Commit Graph

22531 Commits

Author SHA1 Message Date
c7da266cb8 Android JNI generation: Set JNI namespace in build files
This CL removes the use of the @JNINamespace annotation and instead
sets the correct JNI namespace in the build file.

Bug: webrtc:8278
Change-Id: Ia4490399e45a97d56b02c260fd80df4edfa092bf
Reviewed-on: https://webrtc-review.googlesource.com/76440
Commit-Queue: Magnus Jedvert <magjed@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23299}
2018-05-18 10:44:38 +00:00
7e6fcea7de Ports CoreAudioUtil from Chrome to WebRTC.
See https://cs.chromium.org/chromium/src/media/audio/win/core_audio_util_win.h?q=coreaudio&sq=package:chromium&g=0&l=34
for details.

Bug: webrtc:9265
Change-Id: I0fd26620d94a81ccced68d81021c39723a5be2cb
Reviewed-on: https://webrtc-review.googlesource.com/76900
Reviewed-by: Oskar Sundbom <ossu@webrtc.org>
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23298}
2018-05-18 10:25:26 +00:00
7eb8e9fd7b Add RegisterExternalDecoder in VideoCodingModule.
In preparation for landing https://webrtc-review.googlesource.com/c/src/+/72441
a downstream project that uses the VideoCodingModule needs to be able to
inject a decoder object created from the outside, just like how encoders
are possible to inject.

Bug: webrtc:7925
Change-Id: Ibaeffda55f84410436d79f75730e7352e298b9f0
Reviewed-on: https://webrtc-review.googlesource.com/77160
Commit-Queue: Anders Carlsson <andersc@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23297}
2018-05-18 09:43:26 +00:00
460f53bb86 Roll chromium_revision c92ed25217..bcf2616e8e (559015:559838)
Change log: c92ed25217..bcf2616e8e
Full diff: c92ed25217..bcf2616e8e

Roll chromium third_party 51c08cf9af..9d65a3cdda
Change log: 51c08cf9af..9d65a3cdda

Changed dependencies:
* src/base: a7a2409f9b..b802985ef4
* src/build: 03f39fd800..fc8308f6b6
* src/buildtools: a9e946f166..94288c26d2
* src/ios: e070a93062..289c450460
* src/testing: f5b31b58c6..a5fce03148
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/d8600ccc2d..ce9b3742a1
* src/third_party/depot_tools: 8de3800ce5..8fe4d8cbef
* src/third_party/googletest/src: 045e7f9ee4..08d5b1f33a
* src/tools: e024720629..6e6e398687
* src/tools/swarming_client: 88229872dd..833f5ebf89
DEPS diff: c92ed25217..bcf2616e8e/DEPS

No update to Clang.

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

Change-Id: I22bf301fcec0103a1987a92f95ebf86e324dade7
Reviewed-on: https://webrtc-review.googlesource.com/77625
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23296}
2018-05-18 09:37:26 +00:00
c948fe62fd Delete unneeded includes of call/video_config.h.
Bug: webrtc:8830
Change-Id: I6114b47e5524a6d2450108388236478b1ceafb67
Reviewed-on: https://webrtc-review.googlesource.com/77425
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23295}
2018-05-18 09:00:56 +00:00
4639d46eaa Add vpython dependencies needed to run presubmit tests on LUCI
E.g. tools_webrtc/libs/generate_licenses_test.py needs 'mock', which needs the other libs added here.

This is taken from 641bce4223/.vpython (29)

TBR: kwiberg@webrtc.org
No-Try: True
Bug: chromium:749664
Change-Id: I180fc1190d2664c5a82f63857b740334145e7daa
Reviewed-on: https://webrtc-review.googlesource.com/77520
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23294}
2018-05-18 08:10:25 +00:00
638edfc88c Skipping some Opus tests to let the new roll flow.
In order to roll the new version of Opus in WebRTC, this CL disables
some tests that will fail because of [1].

They will be re-enabled and fixed as soon as the new Opus revision is
rolled.

[1] - https://chromium-review.googlesource.com/1061499

TBR=henrik.lundin@webrtc.org

Bug: webrtc:9280
Change-Id: I84870ced66d554f75c2d093dac8103ad7860cae5
Reviewed-on: https://webrtc-review.googlesource.com/77640
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23293}
2018-05-18 07:58:46 +00:00
4c8811b255 Delete some obsolete forward declarations
Bug: None
Change-Id: I3a9b59bf3dd63c206854ab949cf2d606046182c9
Reviewed-on: https://webrtc-review.googlesource.com/77427
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23292}
2018-05-18 07:29:25 +00:00
dd3eae5f94 Revert "Configure and use max bitrate to limit the AIMD controller estimates."
This reverts commit 18d7c7ea7e56444d6d7e6c8fb95b5f426fd7b953.

