Commit Graph

23581 Commits

Author SHA1 Message Date
075cb2b2f7 AEC3: Changes to how the reverberation decay is applied.
In this work we introduce some changes on how the reverberation model for AEC3 is applied. Currently, the exponential modelling of the tails is applied over the linear echo estimates. That might result  in an overestimation of the reverberation tails under certain conditions. In this work, the reverberation model is instead applied over an estimate of the energies at the tails of the linear estimate.

Additionally, the stationary estimator is changed so it does not disable the aec immediately after a burst of activity.

Bug: webrtc:9384,webrtc:9400,chromium:852257
Change-Id: Ia486694ed326cfe231fc688877c0b9b6e2c450ff
Reviewed-on: https://webrtc-review.googlesource.com/82161
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Jesus de Vicente Pena <devicentepena@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23599}
2018-06-13 14:54:04 +00:00
9633cff81a Remove "webrtc_rtp" traces.
They have been disabled by default for years, and should have been made redundant by the event logs.

Bug: webrtc:8982
Change-Id: I491923cbc93378d28f5166d24756b335619d9c12
Reviewed-on: https://webrtc-review.googlesource.com/82800
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23598}
2018-06-13 14:46:24 +00:00
29c36b24a8 Add ow2_asm license
Added empty license for build time dependency - ow2_asm library

Bug: webrtc:9393
Change-Id: I1d43ad986cbb50a26d0f5c88f119383de6f7309a
Reviewed-on: https://webrtc-review.googlesource.com/83166
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23597}
2018-06-13 14:43:23 +00:00
789221f110 Adding WebRTC-Audio-ForceNoTWCC field trial
Bug: webrtc:8243
Change-Id: I74864b8e67cd9c62c5fe26a03efdcdca01d2a93f
Reviewed-on: https://webrtc-review.googlesource.com/83323
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23596}
2018-06-13 12:30:59 +00:00
e3cf3d0496 Use enum class for VideoCodecMode and VideoCodecComplexity.
Bug: webrtc:7660
Change-Id: I6a8ef01f8abcc25c8efaf0af387408343a7c8ba3
Reviewed-on: https://webrtc-review.googlesource.com/81240
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23595}
2018-06-13 12:26:09 +00:00
037b37a192 Add implementation of EncodedFrame::Timestamp.
This is a preparation for inheriting the method from the base class,
and delete the corresponding redundant timestamp member.

TBR: magjed@webrtc.org
Bug: webrtc:9378
Change-Id: I27a0ec83fb20ac3da58ba32b86cf794a154deca0
Reviewed-on: https://webrtc-review.googlesource.com/83123
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23594}
2018-06-13 12:23:49 +00:00
e4a17c572d Moved timing related logic into its own function in webrtc::PayloadRouter.
Bug: none
Change-Id: I4eae7a555132654dc2d0747e7d3a7ff523523058
Reviewed-on: https://webrtc-review.googlesource.com/81242
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23593}
2018-06-13 11:28:46 +00:00
1f4d7a2848 Revert "Refactor the regathering of candidates in P2PTransportChannel."
This reverts commit 14f8aba9967ac2f1789ede12ff66107962757fb5.

Reason for revert: breaking internal tests

Original change's description:
> Refactor the regathering of candidates in P2PTransportChannel.
> 
> The functionality of regathering candidates is refactored to a separate
> regathering controller owned by P2PTransportChannel. This refactoring
> is part of a long-term plan to restructure a modularied
> P2PTransportChannel and it would also benefit the addition of autonomous
> regathering of candidates that is proactive to the ICE states in the
> near future.
> 
> Bug: None
> Change-Id: I74cea974ea628430c77b5d51b7c9179ddffc690d
> Reviewed-on: https://webrtc-review.googlesource.com/75820
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23588}

TBR=deadbeef@webrtc.org,pthatcher@webrtc.org,qingsi@google.com

Change-Id: I8b08351c9a3fcf89e2a25ed2c668c335cbd2d2d0
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Reviewed-on: https://webrtc-review.googlesource.com/83300
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23592}
2018-06-13 05:55:45 +00:00
241d0c16c0 Remove ContinualGatheringPolicy::GATHER_CONTINUALLY_AND_RECOVER.
This policy is not implemented.

