21193 Commits

Author SHA1 Message Date
Tommi
8e545eee1e Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."
This reverts commit 6780c51b23516803dc27173d10ba98d018780447.

Reason for revert:

More details in crbug.com/810292

Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
> 
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
> 
> R=​deadbeef@webrtc.org
> 
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}

TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org

Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
2018-02-08 16:25:31 +00:00
Rasmus Brandt
2018823a68 Remove unused field |output_filename| from TestConfig.
Bug: webrtc:8448
Change-Id: I8bb35f6d66112c6590564815e10cb4ec7b516268
Reviewed-on: https://webrtc-review.googlesource.com/49820
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21960}
2018-02-08 15:23:19 +00:00
henrika
cb87efd7d3 Avoids issues with start of audio when audio was not initialized on Android
Bug: b/72444507
Change-Id: I44d6e03c13a49033682f8f0bdc10256f724068d3
Reviewed-on: https://webrtc-review.googlesource.com/48020
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Fredrik Solenberg <solenberg@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21959}
2018-02-08 15:04:39 +00:00
Per Åhgren
f4d1134bdc Adjusted tunings to increase AEC3 robustness against pipeline issues
Bug: chromium:810371,webrtc:8862
Change-Id: I2bfd3601c41caf608c21bec27133a175e3a7f2c5
Reviewed-on: https://webrtc-review.googlesource.com/49782
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21958}
2018-02-08 14:40:29 +00:00
Rasmus Brandt
2b304f1b2d Simplify CodecSettings helper function.
Bug: webrtc:8448
Change-Id: I4413fbaeab93690047e0f464b907bfd7f078778c
Reviewed-on: https://webrtc-review.googlesource.com/47500
Reviewed-by: Michael Horowitz <mhoro@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21957}
2018-02-08 14:38:59 +00:00
Niels Möller
d0dd90be62 New java ScalingSettings constructors.
Deprecate old constructors. Intended to make java api consistent with
the changes in https://webrtc-review.googlesource.com/c/src/+/46622.

Bug: webrtc:8830
Change-Id: Iadecb5d033b5de841873905af659d8d234b75c7d
Reviewed-on: https://webrtc-review.googlesource.com/49062
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21956}
2018-02-08 14:26:51 +00:00
Niels Möller
6f7bc08457 Rewrite FakeVideoTrackSource to not use VideoCapturer.
Bug: webrtc:6353
Change-Id: I992048868eebca1889e697950003b537b344bb53
Reviewed-on: https://webrtc-review.googlesource.com/49163
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21955}
2018-02-08 09:55:28 +00:00
Tommi
8595993c5b Update several tests: FakeVideoCapturer -> FakeVideoCapturerWithTaskQueue.
Bug: webrtc:8848
Change-Id: Iae41d6e47dbca563918f7283d902eb52b7839b12
Reviewed-on: https://webrtc-review.googlesource.com/49281
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21954}
2018-02-08 09:03:58 +00:00
Autoroller
c1cab0a134 Roll chromium_revision 5f99c89339..3d230c052f (534995:535313)
Change log: 5f99c89339..3d230c052f
Full diff: 5f99c89339..3d230c052f

Changed dependencies:
* src/base: 215c545cfb..05346abed5
* src/build: c0ec7a5422..3ba6ca6d32
* src/ios: e5a513f3b7..68272315d7
* src/testing: 1133d04410..96fadc248d
* src/third_party: aee0b83bb3..cb732ebd07
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/1d86294f15..b4a4bed9ad
* src/third_party/depot_tools: 4a92cc9a1f..6a60d7586a
* src/tools: 48b60acde4..93539cf31f
DEPS diff: 5f99c89339..3d230c052f/DEPS

No update to Clang.

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

Change-Id: Ia145f899591ba8aee473e8b9e8ffdbb5f1a69ce1
Reviewed-on: https://webrtc-review.googlesource.com/49640
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21953}
2018-02-08 06:27:38 +00:00
Joachim Bauch
6780c51b23 Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
from native apps if really necessary.