Reason for revert: 
This seems to cause the auto roller to Chrome to fail on Linux and Mac on the browsertest
WebRtcSimulcastBrowserTest.TestVgaReturnsTwoSimulcastStreams

https://chromium-review.googlesource.com/c/chromium/src/+/1064736


Original change's description:
> Configure and use max bitrate to limit the AIMD controller estimates.
> 
> Bug: webrtc:9275
> Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
> Reviewed-on: https://webrtc-review.googlesource.com/77081
> Reviewed-by: Stefan Holmer <stefan@webrtc.org>
> Reviewed-by: Sebastian Jansson <srte@webrtc.org>
> Commit-Queue: Björn Terelius <terelius@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23287}

TBR=terelius@webrtc.org,stefan@webrtc.org,srte@webrtc.org

Change-Id: I8ed827ab6b2f7d2b70b9889e5a88701bfb974d35
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9275
Reviewed-on: https://webrtc-review.googlesource.com/77660
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23291}
2018-05-18 07:12:26 +00:00
0ab95b97c8 Remove all baremetal bots, baremetal tests will be part of *_rel
See https://chromium-review.googlesource.com/c/chromium/tools/build/+/1058799
After the baremetal machines are moved to swarming, that CL will add the tasks that were previously in *_baremetal bots to *_rel bots. *_rel bots will run all the same tasks as before on a pool of generic machines, but also run a few tests on dedicated baremetal machines (pool:WebRTC-baremetal-try).

No-Try: True
Bug: chromium:755660
Change-Id: I99d62a84aac631b1c127bf661546baecb2a3ae9a
Reviewed-on: https://webrtc-review.googlesource.com/76721
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23290}
2018-05-18 07:03:32 +00:00
90394a4db3 Reland "[desktopCapture] Unify the position info in DIP coordinates on Mac."
This is a reland of 89653d5db46419d2a80898635cb27fed64898db2

Original change's description:
> [desktopCapture] Unify the position info in DIP coordinates on Mac.
> 
> On OSX, the logical(DIP) and physical coordinates are used mixingly.
> For example, the captured image has its size in physical pixels(2x) and
> location in logical(DIP) pixels. Same to the cursor position. This
> causes trouble when we check the relative position of image and cursor
> when there are multiple monitors with different DIP setting connected.
> 
> This cl proposed a solution to use DIP pixel for any location info,
> i.e. top-left of a frame and cursor position. Also propose a method to
> get the current scale factor of a window across multiple monitors. And
> save the current scale factor in DPI of the capture frame.
> Then we can check relative position of cursor and frame correctly
> in DIP pixel and compose them in physical pixel.
> 
> Bug: webrtc:9178
> Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
> Reviewed-on: https://webrtc-review.googlesource.com/71621
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Reviewed-by: Zijie He <zijiehe@chromium.org>
> Commit-Queue: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23263}

Bug: webrtc:9178
Change-Id: I97d9150f7b9a4ed6671733b75613ea9c315d5c1d
Reviewed-on: https://webrtc-review.googlesource.com/77481
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23289}
2018-05-17 18:45:42 +00:00
0643fd66ce Add VP9 K-SVC performance tests.
This adds full stack performance tests for K-SVC mode.

Bug: none
Change-Id: I8e8a6ff72f1b1aed7cb74781b0d7cc645aa12340
Reviewed-on: https://webrtc-review.googlesource.com/77400
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Commit-Queue: Sergey Silkin <ssilkin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23288}
2018-05-17 17:00:42 +00:00
18d7c7ea7e Configure and use max bitrate to limit the AIMD controller estimates.
Bug: webrtc:9275
Change-Id: I9625cd473e1cb198abe08020f5462f1bd64bf2a5
Reviewed-on: https://webrtc-review.googlesource.com/77081
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Sebastian Jansson <srte@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23287}
2018-05-17 16:07:42 +00:00
09ec3cf7c6 Fix logic for camera check. Still run the test if webcam is found
TBR: phoglund@webrtc.org
No-Try: True
Bug: chromium:755660
Change-Id: Iea1ae4a599a644aca18181f68fe155a659163766
Reviewed-on: https://webrtc-review.googlesource.com/77426
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23286}
2018-05-17 14:26:01 +00:00
f8d8d6d00c Use range-based-for instead of std::for_each and std::mem_fun
std::mem_fun is deprecated in C++11, and removed in C++17. Using C++17
option for building libwebrtc causes build failure. This is found during
upgrading WebKit tree from C++14 to C++17.
This patch replaces std::for_each and std::mem_fun with range-based-for.
We also merge loops for streams_ into one.