Bug: None
Change-Id: I6c162d61c2488a4726c20df5c14439f83633a198
Reviewed-on: https://webrtc-review.googlesource.com/76041
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23591}
2018-06-13 01:00:00 +00:00
aed7164bde Updated PeerConnection integration test to fix race condition.
The PeerConnection integration test was creating TurnServers on the
stack on the signaling thread. This could cause a race condition problem
when the test was being taken down. Since the turn server was destructed
on the signaling thread, a socket might still try and send to it after
it was destroyed causing a seg fault. This change creates/destroys the
TestTurnServers on the network thread to fix this issue.

Bug: None
Change-Id: I080098502b737f0972ce2fa5357920de057a3312
Reviewed-on: https://webrtc-review.googlesource.com/81301
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23590}
2018-06-13 00:20:10 +00:00
2cf61e3324 Roll chromium_revision ef538e3112..9df92afb16 (566490:566630)
Change log: ef538e3112..9df92afb16
Full diff: ef538e3112..9df92afb16

Roll chromium third_party cbc2a20101..01aaf419f6
Change log: cbc2a20101..01aaf419f6

Changed dependencies:
* src/base: 02a6c4cdd0..6b48dbc0d2
* src/build: 3c4d6b6d24..169887d089
* src/ios: 0c6d3816a0..b0428063aa
* src/testing: e0597e0b5d..5951b2830b
* src/third_party/libvpx/source/libvpx: 87386826a9..37a0283b18
* src/tools: c7862334ce..e61dbb7de4
DEPS diff: ef538e3112..9df92afb16/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,marpan@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
NO_AUTOIMPORT_DEPS_CHECK=true

Change-Id: I5b0fde60118e7aef7929c7d33461b81078f85f93
Reviewed-on: https://webrtc-review.googlesource.com/83281
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23589}
2018-06-13 00:08:00 +00:00
14f8aba996 Refactor the regathering of candidates in P2PTransportChannel.
The functionality of regathering candidates is refactored to a separate
regathering controller owned by P2PTransportChannel. This refactoring
is part of a long-term plan to restructure a modularied
P2PTransportChannel and it would also benefit the addition of autonomous
regathering of candidates that is proactive to the ICE states in the
near future.

Bug: None
Change-Id: I74cea974ea628430c77b5d51b7c9179ddffc690d
Reviewed-on: https://webrtc-review.googlesource.com/75820
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23588}
2018-06-12 22:51:40 +00:00
b57e169f3c Add a flag to actively reset the SRTP parameters
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.

The flag is added to Android and Objc wrapper as well.

This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).

TBR=sakal@webrtc.org, denicija@webrtc.org

Bug: chromium:835958
Change-Id: I6db025e1c69bf83e1b1908f7df4627430db9920c
Reviewed-on: https://webrtc-review.googlesource.com/83101
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23587}
2018-06-12 20:32:00 +00:00
9dce71b983 Reland "Use absl::optional instead or rtc::Optional"
This reverts commit 28e6a164bf9d2a42545d058bd50d39e1767f7398.

Reason for revert: the static initializer removed from abseil

Original change's description:
> Revert "Use absl::optional instead or rtc::Optional"
>
> This reverts commit 7ba9e92fa0dfb16579f4f6ecd746397bdfdd174d.
>
> Reason for revert: Breaks Chromium static initialized regression test.
> https://ci.chromium.org/p/chromium/builders/luci.chromium.try/android-marshmallow-arm64-rel/5068
>
> Original change's description:
> > Use absl::optional instead or rtc::Optional
> >
> > BUG: webrtc:9078
> > Change-Id: I69aedce324d86e8894b81210a2de17c5ef68fd11
> > Reviewed-on: https://webrtc-review.googlesource.com/77082
> > Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
> > Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#23440}
>
> TBR=danilchap@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org
>
> Change-Id: I09ae74bddc69d0b25c8dfbcacc4ec906b34ca748
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Reviewed-on: https://webrtc-review.googlesource.com/79980
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23449}

TBR=danilchap@webrtc.org,mbonadei@webrtc.org,kwiberg@webrtc.org

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

Change-Id: Ib5dc71fb63fe02b78743b03f8252b962616eead0
Bug: webrtc:9078
Reviewed-on: https://webrtc-review.googlesource.com/82760
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23586}
2018-06-12 19:13:21 +00:00
57dc9e367c Roll chromium_revision ffaf1e2ba6..ef538e3112 (565764:566490)
Change log: ffaf1e2ba6..ef538e3112
Full diff: ffaf1e2ba6..ef538e3112

Roll chromium third_party fb3dc2a0aa..cbc2a20101
Change log: fb3dc2a0aa..cbc2a20101

