webrtc_m130

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Alex Loiko 5aea38c8be Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}.

Tests disabled for Mac only.

Tests fail in this way on perf bot Mac 10.11:

[ RUN      ] CallPerfTest.CaptureNtpTimeWithNetworkDelay
../../call/call_perf_tests.cc:407: Failure
Value of: std::abs(time_offset_ms) < threshold_ms_
  Actual: false
Expected: true



TBR=stefan@webrtc.org

Bug: webrtc:8291
Change-Id: I8d173fcff21f096827f31ddd670b6647796bff4b
Reviewed-on: https://webrtc-review.googlesource.com/4041
Commit-Queue: Alex Loiko <aleloi@webrtc.org>
Reviewed-by: Alex Loiko <aleloi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#19995}

2017-09-27 11:45:46 +00:00

api

Remove unnecessary audio references in PeerConnectionFactory

2017-09-23 14:36:14 +00:00

audio

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

build_overrides

Use hermetic toolchain on Mac, except for local iOS builds

2017-09-11 18:38:48 +00:00

call

Disabling CallPerfTest.{CaptureNtpTimeWithNetworkDelay,CaptureNtpTimeWithNetworkJitter}.

2017-09-27 11:45:46 +00:00

common_audio

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

common_video

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

2016-11-22 18:43:05 +00:00

examples

Reland "Remove precompiled header for AppRTCMobile."

2017-09-27 09:02:15 +00:00

infra

Update webrtc CQ to commit to new location

2017-09-13 18:05:33 +00:00

logging

Remove #include of rtc_stream_config.h from rtc_event_log.h

2017-09-21 09:05:54 +00:00

media

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

modules

audioproc_f with simulated mic analog gain

2017-09-27 10:27:56 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

2017-09-27 09:14:28 +00:00

p2p

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

Reland: Completed the functionalities of SrtpTransport.

2017-09-26 18:12:45 +00:00

resources

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

rtc_base

Remove backwards compatibilty header for Optional

2017-09-20 19:17:42 +00:00

rtc_tools

Remove remaining mentions of gflags

2017-09-25 15:34:41 +00:00

sdk

Android: Generate JNI code for VideoSink and VideoEncoder

2017-09-27 11:25:46 +00:00

stats

Added RTCMediaStreamTrackStats.concealmentEvents

2017-09-18 08:58:06 +00:00

system_wrappers

Delete unused Atomic32 overloads of binary +/- operators.

2017-09-22 08:54:23 +00:00

test

Remove the VoiceEngineObserver callback interface.

2017-09-26 16:35:01 +00:00

tools_webrtc

Fix isac_fix_test on swarming perf bot.

2017-09-27 06:32:15 +00:00

video

Continuously request keyframes if decoding does not recover.

2017-09-26 09:54:58 +00:00

voice_engine

Remove unnecessary send codec initialization from voe::Channel.

2017-09-27 00:13:19 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

Create .git-blame-ignore-revs and add Java format CL to it.

2016-10-20 09:20:39 +00:00

.gitignore

gitignore: Remove webrtc/

2017-09-15 10:51:29 +00:00

.gn

Remove remaining mentions of gflags

2017-09-25 15:34:41 +00:00

.vpython

Add psutil to vpython dependencies (used on builder bots)

2017-09-04 08:04:16 +00:00

AUTHORS

Reland of Fix the video buffer size should take rtt into consideration (patchset #2 id:160001 of https://codereview.chromium.org/3002033002/ )

2017-09-25 13:37:12 +00:00

BUILD.gn

Revert of Add full stack tests for MediaCodec. (patchset #10 id:180001 of https://codereview.webrtc.org/3005253002/ )

2017-09-22 11:30:08 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Update README.md and codereview.settings for new source location

2017-09-13 19:54:59 +00:00

common_types.cc

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +00:00

common_types.h

Added RTCMediaStreamTrackStats.concealmentEvents

2017-09-18 08:58:06 +00:00

DEPS

Roll chromium_revision 69fe0e1a5f..ff8cef57fe (504538:504574)

2017-09-27 04:22:45 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

OWNERS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Reland "Adding PRESUBMIT check to avoid mixing C, C++ and Objc-C/Obj-C++.""

2017-09-19 14:23:00 +00:00

presubmit_test.py

Presubmit: Add check to support b/xxx entry in bug reference.

2017-09-26 13:50:25 +00:00

PRESUBMIT.py

Presubmit: Add check to support b/xxx entry in bug reference.

2017-09-26 13:50:25 +00:00

pylintrc

Fix Python shebang and license for presubmit_test.py

2017-09-15 09:22:50 +00:00

README.chromium

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

Update README.md and codereview.settings for new source location

2017-09-13 19:54:59 +00:00

style-guide.md

Style guide: Attempt to make the L2 and L3 headings more visually distinct

2017-09-09 03:52:23 +00:00

typedefs.h

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

WATCHLISTS

Add myself to the watchlist for webrtc/api/ and webrtc/base/

2017-05-04 13:22:46 +00:00

webrtc.gni

Fixing WebRTC after moving from src/webrtc to src/

2017-09-15 05:02:56 +00:00

whitespace.txt

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

README.md

WebRTC is a free, open software project that provides browsers and mobile applications with Real-Time Communications (RTC) capabilities via simple APIs. The WebRTC components have been optimized to best serve this purpose.

Our mission: To enable rich, high-quality RTC applications to be developed for the browser, mobile platforms, and IoT devices, and allow them all to communicate via a common set of protocols.

The WebRTC initiative is a project supported by Google, Mozilla and Opera, amongst others.

Development

See http://www.webrtc.org/native-code/development for instructions on how to get started developing with the native code.

More info

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%