Bug: webrtc:9277
Change-Id: I44a7e44ea21fc33ffa9a586ddfea570f97dfacb6
Reviewed-on: https://webrtc-review.googlesource.com/77280
Reviewed-by: Niels Moller <nisse@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23285}
2018-05-17 13:51:02 +00:00
c475ac14a9 Update Android native API example to use real camera.
For simplicity, camera with index 0 is used. User also has to manually
give the permission to use the camera for the app.

Bug: webrtc:8769
Change-Id: I371f26f94d629411fd299671b4f3202e84556b80
Reviewed-on: https://webrtc-review.googlesource.com/76982
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23284}
2018-05-17 13:31:31 +00:00
09133af36f Check number of nalus in packet before checking nalu types.
Bug: chromium:840536
Change-Id: Ia4dcf322ad6290691fd01b58fb02cd868714c92e
Reviewed-on: https://webrtc-review.googlesource.com/77121
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23283}
2018-05-17 12:52:11 +00:00
1e9cf7faf8 Fuzzing for video_coding::FrameBuffer2.
Bug: webrtc:7728
Change-Id: I712289a82d408dde1db73a1cc44f0c69a6b639ff
Reviewed-on: https://webrtc-review.googlesource.com/31841
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23282}
2018-05-17 12:41:41 +00:00
8b7ca4abb2 Make packet router send padding on rtp module that last sent media.
Currently we prefer the last added rtp module that supports rtx, and
assume this is the HD stream.

If we suffer a network degradation and stop sending HD, the current
behavior will trigger RTX padding on an inactive stream, which is not
very useful.

With this change, we will prefer the rtp module that last sent media,
which will spread the load a bit across active media streams, but will
be biased toward the one with highest packet rate.

Bug: webrtc:8975
Change-Id: Id52865ccd5263722c66d327b8c80457f63b90385
Reviewed-on: https://webrtc-review.googlesource.com/77360
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23281}
2018-05-17 12:21:11 +00:00
e1c3c01a90 Revert "[desktopCapture] Unify the position info in DIP coordinates on Mac."
This reverts commit 89653d5db46419d2a80898635cb27fed64898db2.

Reason for revert: 

Tentatively revert since I believe this break remoting unittests on Asan/Tsan

https://chromium-review.googlesource.com/c/chromium/src/+/1063330

https://chromium-swarm.appspot.com/task?id=3d8692bedcc85c10&refresh=10&show_raw=1


Original change's description:
> [desktopCapture] Unify the position info in DIP coordinates on Mac.
> 
> On OSX, the logical(DIP) and physical coordinates are used mixingly.
> For example, the captured image has its size in physical pixels(2x) and
> location in logical(DIP) pixels. Same to the cursor position. This
> causes trouble when we check the relative position of image and cursor
> when there are multiple monitors with different DIP setting connected.
> 
> This cl proposed a solution to use DIP pixel for any location info,
> i.e. top-left of a frame and cursor position. Also propose a method to
> get the current scale factor of a window across multiple monitors. And
> save the current scale factor in DPI of the capture frame.
> Then we can check relative position of cursor and frame correctly
> in DIP pixel and compose them in physical pixel.
> 
> Bug: webrtc:9178
> Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
> Reviewed-on: https://webrtc-review.googlesource.com/71621
> Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
> Reviewed-by: Zijie He <zijiehe@chromium.org>
> Commit-Queue: Brave Yao <braveyao@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23263}

TBR=zijiehe@chromium.org,jamiewalch@chromium.org,perkj@webrtc.org,braveyao@webrtc.org

Change-Id: Ica02365925623e21b256d20a21b5625e7ed6f49b
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9178
Reviewed-on: https://webrtc-review.googlesource.com/77461
Reviewed-by: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23280}
2018-05-17 12:11:22 +00:00
739b8169ef Reland "Add support for launching webcam software for use in recipes"
This is a reland of cd469a4ce5315fee62bb77b5e781a18a99da506d
With vpython specified