R=deadbeef@webrtc.org

Bug: webrtc:7670
Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
Reviewed-on: https://webrtc-review.googlesource.com/41420
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21952}
2018-02-07 21:56:01 +00:00
Tommi
8b0ebb9484 Change FakePeriodicVideoCapturer to use a TaskQueue instead of Thread.
This changes callbacks to OnFrame methods to occur on a task queue which
is in line with how it's called in production.

The change is essentially around inheriting from FakeVideoCapturerWithTaskQueue
instead of FakeVideoCapturer, but also removes the dependency on rtc::MessageHandler.

Along the way I'm also updating an ortc test that uses FakePeriodicVideoCapturer
and had a bug that was masked by the fact that FakePeriodicVideoCapturer
previously used rtc::Thread::Current internally, but was being called
by the wrong thread (and there were no checks for it).
As a result, I'm also adding a bunch of checks to help with correct usage.

Bug: webrtc:8841, webrtc:8848
Change-Id: I21b710873b508ebc55f8d2e4545d862766656871
Reviewed-on: https://webrtc-review.googlesource.com/49400
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21951}
2018-02-07 20:51:51 +00:00
Alex Narest
7ff6ca5844 Adds voice concealment periods reporting to neteq_rtpplay.
Change-Id: Ie5a89eacef8c1cf7d5a6220b045d2c331fef199e

Bug: webrtc:8847
Change-Id: Ie5a89eacef8c1cf7d5a6220b045d2c331fef199e
Reviewed-on: https://webrtc-review.googlesource.com/48100
Commit-Queue: Alex Narest <alexnarest@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21950}
2018-02-07 18:41:42 +00:00
Seth Hampson
f209cb52a4 Added ptime to RtpEncodingParameters.
ptime is in the w3 standard, but currently not in our api header.

Bug: webrtc:8819
Change-Id: I5af7ab2c901d129de7bf381aee34ae5bb9039495
Reviewed-on: https://webrtc-review.googlesource.com/46343
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21949}
2018-02-07 17:33:41 +00:00
Tommi
1829af6a39 Extend FakePeriodicVideoCapturer with FakeVideoCapturerWithTaskQueue.
FakeVideoCapturerWithTaskQueue overrides frame related methods
and delivers frame callbacks on a TaskQueue (separate thread),
as is (must be) expected by the implementations being tested.

I'm also moving the implementation out of the header and into
a separate source file.

In this CL, I'm updating one test to use the new class but
more will follow.

Bug: webrtc:8848
Change-Id: I5403c6bcc8b757e9d7fa9c368506667707b37b28
Reviewed-on: https://webrtc-review.googlesource.com/48360
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21948}
2018-02-07 16:42:01 +00:00
Rasmus Brandt
c334ce978c Remove legacy ctor from SimulcastEncoderAdapter.
Bug: None
Change-Id: I9c1472c2aef0133816466916e26378466510054a
Reviewed-on: https://webrtc-review.googlesource.com/47880
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21947}
2018-02-07 16:38:52 +00:00
Edward Lemur
e4275901fe Update frame_analyzer binary.
Update frame_analyzer binary, so that it includes changes to the
perf flags.

Bug: chromium:755660
Change-Id: I9e30cd83b807d2d4fa74a677dc5a69f8985a4ce2
Reviewed-on: https://webrtc-review.googlesource.com/48622
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21946}
2018-02-07 16:33:06 +00:00
Edward Lemur
260c39871b Add support for hyphens to rtc_base/flags
Make it possible to specify flags both with hyphens (--flag-name)
and underscores (--flag_name).

