Commit Graph

22854 Commits

Author SHA1 Message Date
65c61dcfce Android: Add helper class for generating OpenGL shaders
This CL adds a helper class GlShaderBuilder to build an instances of
RendererCommon.GlDrawer that can accept multiple input sources
(OES, RGB, or YUV) using a generic fragment shader as input.

Bug: webrtc:9355
Change-Id: I14a0a280d2b6f838984f7b60897cc0c58e2a948a
Reviewed-on: https://webrtc-review.googlesource.com/80940
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23622}
2018-06-15 09:06:45 +00:00
8643b78750 Moved NackModule and VCMPacket to their own targets
Bug: webrtc:9373
Change-Id: I1e882b734dcafb5c633eabf08bb8a1a6a407a251
Reviewed-on: https://webrtc-review.googlesource.com/81744
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23621}
2018-06-15 09:00:25 +00:00
88aee288f8 Remove support for old test modes in EncodeDecodeTest
This test is so old, it used to be interactive with an automated mode
bolted on to the side. That automatic mode is the only one that's used
nowadays.

Bug: webrtc:8396
Change-Id: I3b473f53ff6afa363b9691e8471a5754f46d3d3f
Reviewed-on: https://webrtc-review.googlesource.com/83583
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23620}
2018-06-15 08:25:51 +00:00
d477129ac0 Remove dead RED code in TestRedFec
Bug: webrtc:8396
Change-Id: I96e70e9290fda0d20f1544d2bfe4307f80ca8693
Reviewed-on: https://webrtc-review.googlesource.com/83585
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23619}
2018-06-15 07:54:51 +00:00
8fbe4f10e2 Remove executable insert_packet_with_timing
It appears to have been created in mid-2013, and hasn't been changed
since except to keep the compiler happy when surrounding code changed.
It crashes when I try to run it without arguments, and no one
remembers how to use it.

Bug: webrtc:8396
Change-Id: I2eae36cf468f28c5bf05c85e6a3aaeebc48a1ffc
Reviewed-on: https://webrtc-review.googlesource.com/83581
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23618}
2018-06-15 07:31:30 +00:00
0f173bde9e Revert "Drop tools/gyp from dependencies."
This reverts commit 724a97d08d35b69d934b1c09fc2e0f4dd4d47f76.

Reason for revert: We have this error during gclient runhooks on some
windows bots:
Downloading https://commondatastorage.googleapis.com/chromium-browser-clang/Win/clang-334100-1.tgz .......... Done.
Traceback (most recent call last):
  File "src/tools/clang/scripts/update.py", line 927, in <module>
    sys.exit(main())
  File "src/tools/clang/scripts/update.py", line 923, in main
    return UpdateClang(args)
  File "src/tools/clang/scripts/update.py", line 470, in UpdateClang
    CopyDiaDllTo(os.path.join(LLVM_BUILD_DIR, 'bin'))
  File "src/tools/clang/scripts/update.py", line 396, in CopyDiaDllTo
    dia_path = os.path.join(GetVSVersion().Path(), 'DIA SDK', 'bin', 'amd64')
  File "src/tools/clang/scripts/update.py", line 388, in GetVSVersion
    import gyp.MSVSVersion
ImportError: No module named gyp.MSVSVersion

Looks like if toolchain is not downloaded before then it failed to
download it from scratch.

Original change's description:
> Drop tools/gyp from dependencies.
> 
> Bug: webrtc:6323
> Change-Id: I894f0ea95fb6707242a061947b4f4602b48910e6
> Reviewed-on: https://webrtc-review.googlesource.com/6763
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Commit-Queue: Niels Moller <nisse@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23613}

TBR=phoglund@webrtc.org,nisse@webrtc.org

Change-Id: I38ff915cccfdcc80af9e2f82130bccd33bf7ea86
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:6323
Reviewed-on: https://webrtc-review.googlesource.com/83744
Reviewed-by: Artem Titov <titovartem@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23617}
2018-06-14 23:41:25 +00:00
0a5fdbb455 Use RTC_HISTOGRAM_ENUMERATION to report SRTP/SRTCP unprotect error.
Besides using the MetricsObserverInterface, using RTC_HISTOGRAM_ENUMERATION
directly using RTC_HISTOGRAM_ENUMERATION to report the error which is
needed by internal projects.