Original change's description:
> Add support for launching webcam software for use in recipes
>
> * Copy ensure_webcam_is_running.py from recipes repo
> * Turn it into a wrapper that can launch another script
>   (fix_python_path is copied from test_env.py as _ForcePythonInterpreter)
> * Support it in mb.py
> * Add it to video_capture_unittests
>
> No-Try: True
> Bug: chromium:755660
> Change-Id: I376724a77e443620724add7818592e9368d02079
> Reviewed-on: https://webrtc-review.googlesource.com/77320
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23275}

No-Try: True
Bug: chromium:755660
Change-Id: Ibf4fbe3fea4b003a23b12332b3a6078a2db99a8d
Reviewed-on: https://webrtc-review.googlesource.com/77460
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23279}
2018-05-17 11:42:21 +00:00
e62e18fe13 Revert "Add support for launching webcam software for use in recipes"
This reverts commit cd469a4ce5315fee62bb77b5e781a18a99da506d.

Reason for revert: ImportError: No module named psutil

Original change's description:
> Add support for launching webcam software for use in recipes
> 
> * Copy ensure_webcam_is_running.py from recipes repo
> * Turn it into a wrapper that can launch another script
>   (fix_python_path is copied from test_env.py as _ForcePythonInterpreter)
> * Support it in mb.py
> * Add it to video_capture_unittests
> 
> No-Try: True
> Bug: chromium:755660
> Change-Id: I376724a77e443620724add7818592e9368d02079
> Reviewed-on: https://webrtc-review.googlesource.com/77320
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23275}

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

Change-Id: I611e49e1c8d8668fe405e462ec4b83e364c3d6e7
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: chromium:755660
Reviewed-on: https://webrtc-review.googlesource.com/77440
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23278}
2018-05-17 11:24:01 +00:00
eeaa8f929c Directly include VideoBitrateAllocation in modules/rtp_rtcp/ targets
Bug: webrtc:9271
Change-Id: Ic7415830588bef9d87bab92943460207890dada6
Reviewed-on: https://webrtc-review.googlesource.com/76960
Commit-Queue: Erik Språng <sprang@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23277}
2018-05-17 11:22:56 +00:00
49fcc10de6 Merge DegradationPreference enums.
This replaces webrtc::VideoSendStream::DegradationPreference with
webrtc::DegradationPreference, and adds "DISABLED".

It's still not wired up from RtpSenderInterface::SetParameters to the
underlying video engine; that would be the next step.

Bug: webrtc:8830
Change-Id: I582ffd04eaef33c73d9892e52e789804c933b864
Reviewed-on: https://webrtc-review.googlesource.com/77024
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23276}
2018-05-17 11:21:52 +00:00
cd469a4ce5 Add support for launching webcam software for use in recipes
* Copy ensure_webcam_is_running.py from recipes repo
* Turn it into a wrapper that can launch another script
  (fix_python_path is copied from test_env.py as _ForcePythonInterpreter)
* Support it in mb.py
* Add it to video_capture_unittests

No-Try: True
Bug: chromium:755660
Change-Id: I376724a77e443620724add7818592e9368d02079
Reviewed-on: https://webrtc-review.googlesource.com/77320
Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23275}
2018-05-17 11:13:01 +00:00
d5addcaf1e Add philipel to modules/video_coding/OWNERS.
Bug: None
Change-Id: I982ac84a58aff27a6f8f9cf50005ac732f540785
Reviewed-on: https://webrtc-review.googlesource.com/77362
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23274}
2018-05-17 10:14:31 +00:00
8eeda499f7 Enable more VideoCodecTests on iOS.
Bug: webrtc:4755
Change-Id: I403834dbe04cc3899847eb10e5595a24e6001507
Reviewed-on: https://webrtc-review.googlesource.com/76602
Commit-Queue: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23273}
2018-05-17 09:13:31 +00:00
bdb504e492 Add sakal@webrtc.org to OWNERS for Android native API example app.
Bug: None
Change-Id: I6a409c561ddd95f586ee5f9338aeeeb21465f369
Reviewed-on: https://webrtc-review.googlesource.com/77340
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23272}
2018-05-17 09:05:21 +00:00
c1ee9d57d1 RtpFrameReferenceFinder fuzzer.
Bug: webrtc:7728
Change-Id: I641772837384a4d8070db2138b93f4157f997d03
Reviewed-on: https://webrtc-review.googlesource.com/74584
Commit-Queue: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23271}
2018-05-17 08:55:31 +00:00
c47c9c0234 Change structure of deps file and tool for adding chromium dep.
Change structure of chromium owned dependencies file to JSON to
simplify work with it in different tools. Also add tool to check in
new chromium owned dep with single command like this:
./tools_webrtc/autoroller/checkin_chromium_dep.py -d <dep name>

Introduce separate file with list of webrtc owned dependencies.