Bug: None
Change-Id: Ic02cdc2d5b9f7c75d06cdb6287a86ed432fd9daa
Reviewed-on: https://webrtc-review.googlesource.com/49204
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21945}
2018-02-07 16:32:01 +00:00
Sami Kalliomäki
2a25be6b06 Update AppRTCMobile AAR-project to SDK version 27.
Bug: None
Change-Id: I9f0b738cc52e5813f4159ffedc58890c3cd3f544
Reviewed-on: https://webrtc-review.googlesource.com/49160
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21944}
2018-02-07 16:22:31 +00:00
Henrik Lundin
2cbc20bb56 NetEq quality tests: avoid default preloading of the buffer
Before this change, the test used to preload the buffer with 10
packets before starting to pull out audio. With this change, the
preloading is determined by a new flag (--preload_packets) which
defaults to 0.

This affects all tests derived from NetEqQualityTest, i.e., all
binaries called neteq_*_quality_test.

Bug: none
Change-Id: I920845b968a81ea9972ce8a8e646df29aff200ba
Reviewed-on: https://webrtc-review.googlesource.com/49261
Reviewed-by: Alex Narest <alexnarest@webrtc.org>
Commit-Queue: Henrik Lundin <henrik.lundin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21943}
2018-02-07 16:19:31 +00:00
Rasmus Brandt
4b381afd8e Enforce that VideoProcessor is only run on a TaskQueue.
Prior to this change, the VideoProcessor was run on the main thread
in the unit tests. Using a TaskQueue there instead, we can be
stricter in the thread checks.

Bug: webrtc:8524
Change-Id: Ice7b68f7344fc52801dff7a98cbc219b7231bfbc
Reviewed-on: https://webrtc-review.googlesource.com/48921
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21942}
2018-02-07 15:42:21 +00:00
Sebastian Jansson
b0acec3679 Testing that media is resumed when congestion ends.
Expanded congestion window test to test that media is resumed after
being stopped due to congestion window getting filled. Previously only
the behavior that padding packets was sent in congested state was tested,
but not that media actually was resumed when feedback from the padding
packets was received.

Bug: None
Change-Id: Icd494c2e5386926f92c4d5dd0e8bc80c81608325
Reviewed-on: https://webrtc-review.googlesource.com/46262
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21941}
2018-02-07 15:31:26 +00:00
Sami Kalliomäki
11c51dd53d Update documentation for VideoEncoder.Callback#onEncodedImage.
Allows assuming that the buffer is not accessed after the call returns.

Bug: b/72675429
No-Try: True
Change-Id: Iff4a05433c6eed6aefec49ce67486966b1ed882f
Reviewed-on: https://webrtc-review.googlesource.com/49161
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21940}
2018-02-07 15:30:21 +00:00
Autoroller
5684921a16 Roll chromium_revision eeca1d8fa2..5f99c89339 (534891:534995)
Change log: eeca1d8fa2..5f99c89339
Full diff: eeca1d8fa2..5f99c89339

Changed dependencies:
* src/base: 76c6e329bb..215c545cfb
* src/build: 7e86dc487b..c0ec7a5422
* src/ios: c61b8482ad..e5a513f3b7
* src/testing: 63e2a50231..1133d04410
* src/third_party: 7e59438107..aee0b83bb3
* src/third_party/depot_tools: e117e46a68..4a92cc9a1f
* src/tools: ba396b0b2e..48b60acde4
DEPS diff: eeca1d8fa2..5f99c89339/DEPS

No update to Clang.

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

Change-Id: I789a763b653f610c4468a0c88108c1b9bb21f8e7
Reviewed-on: https://webrtc-review.googlesource.com/49240
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21939}
2018-02-07 15:06:18 +00:00
henrika
67417157d4 Removes TSAN suppression in AudioDeviceLinuxPulse which is no longer used in PC unittests
BUG=webrtc:5152

Change-Id: I95cef3a3edc62ef9a956706fa768d391ce19c868
Reviewed-on: https://webrtc-review.googlesource.com/49260
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21938}
2018-02-07 15:04:39 +00:00
Erik Språng
7b52f102ef Don't write pacer exit timestamp without pacer
And allow populating network2 timestamp if we want to preserve pacer
timestamp.