Bug: b/110121202, webrtc:9409
Change-Id: I1aaece91200905ea0495229dc2b5e62b1d61279b
Reviewed-on: https://webrtc-review.googlesource.com/83565
Commit-Queue: Zhi Huang <zhihuang@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23616}
2018-06-14 18:35:11 +00:00
9eb38866cd Adds field trial parser.
Bug: webrtc:9346
Change-Id: Ibd07a1753feaa40d4be4d465d61f55bc8a8a9325
Reviewed-on: https://webrtc-review.googlesource.com/80263
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Christoffer Rodbro <crodbro@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23615}
2018-06-14 16:02:38 +00:00
7c32c866c0 Metal view: Update drawable size when rotating.
Bug: webrtc:9407
Change-Id: I8d6651eb4cd22c83a2dddbdbd890f34a61002f97
Reviewed-on: https://webrtc-review.googlesource.com/83586
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Peter Hanspers <peterhanspers@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23614}
2018-06-14 13:46:06 +00:00
724a97d08d Drop tools/gyp from dependencies.
Bug: webrtc:6323
Change-Id: I894f0ea95fb6707242a061947b4f4602b48910e6
Reviewed-on: https://webrtc-review.googlesource.com/6763
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23613}
2018-06-14 13:38:58 +00:00
a6fc6362ed Add ivoc@ and saza@ to audio_processing OWNERS
NOTRY=True

Bug: None
Change-Id: Idab1a031254f527c732bcf939c991c6b17aabd74
Reviewed-on: https://webrtc-review.googlesource.com/83580
Commit-Queue: Sam Zackrisson <saza@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23612}
2018-06-14 12:18:07 +00:00
6507054db1 Android: Add tests for VideoFrame.Buffer.toI420() and cropAndScale()
This CL adds tests that are primarily targeting
VideoFrame.Buffer.toI420() and cropAndScale(), but includes the whole
chain for YuvConverter, GlRectDrawer, and VideoFrameDrawer.

It also includes a couple of fixes to bugs that were exposed by the new
tests.

Bug: webrtc:9186, webrtc:9391
Change-Id: I5eb62979a8fd8def28c3cb2e82dcede57c42216f
Reviewed-on: https://webrtc-review.googlesource.com/83163
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23611}
2018-06-14 11:06:37 +00:00
d1f970dc43 Change echo detector to scoped_refptr
The echo detector is currently stored as a unique_ptr, but when injecting an echo detector, a scoped_refptr makes more sense since the ownership will be shared.

Bug: webrtc:8732
Change-Id: I2180014acb84f1cd5c361864a444b7b6574520f5
Reviewed-on: https://webrtc-review.googlesource.com/83325
Commit-Queue: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23610}
2018-06-14 09:51:41 +00:00
4e952a3f44 Remove unused WavFile::FormatAsString method.
This lets us remove stringstreams from common_audio/

Bug: webrtc:8982
Change-Id: I450d87dc50090e838edabc7c1db645aca9c1b0f7
Reviewed-on: https://webrtc-review.googlesource.com/82163
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23609}
2018-06-14 09:05:20 +00:00
671cae2c7c Handle FileRotatingStreams with long file names
Bug: webrtc:9392
Change-Id: I7b42b1a6ed1b646c244bc64f1bad92a2f38e5539
Reviewed-on: https://webrtc-review.googlesource.com/83162
Commit-Queue: Jonas Olsson <jonasolsson@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23608}
2018-06-14 08:21:48 +00:00
1b36894f06 Reland "Refactor the regathering of candidates in P2PTransportChannel."
This is a reland of 14f8aba9967ac2f1789ede12ff66107962757fb5