Changed dependencies:
* src/base: 2743076235..02a6c4cdd0
* src/build: 459adce3eb..3c4d6b6d24
* src/ios: 71b35d6ee6..0c6d3816a0
* src/testing: f624b1f4b7..e0597e0b5d
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/eca23c365a..fdacd1639e
* src/third_party/depot_tools: e05f18d477..e09b6845cf
* src/third_party/freetype/src: 0589f6e6ee..8f1ed54877
* src/third_party/googletest/src: 145d05750b..9077ec7efe
* src/tools: 76e5757c8f..c7862334ce
DEPS diff: ffaf1e2ba6..ef538e3112/DEPS

Clang version changed 332838:334100
Details: ffaf1e2ba6..ef538e3112/tools/clang/scripts/update.py

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
NO_AUTOIMPORT_DEPS_CHECK=true

Change-Id: Iac45f00b35127f886b32aac8b79578fbe528fb00
Reviewed-on: https://webrtc-review.googlesource.com/83220
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23585}
2018-06-12 18:26:40 +00:00
abe301fe6c Add HeaderExtensions to RtpParameters
Bug: webrtc:7580
Change-Id: I4fcf3e8bc4975a6b2baa6f24a17c254d2bf521d9
Reviewed-on: https://webrtc-review.googlesource.com/78288
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23584}
2018-06-12 17:01:40 +00:00
867e510ef5 Enable send side audio TWCC only if WebRTC-Audio-ForceNoTWCC is not enabled.
This will avoid enabling TWCC for calls having WebRTC-Audio-SendSideBwe enabled on one side of the call but not on the other.

Currently the side supporting audio BWE indicates TWCC extension in SDP but the side that does not support will not. As the result the not supporting side will send TWCC but will not use it and the side supporting audio BWE will not send TWCC.

Bug: webrtc:8243
Change-Id: I4d59e78998982051004b8ad86c24b9be34fc095f
Reviewed-on: https://webrtc-review.googlesource.com/82803
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23583}
2018-06-12 16:01:20 +00:00
910540d55a Explicitly setting use_lld=false on MSVC bots.
In order to unblock the Chromium Roll, WebRTC should set use_lld=false
when MSVC is used (as discussed here [1]).

[1] - https://chromium-review.googlesource.com/c/chromium/src/+/1092611

Bug: None
Change-Id: Ia052d3d8842871c3051fe36991396976f5839f4c
Reviewed-on: https://webrtc-review.googlesource.com/83102
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23582}
2018-06-12 13:49:38 +00:00
7af087a918 Metal renderer does not handle i420 frames correctly.
Bug: webrtc:9389
Change-Id: If036f3f6208f5ce8aea1cabd1d7ccff1dfcc0808
Reviewed-on: https://webrtc-review.googlesource.com/83160
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23581}
2018-06-12 12:56:24 +00:00
dd3e0ab2bf Make rtc_software_fallback_wrappers target visible.
Need to depend on them from Chromium.

Bug: webrtc:7925
Change-Id: Iea1bb3b937c602920bfd87f885c87c790ac7bc17
Reviewed-on: https://webrtc-review.googlesource.com/82061
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23580}
2018-06-12 12:51:34 +00:00
cf15eb57ff Release ADM after passing it to PCF in AppRTC
Bug: webrtc:7452
Change-Id: I27e560c8f86ebf2df2162f30e5f9e5345ec0ecdb
Reviewed-on: https://webrtc-review.googlesource.com/83122
Commit-Queue: Paulina Hensman <phensman@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23579}
2018-06-12 11:42:41 +00:00
d294c85b36 LogMessage::UpdateMinLogSeverity: Don't ignore all but the last stream
Bug reported by andrey.semashev@gmail.com.