Bug: webrtc:8853
Change-Id: I895d5ce8a9cca8ceeec3bf08e2eff02bf3b2f5fd
Reviewed-on: https://webrtc-review.googlesource.com/48640
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21937}
2018-02-07 14:45:43 +00:00
Niels Möller
a8b150888c Stricter declarations in VideoStreamEncoder.
Mark overuse_detector_ pointer const, add a few
RTC_RUN_ON and RTC_PT_GUARDED_BY annotations.

Bug: none
Change-Id: Ibaf6d738f20fbffacfed42c36a34779be52dd9fc
Reviewed-on: https://webrtc-review.googlesource.com/46000
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21936}
2018-02-07 14:44:39 +00:00
Tommi
1f432e014b Update VideoCapturerTrackSourceTest to deliver frames via TaskQueue.
This is inline with how other peerconnection tests run.

Bug: webrtc:8848
Change-Id: Idd32a1000f00e3620220527fd60b83c6f03fdcaa
Reviewed-on: https://webrtc-review.googlesource.com/48140
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21935}
2018-02-07 14:25:08 +00:00
Fredrik Solenberg
ee0a756741 Remove TSAN suppressions in non-existing VoiceEngine code.
Bug: webrtc:4719
Change-Id: I7cda3fdc929adeca66dcdb2bc11b54322222c237
Reviewed-on: https://webrtc-review.googlesource.com/49205
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Fredrik Solenberg <solenberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21934}
2018-02-07 14:17:28 +00:00
Edward Lemur
27b2c9a3b0 Add PRESUBMIT test to make sure all valgrind dependencies are listed.
Add a PRESUBMIT test to make sure new valgrind runtime dependencies are
listed in tools_webrtc/valgrind/valgrind-webrtc.gni, so that they are
isolated and uploaded to swarming.

TBR=phoglund@webrtc.org

Bug: webrtc:7249
Change-Id: I090b790da0b6dd7c1e76c20f96e40df9000cbd62
Reviewed-on: https://webrtc-review.googlesource.com/42740
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21933}
2018-02-07 14:05:19 +00:00
Sami Kalliomäki
95a5f00484 Update HWVideoEncoder to reuse codec buffer instead of copying the data.
In practice, this is safe since WebRTC doesn't access the buffer after the
callback returns. This avoids unnecessary memory allocations causing out of
memory errors.

Bug: b/72675429
Change-Id: I2ed0224f40b7e1fa67c7aba625b99211f9c1e0a3
Reviewed-on: https://webrtc-review.googlesource.com/49162
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21932}
2018-02-07 13:52:58 +00:00
Sebastian Jansson
b537496520 Renamed PacketQueue2 to RoundRobinPacketQueue.
The previous name packet queue 2 had no indication on what the
difference was compared to the regular packet queue. This rename makes
it easier to understand the codebase.

Additionally the PacketQueueInterface class was introduced to make the
class hierarchy easier to follow. The round robin packet queue did not
extend the packet queue so there was no reason for inheriting from the
specific implementation.

Bug: None
Change-Id: Idbce081c751fbacd927632f5e71220887d0b5991
Reviewed-on: https://webrtc-review.googlesource.com/49120
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21931}
2018-02-07 13:11:38 +00:00
Rasmus Brandt
17cdcbb57b Access ImplementationName() from task queue.
Accessing this method from the test thread is illegal,
but doesn't always fail.

Bug: webrtc:8524
Change-Id: Ie0e84cc2fb63268fb6d7cbf0c3a58cb35312c16b
Reviewed-on: https://webrtc-review.googlesource.com/49061
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21930}
2018-02-07 12:38:48 +00:00
Niels Möller
1e06289cdb Delete macro RTC_ACCESS_ON, replaced by RTC_GUARDED_BY.
Both macros do the same thing, as wrappers for
__attribute__((guarded_by)), and more names for the same thing doesn't
add to clarity.