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}

Bug: None
Change-Id: I7308e2aef692edd4f0bf9717a88ba2dfba4383a6
Reviewed-on: https://webrtc-review.googlesource.com/83360
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23607}
2018-06-14 01:40:45 +00:00
5d16d7fd52 Add a DCHECK for null port in FakePortAllocator.
If the socket server of the thread where FakePortAllocator lives is not
configured to be a VirtualSocketServer, there is a chance that we have a
null port in FakePortAllocator::StartGettingPort after creating the test
UDP port (for example, no permission to create a real socket if using a
PhysicalSocketServer), and subsequently this results in a crash when
connecting a signal in the port to a slot.

Bug: webrtc:9406
Change-Id: I1ba4526f7b9e104bed556f61d9348edc426fc1fc
Reviewed-on: https://webrtc-review.googlesource.com/83480
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23606}
2018-06-14 00:37:15 +00:00
60b6c1dfa9 [Unified Plan] Clear RtpSender "SSRC" when the SDP has no send streams
This fixes a crash that occurs with this sequence of events:
1. AddTrack. SetLocalDescription(CreateOffer())
2. RemoveTrack. SetLocalDescription(CreateOffer())
3. AddTrack.

When AddTrack is called again it re-uses the RtpTransceiver/
RtpSender and try to configure the underlying MediaChannel. But the
MediaChannel would DCHECK since the send stream had been destroyed
by the SLD in 2. and would not get created until SLD is called
again.

Bug: webrtc:9401
Change-Id: I4b5572886e17263aaa4ce0408663444d72e09243
Reviewed-on: https://webrtc-review.googlesource.com/83420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23605}
2018-06-14 00:31:15 +00:00
f7d7e90c5e Replace std:remove on vector::erase in streamparams_unittest.cc
Replace std:remove on vector::erase to actually remove object from
the vector.

Change-Id: I1eae7a038769ca4eb45a2f9238ca7aa86b3c38a4
Bug: webrtc:9405
Reviewed-on: https://webrtc-review.googlesource.com/83342
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23604}
2018-06-13 22:20:04 +00:00
aeb0a6475b AEC3: Increase the range of reported echo path delay metrics
TBR: gustaf@webrtc.org
Bug: webrtc:9375,chromium:850538
Change-Id: I037e2cfe24ee297b90b4f70b744f735e43015d92
Reviewed-on: https://webrtc-review.googlesource.com/81748
Reviewed-by: Per Åhgren <peah@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23603}
2018-06-13 18:13:21 +00:00
b7700d370d Android: Fix VideoTrack behavior for adding/removing VideoSinks
In particular:
 * Trying to remove a VideoSink that was not attached should be a no-op.
 * Trying to remove a null VideoSink should be a no-op.
 * Adding the same VideoSink multiple times should a no-op, and should
   not create duplicate native VideoSinks.
 * Trying to add a null VideoSink should throw an exception in the Java
   layer instead of crashing in native code.

This CL also adds tests that verify these behaviors.

Bug: webrtc:9403
Change-Id: I928b7bb7f683634e287d7fec9e26f4179f73c150
Reviewed-on: https://webrtc-review.googlesource.com/83322
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23602}
2018-06-13 15:51:17 +00:00
493c78a9dc Replace all use of rtc::Pathname in generator_unittest.cc.
Bug: webrtc:7345
Change-Id: Ic804fcfd2456e16a3f9e448677d0b7bc857822a8
Reviewed-on: https://webrtc-review.googlesource.com/80484
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23601}
2018-06-13 15:09:24 +00:00
fabb12e042 Introduce list of fields to put into codec agnostic descriptor
Bug: webrtc:9361
Change-Id: Iff44f289ffcecf7e4f997d5001958ab22124910f
Reviewed-on: https://webrtc-review.googlesource.com/81241
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23600}
2018-06-13 14:55:09 +00:00
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