Bug: webrtc:9364
Change-Id: I49ef8969afc5bcd55d9e5ecbe644fe190a436c7b
Reviewed-on: https://webrtc-review.googlesource.com/83124
Reviewed-by: Jonas Olsson <jonasolsson@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23578}
2018-06-12 10:56:21 +00:00
798b28279e Don't update internal state of the FrameBuffer2 when an undecodable frame is inserted.
Bug: chromium:844313
Change-Id: I034bcb47092815695084e37c81150bafbfbc6b9c
Reviewed-on: https://webrtc-review.googlesource.com/79944
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23577}
2018-06-12 09:26:09 +00:00
dadaaee3e8 Remove stringstreams from p2p/
Bug: webrtc:8982
Change-Id: Ibb01bbb7e5f07d266ac7dff45932afe95abc761c
Reviewed-on: https://webrtc-review.googlesource.com/82801
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23576}
2018-06-12 08:08:57 +00:00
39da65b24d remove unused UNSHIPPED trace macros
Bug: webrtc:9387
Change-Id: Id8d80b5187f836dba42510910cfad24685fe6b18
Reviewed-on: https://webrtc-review.googlesource.com/82940
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23575}
2018-06-12 07:53:47 +00:00
e1d617c266 Delay the creation of the platform thread in TestAudioDeviceModule.
This allows constructing TestAudioDeviceModule on a different thread
than the worker thread and avoids unnecessary invoke. Before,
thread->Start() would fail in a thread check.

Bug: b/79961243
Change-Id: I5c55d8feada2b0ae12bc121f3f795e76a8d04059
Reviewed-on: https://webrtc-review.googlesource.com/82941
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23574}
2018-06-12 07:36:28 +00:00
9614a313b8 Remove manual references to exe_and_shlib_deps
After [1], a manual dependency on exe_and_shlib_deps is no longer necessary
since it's automatically added.  This CL removes all remaining manual references
to exe_and_shlib_deps.

[1] d7ed1f0a9c

BUG=chromium:845700
R=tommi@webrtc.org

Change-Id: I92942bc08c0e34c5c39df3c71f56f89476f8d95c
Reviewed-on: https://webrtc-review.googlesource.com/83061
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23573}
2018-06-12 06:07:16 +00:00
7685e86fa6 Pass the RtcEventLog instance to ICE via JsepTransportController.
This CL fixes a bug that the RtcEventLog owned by PeerConnection was not
passed to P2PTransportChannel after JsepTransportController was
introduced to deprecate the legacy TransportController.

Bug: webrtc:9337
Change-Id: I406cd9c0761dfe67f969aa99c6141e1ab38249d5
Reviewed-on: https://webrtc-review.googlesource.com/79964
Commit-Queue: Qingsi Wang <qingsi@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23572}
2018-06-12 05:04:35 +00:00
6c789e08d5 Revert "Add a flag to actively reset the SRTP parameters"
This reverts commit bae103126c5bdaf1361bcff4750eb5ebe10020ee.

Reason for revert: Merge native code change with Android and Objc wrapper.

Original change's description:
> Add a flag to actively reset the SRTP parameters
> 
> Add a new flag to RtcConfiguration. By setting that flag to true, the
> SRTP parameters will be reset whenever the DTLS transports are reset
> after every offer/answer negotiation.
> 
> This should only be used as a workaround for the linked bug, if the
> application knows that the other party is affected (for instance,
> using a version number).
> 
> Bug: chromium:835958
> Change-Id: Ifb4b99f68dc272507728ab59c07627f0d1b9c605
> Reviewed-on: https://webrtc-review.googlesource.com/81642
> Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23570}

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

Change-Id: Ibd7a3b8f45ff8df4af33d758f8fd3e2d5158e8e2
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:835958
Reviewed-on: https://webrtc-review.googlesource.com/83080
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23571}
2018-06-12 00:56:07 +00:00
bae103126c Add a flag to actively reset the SRTP parameters
Add a new flag to RtcConfiguration. By setting that flag to true, the
SRTP parameters will be reset whenever the DTLS transports are reset
after every offer/answer negotiation.

This should only be used as a workaround for the linked bug, if the
application knows that the other party is affected (for instance,
using a version number).

Bug: chromium:835958
Change-Id: Ifb4b99f68dc272507728ab59c07627f0d1b9c605
Reviewed-on: https://webrtc-review.googlesource.com/81642
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23570}
2018-06-11 23:06:26 +00:00
a46bd4b9c7 Reland "Move class VideoCodec from common_types.h to its own api header file."
This is a reland of efc71e565e9b36bcdfb4571f59e34bbd8fabd0cd

Differs from the original cl by not widening the type of
VideoCodec::width and VideoCodec::height.