Bug: none
Change-Id: Iaaf7b21dbf3345ee90fee22c39b636823d195eb0
Reviewed-on: https://webrtc-review.googlesource.com/48361
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Tommi <tommi@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21929}
2018-02-07 10:07:28 +00:00
Sebastian Jansson
e6cefdf9c5 Moved congestion controller to goog_cc folder.
Bug: webrtc:8415
Change-Id: I2070da0cacf1dbfc4b6a89285af3e68fd03497ab
Reviewed-on: https://webrtc-review.googlesource.com/43841
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21928}
2018-02-07 09:50:48 +00:00
Danil Chapovalov
2a5ce2bcf8 Fix clang style errors in rtp_rtcp and dependant targets
Mark functions with override instead of virtual.
Add explicit non-trivial constructors/assign operators/destructors.
Define them in .cc files instead of inlining
use auto* instead of auto when deduced type is raw pointer

Bug: webrtc:163
Change-Id: I4d8a05d6a64fcc2ca16d02c5fcf9488fda832a6d
Reviewed-on: https://webrtc-review.googlesource.com/48781
Commit-Queue: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21927}
2018-02-07 09:48:28 +00:00
Sami Kalliomäki
740f8e72df Android: Fix a race condition in VideoDecoderWrapper.
Fixes a race condition where frame_extra_infos_ is accessed from
multiple threads by adding a lock.

Adds thread safety idioms to the file to guard agains similar mistakes
in the future.

Bug: b/72979294
Change-Id: I0f2f947282a5b3414f1351e9e8e52ad523f7d2f6
Reviewed-on: https://webrtc-review.googlesource.com/48641
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21926}
2018-02-07 08:53:11 +00:00
Autoroller
63a4d99c40 Roll chromium_revision 242dbc9f7b..eeca1d8fa2 (534785:534891)
Change log: 242dbc9f7b..eeca1d8fa2
Full diff: 242dbc9f7b..eeca1d8fa2

Changed dependencies:
* src/base: c56936b597..76c6e329bb
* src/ios: 25470f3759..c61b8482ad
* src/testing: 0d35757b9e..63e2a50231
* src/third_party: afca687501..7e59438107
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/e6e84629db..1d86294f15
* src/tools: 495b0aeb1c..ba396b0b2e
DEPS diff: 242dbc9f7b..eeca1d8fa2/DEPS

No update to Clang.

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

Change-Id: Ie9fcbf61d6b98369e505a7d654b38f1df4edb261
Reviewed-on: https://webrtc-review.googlesource.com/48980
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21925}
2018-02-07 03:45:28 +00:00
Autoroller
03e5ce84de Roll chromium_revision 5ab60ca7a8..242dbc9f7b (534678:534785)
Change log: 5ab60ca7a8..242dbc9f7b
Full diff: 5ab60ca7a8..242dbc9f7b

Changed dependencies:
* src/base: 9fa07591d3..c56936b597
* src/ios: fdc2a4c5f0..25470f3759
* src/testing: bfe88a29a9..0d35757b9e
* src/third_party: 9edc1a2220..afca687501
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a1303e4688..e6e84629db
* src/tools: c1a9e22c04..495b0aeb1c
DEPS diff: 5ab60ca7a8..242dbc9f7b/DEPS

No update to Clang.

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

Change-Id: I991260a042a29a253e15f7f327dd88612993708b
Reviewed-on: https://webrtc-review.googlesource.com/48862
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21924}
2018-02-06 21:52:54 +00:00
Steve Anton
dffead8835 Fail CreateAnswer if signaling state is not correct
This changes CreateAnswer to become compliant with the WebRTC 1.0
specification which details that createAnswer should fail if the
PeerConnection is in a state other than 'have-remote-offer' or
'have-local-pranswer'.

