:52 +00:00
e769ed90c3 Roll chromium_revision 38dcb5ed01..db720b4ab9 (605924:606025)
Change log: 38dcb5ed01..db720b4ab9
Full diff: 38dcb5ed01..db720b4ab9

Changed dependencies
* src/base: 546dc6eba1..fee916f36b
* src/ios: 25df0733e7..95aadfb43f
* src/testing: 7e5b9bf06b..03b25bebb5
* src/third_party: 1dd44c1c64..60e74a707b
* src/third_party/depot_tools: 6d6e0a8009..4d2d5b4bbe
* src/third_party/libFuzzer/src: a305a5eb85..2a53098584
* src/tools: dc7f4d2a7b..a8e76f0ca5
DEPS diff: 38dcb5ed01..db720b4ab9/DEPS

No update to Clang.

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

Change-Id: Id66e1290a2f09f819400864e13222324f2aa3e3c
Reviewed-on: https://webrtc-review.googlesource.com/c/109841
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@{#25544}
2018-11-07 13:33:24 +00:00
50f60cb4b3 Rename software codec classes and move them into api/
We want clients to be able to build their own factories around these
codecs.

Bug: webrtc:7925
Change-Id: Ia8f62d5d85e63ac6e3eb402c5996d8b986625615
Reviewed-on: https://webrtc-review.googlesource.com/c/109529
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Jonathan Yu <yujo@chromium.org>
Cr-Commit-Position: refs/heads/master@{#25543}
2018-11-07 12:24:14 +00:00
ff7020af27 Remove non-default VideoEncoder::EncoderInfo() ctor
This removes the constructor that takes all members as parameters.
A main goal with this struct is to make it easy to add or remove fields
without having to refactor implementations and usage all over the palce,
this breaks down if we need to update the arguments in this constructor
every time.

Also cleans up a few commented-out lines that were forgotten in an
earliers cl.

Bug: webrtc:9722
Change-Id: I22db35d399469bdb3bfd8bdc531b24ead9fe1c3e
Reviewed-on: https://webrtc-review.googlesource.com/c/109570
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25542}
2018-11-07 12:20:59 +00:00
36d907b6cd Update MockVideoEncoder with correct methods.
Add GetEncoderInfo() and remove HasTrustedRateController(), which was
erroneously added here:
https://webrtc-review.googlesource.com/c/src/+/105620

Bug: webrtc:9722
Change-Id: Iaff6eed773c3431b806adb694b6e3564b180188e
Reviewed-on: https://webrtc-review.googlesource.com/c/109586
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25541}
2018-11-07 12:19:54 +00:00
61c6e5643e Revert "Isolating APM API build target: making :api an actual target."
This reverts commit a7f77a7c05b5d26520fd01a773ffb2c8b15b60ff.

Reason for revert: breaking downstream

Original change's description:
> Isolating APM API build target: making :api an actual target.
> 
> This CL is part of a refactoring work to unblock other CLs
> that would generate a circular dependency when including
> modules/audio_processing. It will also allow to easily move
> the APM interface part under //api.
> 
> More in detail, this change moves the APM interface files from
> the build target modules/audio_processing to
> modules/audio_processing:api. It also adds :api as dependency
> where needed.
> 
> Bug: webrtc:9535
> Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
> Reviewed-on: https://webrtc-review.googlesource.com/c/109501
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
> Reviewed-by: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Sam Zackrisson <saza@webrtc.org>
> Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25539}

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

Change-Id: I974c6237311e7c06970aa62e5f6940f3aa80113d
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9535
Reviewed-on: https://webrtc-review.googlesource.com/c/109820
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25540}
2018-11-07 11:28:03 +00:00
a7f77a7c05 Isolating APM API build target: making :api an actual target.
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.

More in detail, this change moves the APM interface files from
the build target modules/audio_processing to
modules/audio_processing:api. It also adds :api as dependency
where needed.

Bug: webrtc:9535
Change-Id: I72829e22d08ba4d75985f0421e6e8bf0216ebecd
Reviewed-on: https://webrtc-review.googlesource.com/c/109501
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25539}
2018-11-07 10:34:51 +00:00
7553c02b1e Update ObjCVideoEncoder to use GetEncoderInfo()
This method replaces GetScalingSettings(), GetImpementationName() and
SupportsNativeHandle().