Bug: webrtc:8366
Change-Id: I30a828af34cd105ce7e6bc76d6b5889e6bf7574d
Reviewed-on: https://webrtc-review.googlesource.com/76840
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Artem Titov <titovartem@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23270}
2018-05-17 08:52:31 +00:00
ff1de0af6b Add Android native API: CreateJavaVideoSource
Adds Android native API for creating VideoTrackSourceInterface objects
that can be fed frames using VideoCapturer.CapturerObserver.

NativeCapturerObserver is moved out of VideoSource because it will now
be used without a VideoSource. It now takes a pointer to
AndroidVideoTrackSource directly instead of VideoTrackSourceProxy.

VideoSource and NativeCapturerObserver JNI code is moved away from
androidvideotracksource.cc to their own files. This allows using
AndroidVideoTrackSource independently.

Bug: webrtc:8769
Change-Id: Ifb9e1eb27d4c8237597d19d932ca6e863abb4d27
Reviewed-on: https://webrtc-review.googlesource.com/76924
Reviewed-by: Paulina Hensman <phensman@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23269}
2018-05-17 07:41:51 +00:00
8e7a62beb2 Delete unused include of fakevideocapturer.h.
Bug: webrtc:6353
Change-Id: I007320e821e44bbd93776ff76d76e550a7f94602
Reviewed-on: https://webrtc-review.googlesource.com/76922
Commit-Queue: Niels Moller <nisse@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23268}
2018-05-17 07:23:01 +00:00
502db3df4e Replace MessageHandler by AsyncInvoker in P2PTransportChannel.
The existing asynchronous task execution in P2PTransportChannel is
implemented by posting messages to its network thread (a rtc::Thread)
and consuming these messages as a MessageHandler. The readability of
the implementation can be improved by using AsyncInvoker, which is
exactly designed for this scenario.

Bug: None
Change-Id: Ibee830d0d2bc19fc1ca5b894f194d9b69c40eef4
Reviewed-on: https://webrtc-review.googlesource.com/74642
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23267}
2018-05-17 01:05:31 +00:00
10a0e516bf Improve text logs in the network stack.
1) Network info is appended with its network ID assigned by the network
manager so that we can cross-reference networks by IDs in the log.
2) The local network info is added to the candidate pair string
representation so that we do not need the cross reference to the
logs of candidate gathering to find out the network where the local
candidate is from.
3) A flag is added to the candidate pair string representation to
indicate if this pair is the selected one.
4) Sorting of candidate pairs is logged with the reason of sorting
request.
5) Network filtering that takes place in the port allocator is
explicitly logged.

Bug: None
Change-Id: Iaa337394cad803515e26e254814aa04ed2213eab
Reviewed-on: https://webrtc-review.googlesource.com/72522
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23266}
2018-05-16 23:46:22 +00:00
90e3fbdd37 Activating the AEC3 audibility improvements functionality
This CL turns on the previously implemented AEC3 audibility
improvements, which before has been off by default.

Bug: webrtc:9193,chromium:836790
Change-Id: Ibcd057ba5dd002718d62fd83db33d01d9563b8ea
Reviewed-on: https://webrtc-review.googlesource.com/77123
Reviewed-by: Gustaf Ullberg <gustaf@webrtc.org>
Reviewed-by: Jesus de Vicente Pena <devicentepena@webrtc.org>
Commit-Queue: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23265}
2018-05-16 16:47:16 +00:00
85cb19fec7 Revert "Reland "Adding absl includes and defines to rtc_* templates.""
This reverts commit 9632112a16d70a146e917db4de761e6253dfc364.

