21217 Commits

Author SHA1 Message Date
Kári Tristan Helgason
0d3c9a3f2b Delete RTCAVFoundationVideoSource and related classes.
Bug: webrtc:8852
Change-Id: Ie073fe3f7bafc3d22fafef51f659e340d5a9250f
Reviewed-on: https://webrtc-review.googlesource.com/48620
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Anders Carlsson <andersc@webrtc.org>
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21985}
2018-02-12 14:41:25 +00:00
Mirko Bonadei
a55bdc2406 Fix 'gn gen' with target_os="win" on Linux.
This is the first step to enable Windows cross-compilation when the
host_os is Linux.

There are compilation errors ahead but at least it is possible to
generate ninja files.

Bug: webrtc:8875
Change-Id: I91a238bcb5e8f7670a6d19805e1ac032511fd46e
Reviewed-on: https://webrtc-review.googlesource.com/51821
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21984}
2018-02-12 14:29:35 +00:00
Patrik Höglund
afa1546153 Fix incorrect static method call.
cpuMonitor is actually null at the time of the call, but it works
because isSupported doesn't touch 'this' (being a static call).

Bug: None
Change-Id: I177807ee04075d16356878ec72262546d0547aa1
Reviewed-on: https://webrtc-review.googlesource.com/51861
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21983}
2018-02-12 14:24:55 +00:00
Edward Lemur
0c15a09293 Don't use gtest-parallel when running webrtc_perf_tests.
When we run webrtc_perf_tests with gtest-parallel, each test is run
individually, and this results in the file with the perf results being
overwritten each time.

To avoid this, we won't use gtest-parallel when running webrtc_perf_tests,
so we will simply run the binary directly.

TBR=phoglund@chromium.org

Bug: chromium:755660
Change-Id: I24db36e512fcf604a3de2adf4d0b4325b2c3d1ae
Reviewed-on: https://webrtc-review.googlesource.com/49340
Reviewed-by: Edward Lemur <ehmaldonado@webrtc.org>
Reviewed-by: Henrik Lundin <henrik.lundin@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21982}
2018-02-12 13:10:04 +00:00
Autoroller
88a38e32e7 Roll chromium_revision 6fcad09d4a..24887ed1ab (535596:535697)
Change log: 6fcad09d4a..24887ed1ab
Full diff: 6fcad09d4a..24887ed1ab

Changed dependencies:
* src/base: 83ec53d6ec..f1f7eec82e
* src/build: 03220ef360..c7a79acbd2
* src/testing: 7d3f4563c0..81576293e6
* src/third_party: efea12dde9..50ccc08052
* src/tools: ffaf09e5ff..4ee4992a32
DEPS diff: 6fcad09d4a..24887ed1ab/DEPS

Clang version changed 321529:324578
Details: 6fcad09d4a..24887ed1ab/tools/clang/scripts/update.py

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

Change-Id: I2c306df992e9b00c865207c28a56f79ab9bf195f
Reviewed-on: https://webrtc-review.googlesource.com/50360
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21981}
2018-02-12 10:46:35 +00:00
Niels Möller
cb768a8831 Delete unused code in videoengine_unittest.h.
Bug: None
Change-Id: Id59ac4da920b05b846dfcec973ea57365b0d3e81
Reviewed-on: https://webrtc-review.googlesource.com/49341
Reviewed-by: Magnus Flodman <mflodman@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21980}
2018-02-12 09:17:40 +00:00
Mirko Bonadei
b8a7d9d09f Updating usage of install-sysroot.py to stop relying on GYP_DEFINES.
The --running-as-hook flag has been removed in
https://chromium-review.googlesource.com/c/chromium/src/+/907673.

This CL mirrors the changes done in the Chromium src/DEPS file.