Bug: webrtc:9890
Change-Id: I8a4b13414f66c41f6697ed84854424ab2d8e18e4
Reviewed-on: https://webrtc-review.googlesource.com/c/109460
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25538}
2018-11-07 10:00:19 +00:00
7b3c76b44f Reland "Delete rtc::Pathname"
This is a reland of 6b9dec0d16f2df59fa2820c5ec1341be52fb9f32

Original change's description:
> Delete rtc::Pathname
> 
> Bug: webrtc:6424
> Change-Id: Iec01dc5dd1426d4558983b828b67af872107d723
> Reviewed-on: https://webrtc-review.googlesource.com/c/108400
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25479}

Bug: webrtc:6424
Change-Id: Ic7b42d435ffd8b93f603acebe68e8a92366bb197
Reviewed-on: https://webrtc-review.googlesource.com/c/109561
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25537}
2018-11-07 09:57:55 +00:00
17fc7e2320 Add counter to the end of FakeEncoder frames in order to make them unique.
Bug: none
Change-Id: Ia5dccbe1868cc3fda5693d7e13eac21f75a08b83
Reviewed-on: https://webrtc-review.googlesource.com/c/109582
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25536}
2018-11-07 09:08:35 +00:00
c572ff3c71 Add default constructor for rtc::Event
Bug: webrtc:9962
Change-Id: Icaa91e657e6881fcb1553f354c07866109a0ea68
Reviewed-on: https://webrtc-review.googlesource.com/c/109500
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25535}
2018-11-07 08:57:50 +00:00
3ea7b83fa3 Resolve the race condition between mDNS name registration and
cricket::Port::SignalPortComplete.

The mDNS name registration is asynchronously executed by the mDNS
responder, and a host candidate with an mDNS name is only gathered after
this completes. SignalPortComplete however is currently done
synchronously by UDPPort, and any candidate gathered by a UDPPort after
this signal is fired would be discarded.

Bug: webrtc:9964, webrtc:9605
Change-Id: If8aaf193ef26c06bd118e6418b62ba0de5e87e3c
Reviewed-on: https://webrtc-review.googlesource.com/c/109541
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Zach Stein <zstein@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25534}
2018-11-07 02:59:41 +00:00
8770ce7074 Roll chromium_revision 03cf97f6d8..38dcb5ed01 (605818:605924)
Change log: 03cf97f6d8..38dcb5ed01
Full diff: 03cf97f6d8..38dcb5ed01

Changed dependencies
* src/base: ad50d06b11..546dc6eba1
* src/build: 41398b47eb..9f8abf9183
* src/ios: 53572a08cb..25df0733e7
* src/testing: 4cd36e3e50..7e5b9bf06b
* src/third_party: 02af94e28a..1dd44c1c64
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/0273c5e3c2..2e722b007d
* src/third_party/libvpx/source/libvpx: fa0076282e..7808cc796e
* src/tools: 41d11031b4..dc7f4d2a7b
DEPS diff: 03cf97f6d8..38dcb5ed01/DEPS

No update to Clang.

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

Change-Id: Iec560e1c1bf8b7866990283ea3e8541827a3da66
Reviewed-on: https://webrtc-review.googlesource.com/c/109760
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25533}
2018-11-07 02:29:35 +00:00
bb091db398 Roll chromium_revision 793c8566ab..03cf97f6d8 (605715:605818)
Change log: 793c8566ab..03cf97f6d8
Full diff: 793c8566ab..03cf97f6d8

Changed dependencies
* src/base: 2cc4eaccb9..ad50d06b11
* src/build: b02fb51ba9..41398b47eb
* src/ios: 39b74f70b6..53572a08cb
* src/testing: 93d10ee9f8..4cd36e3e50
* src/third_party: 7fe6380b56..02af94e28a
* src/third_party/android_deps/libs/com_google_android_play_core: version:1.3.0-cr0..version:1.3.5-cr0
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6ba6af1e54..0273c5e3c2
* src/tools: 198e0eceb8..41d11031b4
DEPS diff: 793c8566ab..03cf97f6d8/DEPS

No update to Clang.