Original change's description:
> Move class VideoCodec from common_types.h to its own api header file.
>
> Bug: webrtc:7660
> Change-Id: I91f19bfc2565461328f30081f8383e136419aefb
> Reviewed-on: https://webrtc-review.googlesource.com/79881
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
> Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23544}

Bug: webrtc:7660
Change-Id: I7cf74a85a61ea2b831e6f32b3b3e17514ebefec8
Reviewed-on: https://webrtc-review.googlesource.com/82140
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23569}
2018-06-11 19:23:20 +00:00
c17ca5354a Delete deprecated VideoTrackSource constructor.
This is a followup to
https://webrtc-review.googlesource.com/c/src/+/78403

Bug: None
Change-Id: I6dc29b13b333ff8836d7d0f3dc21aba0ad66b5bb
Reviewed-on: https://webrtc-review.googlesource.com/80243
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23568}
2018-06-11 19:15:10 +00:00
2ac64467c4 Document that preferred VideoFrame constructor takes no RTP timestamp.
And update most internal calls to use it.

Bug: webrtc:5740, webrtc:9372
Change-Id: Ib57d4ebfa7b0729af6d22981a792f0fdadf8a13f
Reviewed-on: https://webrtc-review.googlesource.com/81743
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23567}
2018-06-11 18:42:40 +00:00
425f713d24 Delete unused methods in VCMFrameBuffer and VCMSessionInfo.
Bug: None
Change-Id: Ia97bb14ac9fa1a31dae248fc5a0f58e07b588ec7
Reviewed-on: https://webrtc-review.googlesource.com/82164
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23566}
2018-06-11 18:39:50 +00:00
6d19180030 Fix increase in send rate when not receiving feedback
Store the last known throughput estimate and use that if we're lacking a new measurement.

Bug: webrtc:9363
Change-Id: Ib7a9a495b446bd0f5799382cc095ccff894e0c2b
Reviewed-on: https://webrtc-review.googlesource.com/81749
Commit-Queue: Anastasia Koloskova <koloskova@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23565}
2018-06-11 16:44:19 +00:00
6bac5c182a Remove TimingFrameFlags from its old place after it was moved
Bug: webrtc:9351
Change-Id: I97df702adddb600804fe643e17d84331073f4ba1
Reviewed-on: https://webrtc-review.googlesource.com/81681
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23564}
2018-06-11 16:38:49 +00:00
80a0a4c482 Remove software fallback in Android hardware codec factories.
Remove backwards compatiblity. Users who need software codecs should
migrate to the DefaultVideoCodecFactories.

Bug: webrtc:7925
Change-Id: Ifb41c9511d53c17c83222422c221b595bc056cb2
Reviewed-on: https://webrtc-review.googlesource.com/82920
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23563}
2018-06-11 15:59:09 +00:00
24db1c91a1 Remove unused iostream import
Bug: webrtc:8982
Change-Id: I789babea16ec4a51fda14340dc617f1aaf0fa80a
Reviewed-on: https://webrtc-review.googlesource.com/82820
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23562}
2018-06-11 15:28:19 +00:00
fddaf7528a AEC3: Increase the look window in the delay estimator.
Bug: webrtc:9374,chromium:850525
Change-Id: I587cb7951acf8e5ec92d9941f1979ba2c9887876
Reviewed-on: https://webrtc-review.googlesource.com/81747
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23561}
2018-06-11 15:22:59 +00:00
43568dd67e Remove stringstreams from pc/
Bug: webrtc:8982
Change-Id: I85ae004e50da2c84b3cb018c6111d8c9db69fbec
Reviewed-on: https://webrtc-review.googlesource.com/82165
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23560}
2018-06-11 15:20:59 +00:00
6cdd546ba2 Remove default argument in GetSimulcastConfig.
Bug: webrtc:9368
Change-Id: Id0f5392daaeda826abad36604550317ce9431544
Reviewed-on: https://webrtc-review.googlesource.com/81662
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23559}
2018-06-10 13:44:48 +00:00
fe89271bee Android: Use JavaToNativeString in PeerConnectionFactory_InitializeFieldTrials
Bug: b/109736242
Change-Id: I56a404c2e6c8c842b23465249f72897cee0878b5
Reviewed-on: https://webrtc-review.googlesource.com/81742
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23558}
2018-06-09 17:38:06 +00:00
b7beba4c93 Roll chromium_revision e9fcce491e..ffaf1e2ba6 (565609:565764)
Change log: e9fcce491e..ffaf1e2ba6
Full diff: e9fcce491e..ffaf1e2ba6

Roll chromium third_party b1a95b2d51..fb3dc2a0aa
Change log: b1a95b2d51..fb3dc2a0aa

Changed dependencies:
* src/base: 6d59c6b0f4..2743076235
* src/build: 472d5df73f..459adce3eb
* src/ios: 2c466eefd8..71b35d6ee6
* src/testing: 4162fb6122..f624b1f4b7
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c9718952e5..eca23c365a
* src/third_party/depot_tools: 8e6f58c7e6..e05f18d477
* src/third_party/libFuzzer/src: fda403cf93..873dc11d9a
* src/tools: 1702b5a977..76e5757c8f
DEPS diff: e9fcce491e..ffaf1e2ba6/DEPS

No update to Clang.