Bug: chromium:807986
Change-Id: Ib952eb0dbd8149e4f8bdfa2323cb8f23e1d63e0b
Reviewed-on: https://webrtc-review.googlesource.com/51760
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21979}
2018-02-12 08:37:00 +00:00
Per Åhgren
4b9124e432 Deactivated the computation of the reverb in AEC3
TBR: gustaf@webrtc.org
BUG: chromium:810951,webrtc:8872
Change-Id: I79194f964754d0f156a5206dbeb49606617e8bb5
Reviewed-on: https://webrtc-review.googlesource.com/50502
Commit-Queue: Per Åhgren <peah@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21978}
2018-02-10 00:35:11 +00:00
Steve Anton
8acdd1a1dc Parameterize PeerConnection signaling tests for Unified Plan
Bug: webrtc:8765
Change-Id: I50fbcfab66edb70c069b2dcb803c78b516e428c9
Reviewed-on: https://webrtc-review.googlesource.com/47582
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Commit-Queue: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21977}
2018-02-09 19:48:09 +00:00
Alex Leung
82d0817d6c Add callback when new audio data is ready
Bug: webrtc:8864
Change-Id: I476e9430da281f6815eb1af8ffd98afd9b664a63
Reviewed-on: https://webrtc-review.googlesource.com/49981
Commit-Queue: Alex Leung <alexleung@google.com>
Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
Reviewed-by: Ivo Creusen <ivoc@webrtc.org>
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21976}
2018-02-09 19:31:49 +00:00
Ying Wang
338bd62ab4 Includes the time(ms) that spent in network to test results.
Bug: None
Change-Id: I7af7055b924e3f68b0fd4ebb633190246275159f
Reviewed-on: https://webrtc-review.googlesource.com/50400
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21975}
2018-02-09 17:12:59 +00:00
Rasmus Brandt
2d40ad3f04 Remove dead code GetSimulcastSsrcs from simulcast.{cc,h}.
Bug: None
Change-Id: Ib1315bb65ebbf6c33008c9522451d9782119f5ca
Reviewed-on: https://webrtc-review.googlesource.com/47561
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21974}
2018-02-09 15:57:39 +00:00
Rasmus Brandt
75e38d2dc3 Remove unused fields from VideoCodecVP8.
Bug: None
Change-Id: I6f29ad5ce04582003e9be7292d04ea18f9335372
Reviewed-on: https://webrtc-review.googlesource.com/47660
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21973}
2018-02-09 15:55:59 +00:00
Rasmus Brandt
60bb6fe37a Protect VideoReceiveStream<->FlexfecReceiveStream sink association with unit test.
BUG=none

Change-Id: Id0c504f62d70febc5e846657dc2966f5e9acef39
Reviewed-on: https://webrtc-review.googlesource.com/17301
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21972}
2018-02-09 14:03:49 +00:00
Erik Språng
845a26214d Prevent potential integer overflow in sps parser
Bug: webrtc:8275, chromium:800698
Change-Id: I4dcba8ba480cd2a1b97dc09e97f585f2b3cf3279
Reviewed-on: https://webrtc-review.googlesource.com/40443
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Erik Språng <sprang@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21971}
2018-02-09 13:52:48 +00:00
Stefan Holmer
32e930fffa Add a freeze metric to the video quality test.
Defined as time between freezes (nothing rendered for 150 ms).

Bug: webrtc:8861
Change-Id: I56eae3beb7278b6d1894a0593ae3092c9f3cb1cc
Reviewed-on: https://webrtc-review.googlesource.com/49780
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21970}
2018-02-09 13:01:47 +00:00
Oleh Prypin
96f3ca13b0 Revert "Roll chromium_revision 6fcad09d4a..004f3b4b40 (535596:535645) + sysroot fix"
This reverts commit c9e6a592265dd45a02dcf7b34144733f1a886370.

Reason for revert: arm sysroot not being downloaded for linux arm32 build

Original change's description:
> Roll chromium_revision 6fcad09d4a..004f3b4b40 (535596:535645) + sysroot fix
> 
> Adapt DEPS hooks after the change to install-sysroot.py, the same way it's done in
> a07b9feb20
> 
> Change log: 6fcad09d4a..004f3b4b40
> Full diff: 6fcad09d4a..004f3b4b40
> 
> Changed dependencies:
> * src/base: 83ec53d6ec..782ae7073a
> * src/build: 03220ef360..c1972dd397
> * src/testing: 7d3f4563c0..81576293e6
> * src/third_party: efea12dde9..e3de125b3f
> * src/tools: ffaf09e5ff..54f1b52f74
> DEPS diff: 6fcad09d4a..004f3b4b40/DEPS
> 
> No update to Clang.
> 
> BUG=None
> CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal
> 
> No-Try: True
> Change-Id: I8026ac29cf127b863a2d60a36fdcdc7e6311aa45
> Reviewed-on: https://webrtc-review.googlesource.com/50183
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21968}

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

Change-Id: I7bd3ce6b93e4c5f1e4884dc2bfb5e8b54bac2876
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: None
Cq-Include-Trybots: master.internal.tryserver.corp.webrtc:linux_internal
Reviewed-on: https://webrtc-review.googlesource.com/50340
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21969}
2018-02-09 12:56:59 +00:00
Oleh Prypin
c9e6a59226 Roll chromium_revision 6fcad09d4a..004f3b4b40 (535596:535645) + sysroot fix
Adapt DEPS hooks after the change to install-sysroot.py, the same way it's done in
a07b9feb20

Change log: 6fcad09d4a..004f3b4b40
Full diff: 6fcad09d4a..004f3b4b40

Changed dependencies:
* src/base: 83ec53d6ec..782ae7073a
* src/build: 03220ef360..c1972dd397
* src/testing: 7d3f4563c0..81576293e6
* src/third_party: efea12dde9..e3de125b3f
* src/tools: ffaf09e5ff..54f1b52f74
DEPS diff: 6fcad09d4a..004f3b4b40/DEPS

No update to Clang.