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

Change-Id: Ib5f5792f65f4232ea91e2a78b21e2013e8b4f1df
Reviewed-on: https://webrtc-review.googlesource.com/c/109642
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25532}
2018-11-06 21:31:46 +00:00
2cd3b4c819 Fixing bug in SimulatedNetwork where packets stop.
Bug: webrtc:9952
Change-Id: I68491f1d18fee317165999453776a35cea41e71f
Reviewed-on: https://webrtc-review.googlesource.com/c/109009
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25531}
2018-11-06 19:04:37 +00:00
0f54f211e6 Removes deprecated GetSentPacket from PacketResult.
Bug: webrtc:9934
Change-Id: If4dc15a3122cbcafc87a93042bc46e49276860b6
Reviewed-on: https://webrtc-review.googlesource.com/c/109023
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25530}
2018-11-06 19:03:13 +00:00
dc98b9b975 AEC3: Corrected include
Bug: webrtc:8671
Change-Id: I3267c4d48cb52cc7bf305ecd7ec3f3a6222276be
Reviewed-on: https://webrtc-review.googlesource.com/c/109569
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25529}
2018-11-06 18:57:19 +00:00
c564a7b522 Roll chromium_revision 7841106b37..793c8566ab (605607:605715)
Change log: 7841106b37..793c8566ab
Full diff: 7841106b37..793c8566ab

Changed dependencies
* src/base: 068a8bf504..2cc4eaccb9
* src/build: db0889426f..b02fb51ba9
* src/ios: 293ed1e431..39b74f70b6
* src/testing: 12f7c4a1e6..93d10ee9f8
* src/third_party: 3aef1b31a6..7fe6380b56
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/9a5dbc6f44..6ba6af1e54
* src/third_party/depot_tools: 459aaba034..6d6e0a8009
* src/third_party/icu: 42d5027992..834113aab5
* src/tools: ee42646d87..198e0eceb8
DEPS diff: 7841106b37..793c8566ab/DEPS

No update to Clang.

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

Change-Id: Iaa737b9c3e0bc8e64bb2ee436a351b6eebe4e541
Reviewed-on: https://webrtc-review.googlesource.com/c/109600
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25528}
2018-11-06 17:33:57 +00:00
8ffd71026f Update Android encoder to use GetEncoderInfo()
This method replaces GetScalingSettings(), SupportsNativeHandle() and
GetImplementationName().

Bug: webrtc:9890
Change-Id: I755cd4c6b1f04853a35f1185a84bda7c8c8efb62
Reviewed-on: https://webrtc-review.googlesource.com/c/109440
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25527}
2018-11-06 17:30:48 +00:00
020e583291 AEC3: Compensate comfort noise level for loss due to filter bank
The analysis and synthesis windowing cause loss of power when
cross-fading the noise where frames are completely uncorrelated
(generated with random phase).

This CL also removes duplicate code and enables platform specific
optimizations for ARM in the comfort noise generation.

Bug: webrtc:9967,chromium:902262
Change-Id: Iffd59b301876442079d4a5f2c7fac55a3522397c
Reviewed-on: https://webrtc-review.googlesource.com/c/109581
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25526}
2018-11-06 16:17:02 +00:00
83b00f020e AEC3: Computation of comfort noise gains from suppression gains
This change corrects the computation of the comfort noise gains.

Previously the comfort noise gain of band k, CG_k, was computed
from suppression gain of band k, SG_k, as:
CG_k = 1 - SG_k

But since the two signals are uncorrelated (the comfort noise
is randomly generated), the correct gain to maintain power is:
CG_k = sqrt(1 - SG_k^2).

Bug: webrtc:9967,chromium:902262
Change-Id: I393495742163d5e658bca4ab2f7a5067ab15af01
Reviewed-on: https://webrtc-review.googlesource.com/c/109580
Commit-Queue: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25525}
2018-11-06 16:10:52 +00:00
34fc346a0c Add support for computing iOS code coverage
Also disable failing PosixSignalDeliveryTest* tests for iOS

Bug: chromium:844647
Change-Id: I64bb233bef2f06f6778f2d475b6d3ad685fb9143
Reviewed-on: https://webrtc-review.googlesource.com/c/105641
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Peter Hanspers <peterhanspers@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25524}
2018-11-06 15:18:51 +00:00
277b6ea850 Isolating APM API build target: adding dummy :api target.
This CL is part of a refactoring work to unblock other CLs
that would generate a circular dependency when including
modules/audio_processing. It will also allow to easily move
the APM interface part under //api.

More in detail, this change adds a dummy build target named :api
in modules/audio_processing. It is needed to adapt the downstream
projects before the APM interface files are moved to the :api target.

A follow up CL will make :api an actual target and will remove
the interface files from :audio_processing.