Bug: webrtc:8813
Change-Id: I7ca41bdebda1ea163aec8815267c1bbfd7d6d11e
Reviewed-on: https://webrtc-review.googlesource.com/47581
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21923}
2018-02-06 19:19:54 +00:00
Daniel Lazarenko
2870b0a57e Expose a link-local network interfaces enumeration option
The bug 8432 is caused by trying to connect through a
"link-local" interface (IP address 169.254.0.x/16),
which is listed among the iPhone network interfaces.
The bug is not happening if the link-local network interfaces
are skipped in the ICE candidate gethering process.

To control this behaviour an option - disable_link_local_networks -
is added inside the RTCConfiguration.
It is used to set the new BasicPortAllocatorSession flag -
PORTALLOCATOR_DISABLE_LINK_LOCAL_NETWORKS.
The port allocator flag is added if the configuration option is set.

IPIsLinkLocal IPAddress function and its friends (IPIsLoopback, IPIsPrivate)
are refactored to work on both IPv4 and IPv6.
Unit test IPIsLinkLocal.

Bonus: fix a bug in IPIsLinkLocalV6:
take into account just 10 network mask bits instead of 16.

Bug: webrtc:8432
Change-Id: Ibe8f677a36098057b7fcad5c798380727b23359b
Reviewed-on: https://webrtc-review.googlesource.com/36380
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Zhi Huang <zhihuang@webrtc.org>
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21922}
2018-02-06 19:12:04 +00:00
Steve Anton
5dfde18c77 Change PeerConnection stats interface to be more flexible
This removes the SessionStats object and replaces it with two
methods on PeerConnection: GetTransportNamesByMid and
GetTransportStatsByNames for use by the stats collectors. These
methods are more flexible and can cover cases where there are more
than one video/audio channel.

Bug: webrtc:8764
Change-Id: Id400cc548fc43675462ff6175a7fa9c9f4fd5948
Reviewed-on: https://webrtc-review.googlesource.com/47244
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21921}
2018-02-06 19:02:44 +00:00
Seth Hampson
8234ead6d9 Allows the application to set active simulcast streams.
Currently all simulcast streams are set as active by default. This
update takes the values of the rtp encoding parameters and wires those
values down to the VideoSendStream and VideoStreamEncoder, so that the
appropriate simulcast streams can be turned off and on. This includes
adding more application specific controls in the EncoderStreamFactory.

Bug: webrtc:8653
Change-Id: Iaa7da3209cea0f0db72543981a319e319705cb00
Reviewed-on: https://webrtc-review.googlesource.com/47245
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Peter Thatcher <pthatcher@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21920}
2018-02-06 18:51:14 +00:00
Jiawei Ou
c0216b8e68 Fix the iOS Framework dependency
`Foundation.framework` is not just for mac build, it is also needed on iOS build.

Bug: None
Change-Id: I94694102afbebbe60182521892e51c57760eb7c2
Reviewed-on: https://webrtc-review.googlesource.com/47656
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Jiawei Ou <ouj@fb.com>
Cr-Commit-Position: refs/heads/master@{#21919}
2018-02-06 17:56:54 +00:00
Seth Hampson
e2f69cfeef Reland "Updates tests for turning simulcast streams on/off."
This is a reland of 8fb22e71ee9bd77676838c5723f7e89a74a64aa9.