BUG=None
CQ_INCLUDE_TRYBOTS=master.internal.tryserver.corp.webrtc:linux_internal

No-Try: True
Change-Id: I8026ac29cf127b863a2d60a36fdcdc7e6311aa45
Reviewed-on: https://webrtc-review.googlesource.com/50183
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21968}
2018-02-09 11:07:54 +00:00
Sami Kalliomäki
78498cf5ee Implements JavaToNativeStringMap and adds tests for native API.
Implements JavaToNativeStringMap that is a replacement for
JavaToStdMapStrings. It uses a new template method JavaToNativeMap. Also
adds testing support for native API and a test for JavaToNativeStringMap.

Bug: webrtc:8769
Change-Id: I580d4992a899ebe02da39af450fa51d52ee9b88b
Reviewed-on: https://webrtc-review.googlesource.com/48060
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21967}
2018-02-09 10:34:44 +00:00
Mirko Bonadei
1bc1ec43a3 Re-enabling libyuv 'gn check'.
Libyuv is now 'gn check' compatible and the fixed version has been
rolled into chromium (r1697).

Bug: webrtc:8850
Change-Id: Iaaeae229571fd02045322c4f8addadd75f889bdb
No-Try: True
Reviewed-on: https://webrtc-review.googlesource.com/50180
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21966}
2018-02-09 10:31:04 +00:00
Sergey Silkin
597472ebc3 Removing kNoVisualizationParams. Use nullptr directly.
Bug: none
Change-Id: Ifcffdf37d0dc4b9caa47b1117fc14e21bffe2cd9
Reviewed-on: https://webrtc-review.googlesource.com/49942
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21965}
2018-02-09 07:52:54 +00:00
Autoroller
ecfe2e86a6 Roll chromium_revision feadf7258e..6fcad09d4a (535428:535596)
Change log: feadf7258e..6fcad09d4a
Full diff: feadf7258e..6fcad09d4a

Changed dependencies:
* src/base: 69eb3646b6..83ec53d6ec
* src/build: 19190cb080..03220ef360
* src/ios: 075f6c8518..0959011404
* src/testing: 453c6a4ddb..7d3f4563c0
* src/third_party: 10c57e38e2..efea12dde9
* src/third_party/android_tools: https://chromium.googlesource.com/android_tools.git/+log/c9f9bbf0a6..9a70d48fcd
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/7e5dd25d47..61dedd6815
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/aa41a69e72..e653c4b823
* src/third_party/depot_tools: 6a60d7586a..71236c3af6
* src/third_party/libyuv: ffec313dbe..b792e0dbc1
* src/tools: 500944dbb2..ffaf09e5ff
DEPS diff: feadf7258e..6fcad09d4a/DEPS

No update to Clang.

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

Change-Id: I7b6d4ac17cbaed4c0736a9d0cb876ee4f5c3019c
Reviewed-on: https://webrtc-review.googlesource.com/50040
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21964}
2018-02-09 01:12:44 +00:00
Autoroller
d150534002 Roll chromium_revision 3d230c052f..feadf7258e (535313:535428)
Change log: 3d230c052f..feadf7258e
Full diff: 3d230c052f..feadf7258e

Changed dependencies:
* src/base: 05346abed5..69eb3646b6
* src/build: 3ba6ca6d32..19190cb080
* src/ios: 68272315d7..075f6c8518
* src/testing: 96fadc248d..453c6a4ddb
* src/third_party: cb732ebd07..10c57e38e2
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/b4a4bed9ad..aa41a69e72
* src/tools: 93539cf31f..500944dbb2
DEPS diff: 3d230c052f..feadf7258e/DEPS

No update to Clang.

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

Change-Id: I96a36c4826a743e08e43fe4d48af7b93450603d1
Reviewed-on: https://webrtc-review.googlesource.com/49920
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21963}
2018-02-09 00:01:44 +00:00
Edward Lemur
2b67f5c65f MB: Add a way to run tests on swarming without using gtest-parallel.
Adds a new test_type 'raw' to run tests on swarming without wrapping it
on gtest-parallel.

This will be used to run webrtc_perf_tests directly.

Bug: chromium:755660
Change-Id: I8558faadf242d1db1ad3e13083941886c92b1bd9
Reviewed-on: https://webrtc-review.googlesource.com/49360
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Edward Lemur <ehmaldonado@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21962}
2018-02-08 19:37:19 +00:00
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