Bug: webrtc:9535
Change-Id: Ifb4e1a0ac7e482a8a089ef858d7e9a91f974e51f
Reviewed-on: https://webrtc-review.googlesource.com/c/109585
Reviewed-by: Sam Zackrisson <saza@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25523}
2018-11-06 14:44:31 +00:00
3ddaf3c408 Revert "Add support for screen sharing with PipeWire on Wayland"
This reverts commit dd20c9c1e3f681f6c33d1879c76f588bd4b095bd.

Reason for revert: Speculative revert; looks like it causes crashes on official builders. See crbug.com/901319.

Original change's description:
> Add support for screen sharing with PipeWire on Wayland
>
> Currently, when users want to use the screen sharing and are using the
> Wayland display server (the default on Fedora distribution), then it
> doesn't work, because the WebRTC only includes the X11 implementation.
> This change adds the support by using the PipeWire multimedia server.
>
> The PipeWire implementation in WebRTC stays in
> screen-capturer-pipewire.c and is guarded by the rtc_use_pipewire build
> flag that is automatically enabled on Linux.
>
> More information are included in the relevant commit messages.
>
> Tested on the current Chromium master and Firefox.
>
> The sysroot changes are requested in:
> https://chromium-review.googlesource.com/c/chromium/src/+/1258174
>
> Co-authored-by: Jan Grulich <grulja@gmail.com>
> Co-authored-by: Eike Rathke <erathke@redhat.com>
> Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
>
> BUG=chromium:682122
>
> Change-Id: I212074a4bc437b99a77bf383266026c5bfae7c4a
> Reviewed-on: https://webrtc-review.googlesource.com/c/103504
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#25461}

TBR=phoglund@webrtc.org,jamiewalch@chromium.org,niklas.enbom@webrtc.org,braveyao@webrtc.org,tomas.popela@gmail.com

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

NOPRESUBMIT=true

Bug: chromium:682122, chromium:901319
Change-Id: I4ca5da77daea73cae1232953a0d633900a85a93d
Reviewed-on: https://webrtc-review.googlesource.com/c/109584
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25522}
2018-11-06 14:17:34 +00:00
82c07ea1da Tune huge video frames detection threshold for GetStats googHugeFramesSent stat
Bug: none
Change-Id: I4e29c8c0e4438fb4dace56409ae4503abb598f86
Reviewed-on: https://webrtc-review.googlesource.com/c/109567
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25521}
2018-11-06 13:54:24 +00:00
4f3cc6efbe Make VideoSendStreamTest.NoPaddingWhenVideoIsMuted less flaky
Make sure padding is sent by setting min transmit bitrate.
After camera has been restarted, stop camera again and make sure padding
is still produced. This avoids hoping for padding to be produced based
on assumptions of how padding allocation works with simulcast.

Bug: webrtc:9968
Change-Id: I7984abed4d04b35d3885f7edd62ca82f28bdcdb7
Reviewed-on: https://webrtc-review.googlesource.com/c/109564
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25520}
2018-11-06 13:34:59 +00:00
a8f54617c0 nit: Use make_unique in rtp_video_stream_receiver.cc
Bug: None
Change-Id: I5b844cbb5be94c3b7a1866b4f5ba09087701dd97
Reviewed-on: https://webrtc-review.googlesource.com/c/109563
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25519}
2018-11-06 12:27:34 +00:00
27f31727d0 Simplify use of events in TestAudioDevice
Create events with |manual_reset| and |initially_signalled| both false
(used to be both true). Delete calls to Set and Reset events from the
{Start,Stop}{Playout,Recording} methods. Then, for each event, there
remains a single call to Set, in the ProcessingAudio loop, and a
single call to Wait, in WaitForPlayoutEnd and WaitForRecordingEnd,
respectively.

Bug: webrtc:9962
Change-Id: Ia358b4a36896e2378ad6166f3786d8d71392bf1b
Reviewed-on: https://webrtc-review.googlesource.com/c/109562
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25518}
2018-11-06 12:18:05 +00:00
361dbc1973 Android: Add option to set presentation timestamp in EglRenderer
Bug: b/119004693
Change-Id: I78b676a4417ac313e7fbbea009c8dd586707b1af
Reviewed-on: https://webrtc-review.googlesource.com/c/109503
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25517}
2018-11-06 12:11:20 +00:00
967f7d5497 Add audio level to CSRC class
This patch adds (optional) csrc to ContributingSources.
This will be used if using virtual audio ssrc, since
the audio level is otherwise unaccessible in that configuration.