Reason for revert: It breaks the WebRTC roll into Chromium.
https://chromium-review.googlesource.com/c/chromium/src/+/1061476

Original change's description:
> Reland "Adding absl includes and defines to rtc_* templates."
> 
> This reverts commit d161eda477491b2b97fb3f26d229c625a2a0e9b8.
> 
> Reason for revert: The problem with iOS trybots should be fixed.
> 
> Original change's description:
> > Revert "Adding absl includes and defines to rtc_* templates."
> >
> > This reverts commit 9d8f3850f4c4faad5dc5ab32ab6f2c9c43df7b6c.
> >
> > Reason for revert: Breaks some trybots: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Release/builds/12793.
> >
> > Original change's description:
> > > Adding absl includes and defines to rtc_* templates.
> > >
> > > This CL implicitly adds the -I compiler flag and absl macros to WebRTC
> > > templates. In order to include absl headers using relative paths, WebRTC
> > > needs to ensure that all its build targets are able to see absl headers.
> > >
> > > This can also be done with public_deps, but WebRTC is trying to avoid
> > > it because it creates problems with other build systems. Given this
> > > constraint, using rtc_* templates is the most reliable solution.
> > >
> > > Please note that rtc_* templates are adding absl includes and defines
> > > as public_configs, this means that build targets with WebRTC targets
> > > in their public_deps will propagate these configs following the GN
> > > guideline.
> > >
> > > Bug: webrtc:8821
> > > Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
> > > Reviewed-on: https://webrtc-review.googlesource.com/70367
> > > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > > Cr-Commit-Position: refs/heads/master@{#22927}
> >
> > TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> >
> > Change-Id: Id8e1f881c57553386566eb1970f6b9f8632cab37
> > No-Presubmit: true
> > No-Tree-Checks: true
> > No-Try: true
> > Bug: webrtc:8821
> > Reviewed-on: https://webrtc-review.googlesource.com/71000
> > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22928}
> 
> TBR=phoglund@webrtc.org,mbonadei@webrtc.org
> 
> Bug: webrtc:8821
> Change-Id: I6ee2eda97bbcd4c9be25c9c4073272192b0373f8
> Reviewed-on: https://webrtc-review.googlesource.com/71700
> Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23251}

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

Change-Id: I61fb749797314ca514691b341c66f7f39ef45491
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:8821
Reviewed-on: https://webrtc-review.googlesource.com/77220
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23264}
2018-05-16 16:46:12 +00:00
89653d5db4 [desktopCapture] Unify the position info in DIP coordinates on Mac.
On OSX, the logical(DIP) and physical coordinates are used mixingly.
For example, the captured image has its size in physical pixels(2x) and
location in logical(DIP) pixels. Same to the cursor position. This
causes trouble when we check the relative position of image and cursor
when there are multiple monitors with different DIP setting connected.

This cl proposed a solution to use DIP pixel for any location info,
i.e. top-left of a frame and cursor position. Also propose a method to
get the current scale factor of a window across multiple monitors. And
save the current scale factor in DPI of the capture frame.
Then we can check relative position of cursor and frame correctly
in DIP pixel and compose them in physical pixel.

Bug: webrtc:9178
Change-Id: I3c076aeac2d6f2c1f63d000d7fff03500aa375ac
Reviewed-on: https://webrtc-review.googlesource.com/71621
Reviewed-by: Jamie Walch <jamiewalch@chromium.org>
Reviewed-by: Zijie He <zijiehe@chromium.org>
Commit-Queue: Brave Yao <braveyao@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23263}
2018-05-16 16:44:01 +00:00
a8bf169906 Adds BandwidthSampler for BBR.
This prepares for making the BBR implementation more identical to the
implementation in Quic, this is to ensure that results are comparable.

Bug: webrtc:8415
Change-Id: Ic2dc4394dc9923e5109ffa5f146c23b527f0c395
Reviewed-on: https://webrtc-review.googlesource.com/76582
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23262}
2018-05-16 16:11:21 +00:00
e7659df12f Suppressing /wd4702.
Bug: webrtc:9274
Change-Id: I420a96202d9de323341ef7831bbf90f986958003
Reviewed-on: https://webrtc-review.googlesource.com/77040
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23261}
2018-05-16 10:59:13 +00:00
fe826f33f3 Removing warning suppression flags from media/BUILD.gn.
Bug: webrtc:9251
Change-Id: I52e1d06311fd973956ca410540ad99f5fb792cda
Reviewed-on: https://webrtc-review.googlesource.com/77041
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23260}
2018-05-16 10:07:32 +00:00
81ae9c850e Removing clang-cl warning suppression flags in rtc_base.
Bug: webrtc:9251
Change-Id: I06112d1d683a65b1555b5201deb236928dd8d196
Reviewed-on: https://webrtc-review.googlesource.com/77002
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23259}
2018-05-16 10:00:22 +00:00
74642bafef Roll chromium_revision 06a3103dba..c92ed25217 (558901:559015)
Change log: 06a3103dba..c92ed25217
Full diff: 06a3103dba..c92ed25217