TBR=buildbot@webrtc.org,
BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
NO_AUTOIMPORT_DEPS_CHECK=true

Change-Id: Ib9d3e2073829e4386d23970dda3f50f5bf2ee4da
Reviewed-on: https://webrtc-review.googlesource.com/82380
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23557}
2018-06-08 23:16:19 +00:00
1ef88aee29 Add kNumValues to IceCandidate* enums and move kUnknown to the front.
Bug: None
Change-Id: Ia20fc06a96d78b4f842a849a64f7c580b6663fd0
Reviewed-on: https://webrtc-review.googlesource.com/82281
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23556}
2018-06-08 22:15:38 +00:00
b90e63c620 Fix: NetEq PacketBuffer logs discarded packet with wrong codec level when new packet replaces the lower level packet
Bug: webrtc:9370
Change-Id: I59606ef6ea9bbf26de844a2fd3f597856271a86a
Reviewed-on: https://webrtc-review.googlesource.com/81700
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Reviewed-by: Minyue Li <minyue@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23555}
2018-06-08 14:58:18 +00:00
ec9c745228 Adds support for new Windows ADM with limited API support.
Summary of what this CL does:

Existing users can keep using the old ADM for Windows as before.

A new ADM for Windows is created and a dedicated factory method is used
to create it. The old way (using AudioDeviceImpl) is not utilized.

The new ADM is based on a structure where most of the "action" takes
place in new AudioInput/AudioOutput implementations. This is inline
with our mobile platforms and also makes it easier to break out common
parts into a base class.

The AudioDevice unittest has always mainly focused on the "Start/Stop"-
parts of the ADM and not the complete ADM interface. This new ADM supports
all tests in AudioDeviceTest and is therefore tested in combination with
the old version. A value-parametrized test us added for Windows builds.

Improved readability, threading model and makes the code easier to maintain.

Uses the previously landed methods in webrtc::webrtc_win::core_audio_utility.

Bug: webrtc:9265
Change-Id: If2894b44528e74a181cf7ad1216f57386ee3a24d
Reviewed-on: https://webrtc-review.googlesource.com/78060
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23554}
2018-06-08 14:44:38 +00:00
488eb98616 Setting resolution alignment to 4 on iOS.
Bug: webrtc:9381
Change-Id: I6fb6cc6ffa197ca581462e308a857ac38e10b9a1
Reviewed-on: https://webrtc-review.googlesource.com/82162
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23553}
2018-06-08 14:17:07 +00:00
443e71f528 Revert "Disabling VeryLowBitrateVP9 to unblock roll."
This reverts commit 16e28d143a32ff3552efe0a014178f68006812b8.

Reason for revert: Fix has supposedly landed upstream.

Original change's description:
> Disabling VeryLowBitrateVP9 to unblock roll.
> 
> This should be re-enabled very soon since the libvpx thinks this
> is fixed upstream and is only waiting for merge.
> 
> TBR=marpan@google.com
> 
> Bug: webrtc:9292
> Change-Id: Ib78ea1462059c333b7168a52756329dc9a385b54
> Reviewed-on: https://webrtc-review.googlesource.com/81660
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23525}

TBR=phoglund@webrtc.org,marpan@google.com

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

Bug: webrtc:9292
Change-Id: I995953070536e8ee3540e7c30bc11dc1200e0463
Reviewed-on: https://webrtc-review.googlesource.com/82200
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23552}
2018-06-08 13:55:25 +00:00
2d61162831 Remove stringstreams from media/sctp/
Bug: webrtc:8982
Change-Id: I0d92f56b628e7cd50e7c853d3bfe0049dcf71425
Reviewed-on: https://webrtc-review.googlesource.com/76563
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23551}
2018-06-08 13:51:51 +00:00
25b41f8c11 remove unused stringstream import
No-Try: true
Bug: webrtc:8982
Change-Id: I24537a3d4fab2d0caa4e62ed791c9939be8e4567
Reviewed-on: https://webrtc-review.googlesource.com/77120
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23550}
2018-06-08 13:03:34 +00:00