Original change's description:
> Updates tests for turning simulcast streams on/off.
>
> Due to libvpx we were restricted to always turning the low simulcast
> stream on, or else the encoder would always label the active streams'
> encoded frames as key frames. Now that libvpx has been updated and
> rolled in, this change updates tests to reflect that it is working.
>
> Bug: webrtc:8653
> Change-Id: I065ef817ace2292605e27e135802cf4a3e90647e
> Reviewed-on: https://webrtc-review.googlesource.com/46340
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Erik Språng <sprang@webrtc.org>
> Commit-Queue: Seth Hampson <shampson@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21831}

TBR=sprang@webrtc.org

Bug: webrtc:8653
Change-Id: I32fa92649f3ff40b1e364f880040e52ae698f74d
Reviewed-on: https://webrtc-review.googlesource.com/46860
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Seth Hampson <shampson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21918}
2018-02-06 17:18:24 +00:00
Per Åhgren
29f14322d1 Improved robustness and recovery speed in AEC3 during echo path changes
This CL adds robustness in terms of echo removal and faster recovery
in order to regain echo canceller transparency after echo path changes.

The CL does:
-Improve the adaptation rate of the linear filter.
-Increase the look-window used before the linear filter has adapted.
-Decrease the effects of missed detection of residual echo.
-Increase the safety margin before allowing the suppressor gain to
increase.

Bug: chromium:804873,webrtc:8788
Change-Id: I28eedc4c8d0a4f0bc7b79c02d6d59bf00fddd566
Reviewed-on: https://webrtc-review.googlesource.com/48721
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21917}
2018-02-06 15:07:54 +00:00
Mirko Bonadei
5c8622fa25 Removing backwards compatible rtc_event_log_impl target.
TBR=phoglund@webrtc.org

Bug: None
Change-Id: I035b54dd0acb390af69ff99dd12a37d0c7af802f
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/47383
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21916}
2018-02-06 13:54:13 +00:00
Autoroller
9f5abb6ad7 Roll chromium_revision f5b70e88bf..5ab60ca7a8 (534436:534678)
Change log: f5b70e88bf..5ab60ca7a8
Full diff: f5b70e88bf..5ab60ca7a8

Changed dependencies:
* src/base: 29406da0af..9fa07591d3
* src/build: c02da72816..7e86dc487b
* src/buildtools: f115f47867..2637e7e911
* src/ios: 241ed56a61..fdc2a4c5f0
* src/testing: 8f18207b60..bfe88a29a9
* src/third_party: 158d4d1e74..9edc1a2220
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b1069a4b3a..a1303e4688
* src/third_party/depot_tools: 3929e9ee94..e117e46a68
* src/third_party/googletest/src: 0062e4869f..ea31cb15f0
* src/third_party/icu: c8ca2962b4..d888fd2a1b
* src/tools: c74aa3f54b..c1a9e22c04
DEPS diff: f5b70e88bf..5ab60ca7a8/DEPS

No update to Clang.

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

Change-Id: I78eaea645b4b2a5081bf93c1c9691e6a96719d45
Reviewed-on: https://webrtc-review.googlesource.com/48600
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21915}
2018-02-06 13:44:13 +00:00
Mirko Bonadei
7fc0259160 check_includes=fase on audio_device_generic & objc_codec_factory_helper.
TBR=phoglund@webrtc.org

Bug: webrtc:8850
Change-Id: Iebc55a12d3a021aafe753778069ac8c90ccf4d3a
No-Try: True
No-Presubmit: true
Reviewed-on: https://webrtc-review.googlesource.com/48621
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21914}
2018-02-06 13:28:03 +00:00
Stefan Holmer
c6b224abc0 Pass the external fec_controller_ to VideoSendStream when available.
Bug: None
Change-Id: I179c81de2cb7da2a2742c3ebc333a1e0ea15bcc8
Reviewed-on: https://webrtc-review.googlesource.com/48522
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21913}
2018-02-06 13:21:52 +00:00
Gustaf Ullberg
f3a0e77e72 Make gustaf and peah OWNERS of api/audio
Bug: webrtc:8851
Notry: true
Change-Id: I543e1dfbb1707b7ea20a90dce0970e0b10889859
Reviewed-on: https://webrtc-review.googlesource.com/48461
Commit-Queue: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21912}
2018-02-06 13:11:12 +00:00