Roll chromium third_party 07a7c038b7..51c08cf9af
Change log: 07a7c038b7..51c08cf9af

Changed dependencies:
* src/base: da654fb628..a7a2409f9b
* src/build: c9de5e4fe8..03f39fd800
* src/ios: 83ea3fdb9e..e070a93062
* src/testing: 0922117f9b..f5b31b58c6
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a8692f3255..d8600ccc2d
* src/third_party/depot_tools: cbf0204555..8de3800ce5
* src/third_party/libvpx/source/libvpx: 28801f91c4..d99abe9a9a
* src/tools: 279b289fcd..e024720629
DEPS diff: 06a3103dba..c92ed25217/DEPS

Clang version changed 331747:332335
Details: 06a3103dba..c92ed25217/tools/clang/scripts/update.py

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

Change-Id: I25b07871a270ab58ff1111a9061cdcc84e5b8891
Reviewed-on: https://webrtc-review.googlesource.com/77060
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23258}
2018-05-16 09:16:22 +00:00
8dd4db49e2 Create/destroying codecs on task queue + switch to TaskQueueForTest.
After https://webrtc-review.googlesource.com/c/src/+/70740, we are
creating/destroying the codecs on a task queue in the VideoStreamEncoder. This
CL updates the VideoCodecTest to do the same.

Also, this CL switches from manually Wait()'ing on the task queue to using
TaskQueueForTest::SendTask.

Bug: None
Change-Id: Ia0398b24e32e9cc5361ba5ee4c08441116def18e
Reviewed-on: https://webrtc-review.googlesource.com/76800
Commit-Queue: Rasmus Brandt <brandtr@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23257}
2018-05-16 08:15:23 +00:00
59130a11b8 Delete deprecated version of VideoCodecInitializer::SetupCodec.
A followup to https://webrtc-review.googlesource.com/71380 and
https://webrtc-review.googlesource.com/69986, deleting the
nack_enabled flag.

Bug: webrtc:8830
Change-Id: Ie53f7a1e131be5347936ed44cee7167295026d57
Reviewed-on: https://webrtc-review.googlesource.com/76760
Reviewed-by: Rasmus Brandt <brandtr@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23256}
2018-05-16 08:03:24 +00:00
e9c2088e40 Delete unused header file unittest_utils.h.
Became unused with cl https://webrtc-review.googlesource.com/40740,
and last spurious reference dropped in cl
https://webrtc-review.googlesource.com/43360.

Bug: None
Change-Id: Ib5f27a437c1ae8f7cc2df84d6d9eda8c297481d6
Reviewed-on: https://webrtc-review.googlesource.com/76981
Reviewed-by: Sergey Silkin <ssilkin@webrtc.org>
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Commit-Queue: Niels Moller <nisse@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23255}
2018-05-16 07:26:42 +00:00
5f12f09492 Roll chromium_revision 5e8d8448f9..06a3103dba (558688:558901)
Change log: 5e8d8448f9..06a3103dba
Full diff: 5e8d8448f9..06a3103dba

Roll chromium third_party 156f0a4ab7..07a7c038b7
Change log: 156f0a4ab7..07a7c038b7

Changed dependencies:
* src/base: fa869fa59b..da654fb628
* src/build: f7d99e7a2f..c9de5e4fe8
* src/ios: dab8567b66..83ea3fdb9e
* src/testing: bfb20b1a58..0922117f9b
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/075ca3000a..a8692f3255
* src/third_party/depot_tools: babd098f36..cbf0204555
* src/tools: c708703ecd..279b289fcd
DEPS diff: 5e8d8448f9..06a3103dba/DEPS

No update to Clang.

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

Change-Id: I74371c63ffa8af43bd2cc8b47192a79362890cf1
Reviewed-on: https://webrtc-review.googlesource.com/77023
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23254}
2018-05-16 01:58:02 +00:00
4f81038a52 Revert "Injectable logging"
This reverts commit 59216ec4a4151b1ba5478c8f2b5c9f01f4683d7f.