BUG=webrtc:3333

Change-Id: Ied263b8f0850553cd637fd6bead373ed4252fd1e
Reviewed-on: https://webrtc-review.googlesource.com/c/109281
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25516}
2018-11-06 12:10:05 +00:00
df351f4b54 Update FakeEncoder to use EncoderInfo
Bug: webrtc:9890
Change-Id: Ia6f268b35b5cc62fb6318f59bd66bbf65e97a23d
Reviewed-on: https://webrtc-review.googlesource.com/c/109565
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25515}
2018-11-06 12:01:47 +00:00
254d3db59a Add missing #include to absl/memory/memory.h from audio_encoder_cng.cc
absl::make_unique is used in this file without absl/memory/memory.h
#include, that causes a build error on C++17 build of Chromium.

Bug: chromium:752720
Change-Id: I78fe9f76a6ea670a4250b4cf25c3c02cf4c4beb6
Reviewed-on: https://webrtc-review.googlesource.com/c/109540
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25514}
2018-11-06 10:57:47 +00:00
fbf168386b Add HdrMetadata to VideoFrame
Bug: webrtc:8651
Change-Id: I28aacbc63e346328633fb862662343f47e966bf1
Reviewed-on: https://webrtc-review.googlesource.com/c/108320
Commit-Queue: Johannes Kron <kron@webrtc.org>
Reviewed-by: Emircan Uysaler <emircan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25513}
2018-11-06 09:29:54 +00:00
4f0f3d5a22 Remove unused member variable - RTCPSender::using_nack_
Bug: None
Change-Id: I346c6a195d2d21387f89e41cfc070f33466ef195
Reviewed-on: https://webrtc-review.googlesource.com/c/109462
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Elad Alon <eladalon@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25512}
2018-11-06 09:21:42 +00:00
63ada787b5 Remove outdated TODO
Bug: webrtc:9535
Change-Id: I8f7a719eb9f32a91f45620453568e5f7d2264de8
Reviewed-on: https://webrtc-review.googlesource.com/c/109461
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25511}
2018-11-06 08:39:16 +00:00
3ea187803b Add severity into RTC logging callbacks
Bug: webrtc:9945
Change-Id: I5022f63103503d2213492d3cd1a6953fe658fda7
Reviewed-on: https://webrtc-review.googlesource.com/c/108981
Commit-Queue: Jiawei Ou <ouj@fb.com>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25510}
2018-11-06 07:53:01 +00:00
edfb883c85 Roll chromium_revision 11d7305a72..7841106b37 (605505:605607)
Change log: 11d7305a72..7841106b37
Full diff: 11d7305a72..7841106b37

Changed dependencies
* src/build: c9ef69b989..db0889426f
* src/ios: f2f46d1dd4..293ed1e431
* src/testing: 8ac08c41f8..12f7c4a1e6
* src/third_party: faebc5969e..3aef1b31a6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/6b1f256782..9a5dbc6f44
* src/third_party/depot_tools: 7e0b0c498a..459aaba034
* src/tools: 5d0c86c1c7..ee42646d87
DEPS diff: 11d7305a72..7841106b37/DEPS

No update to Clang.

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

Change-Id: I66b792c58552a2775d1a89dfc14c4e71aa0e6dc2
Reviewed-on: https://webrtc-review.googlesource.com/c/109524
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#25509}
2018-11-06 06:20:09 +00:00
d7db17b992 Roll chromium_revision bf7ad46dee..11d7305a72 (605401:605505)
Change log: bf7ad46dee..11d7305a72
Full diff: bf7ad46dee..11d7305a72

Changed dependencies
* src/base: 6baa5ba029..068a8bf504
* src/build: b906840601..c9ef69b989
* src/ios: 435391a2bd..f2f46d1dd4
* src/testing: a266fda43c..8ac08c41f8
* src/third_party: 519b3029bb..faebc5969e
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/8c035b35ab..6b1f256782
* src/tools: 7b0ad7dabb..5d0c86c1c7
DEPS diff: bf7ad46dee..11d7305a72/DEPS

No update to Clang.

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

Change-Id: I96ef3feaf4abbcd82275435553766fba64fc5649
Reviewed-on: https://webrtc-review.googlesource.com/c/109486
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@{#25508}
2018-11-05 23:40:03 +00:00