Reason for revert:  forces all logs to have identical tag

Original change's description:
> Injectable logging
> 
> Allows passing a Loggable to PCFactory.initializationOptions, which
> is then injected to Logging.java and logging.h. Future log messages
> in both Java and native will then be passed to this Loggable.
> 
> Bug: webrtc:9225
> Change-Id: I2ff693380639448301a78a93dc11d3a0106f0967
> Reviewed-on: https://webrtc-review.googlesource.com/73243
> Commit-Queue: Paulina Hensman <phensman@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23241}

TBR=magjed@webrtc.org,sakal@webrtc.org,kwiberg@webrtc.org,phensman@webrtc.org

Change-Id: I27c9587238325b69b26166434740869021b7db8a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9225
Reviewed-on: https://webrtc-review.googlesource.com/76885
Reviewed-by: Alex Glaznev <glaznev@webrtc.org>
Commit-Queue: Alex Glaznev <glaznev@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23253}
2018-05-15 22:33:53 +00:00
d7624f2d90 Removing clang-cl warning suppression flags in video_capture.
After landing https://webrtc-review.googlesource.com/76720 and
https://webrtc-review.googlesource.com/76722 it is possible to remove
clang-cl only warning suppression flags.

These were needed because winsdk_samples was included using include_dirs
(-I) instead of -imsvc (which is similar to -isystem in GCC and clang).

Bug: webrtc:9251
Change-Id: I2bc19f0ba552342c49a29c07e4fb84d439003dfb
Reviewed-on: https://webrtc-review.googlesource.com/76781
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23252}
2018-05-15 17:33:32 +00:00
9632112a16 Reland "Adding absl includes and defines to rtc_* templates."
This reverts commit d161eda477491b2b97fb3f26d229c625a2a0e9b8.

Reason for revert: The problem with iOS trybots should be fixed.

Original change's description:
> Revert "Adding absl includes and defines to rtc_* templates."
>
> This reverts commit 9d8f3850f4c4faad5dc5ab32ab6f2c9c43df7b6c.
>
> Reason for revert: Breaks some trybots: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Release/builds/12793.
>
> Original change's description:
> > Adding absl includes and defines to rtc_* templates.
> >
> > This CL implicitly adds the -I compiler flag and absl macros to WebRTC
> > templates. In order to include absl headers using relative paths, WebRTC
> > needs to ensure that all its build targets are able to see absl headers.
> >
> > This can also be done with public_deps, but WebRTC is trying to avoid
> > it because it creates problems with other build systems. Given this
> > constraint, using rtc_* templates is the most reliable solution.
> >
> > Please note that rtc_* templates are adding absl includes and defines
> > as public_configs, this means that build targets with WebRTC targets
> > in their public_deps will propagate these configs following the GN
> > guideline.
> >
> > Bug: webrtc:8821
> > Change-Id: I4aa594a524f4bd045bcb3e80d76cc27f06fe01d7
> > Reviewed-on: https://webrtc-review.googlesource.com/70367
> > Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> > Cr-Commit-Position: refs/heads/master@{#22927}
>
> TBR=phoglund@webrtc.org,mbonadei@webrtc.org
>
> Change-Id: Id8e1f881c57553386566eb1970f6b9f8632cab37
> No-Presubmit: true
> No-Tree-Checks: true
> No-Try: true
> Bug: webrtc:8821
> Reviewed-on: https://webrtc-review.googlesource.com/71000
> Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
> Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#22928}

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

Bug: webrtc:8821
Change-Id: I6ee2eda97bbcd4c9be25c9c4073272192b0373f8
Reviewed-on: https://webrtc-review.googlesource.com/71700
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23251}
2018-05-15 17:30:33 +00:00
b7d9d8346f Implement RtpCodecParameters::parameters
This will return all the fmtp parameters for the codecs, except for
DTMF codes that don't fit the key=value pattern.

Bug: webrtc:7112
Change-Id: I06a203ff64df2c3bc9bc2082cd0f374718b23510
Reviewed-on: https://webrtc-review.googlesource.com/71801
Commit-Queue: Florent Castelli <orphis@webrtc.org>
Reviewed-by: Sami Kalliomäki <sakal@webrtc.org>
Reviewed-by: Kári Helgason <kthelgason@webrtc.org>
Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23250}
2018-05-15 17:12:02 +00:00