webrtc_m130

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Sami Kalliomäki d189a7cbb7 Add Android Studio project for AppRTCMobile that builds against AAR.

This project can be used for testing the AAR before publishing. Removes
dependency to Chromium from the tests to support Android Studio.

Bug: webrtc:8365
Change-Id: I7568a3f636fd7d478d274b4766f33ab00f28a6f0
Reviewed-on: https://webrtc-review.googlesource.com/7608
Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
Commit-Queue: Sami Kalliomäki <sakal@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20269}

2017-10-13 08:46:27 +00:00

api

TurnCustomizer - an interface for modifying stun messages sent by TurnPort

2017-10-11 07:45:29 +00:00

audio

Fix Gn untracked headers in webrtc/call.

2017-10-10 15:13:48 +00:00

build_overrides

Use hermetic toolchain on Mac, except for local iOS builds

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

call

Disable RampUpTest.UpDownUpTransportSequenceNumberPacketLoss.

2017-10-10 19:16:20 +00:00

common_audio

Adding NOLINT for typedefs.h and common_types.h

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

common_video

Add explicit includes of refcountedobject.h where it is used.

2017-10-06 13:00:14 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Add Android Studio project for AppRTCMobile that builds against AAR.

2017-10-13 08:46:27 +00:00

infra

Remove Rietveld CQ config.

2017-10-12 05:02:45 +00:00

logging

Prevent RtcEventLogTestHelper from crashing

2017-10-11 18:12:57 +00:00

media

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

modules

Create links to single experiments in APM-QA exported results.

2017-10-13 07:48:47 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

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

p2p

Fix bug in recently introduced TurnCustomizer

2017-10-13 05:08:26 +00:00

Keep count of libsrtp clients, and only deinitialize when it goes to 0.

2017-10-13 05:11:27 +00:00

resources

Fixing WebRTC after moving from src/webrtc to src/

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

rtc_base

Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy.

2017-10-13 08:32:07 +00:00

rtc_tools

Don't append the bitrate twice to the outgoing bitrate plot in event_log_visualizer.

2017-10-04 16:35:45 +00:00

sdk

Refactor HardwareVideoDecoderTest.

2017-10-13 08:40:07 +00:00

stats

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

2017-10-02 10:47:00 +00:00

system_wrappers

Delete static_instance.h and fix_interlocked_exchange_pointer_win.h

2017-10-05 13:19:21 +00:00

test

Make the complexity of the square generator configurable.

2017-10-10 15:02:58 +00:00

tools_webrtc

Revert "Use subprocess2 for downloading tools."

2017-10-13 06:06:37 +00:00

video

Move HistogramPercentileCounter to rtc_base from RecieveStatisticProxy.

2017-10-13 08:32:07 +00:00

voice_engine

Add explicit includes of refcountedobject.h where it is used.

2017-10-06 13:00:14 +00:00

.clang-format

Tune ObjC clang-format configuration

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

.git-blame-ignore-revs

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.gitignore

MB: Add support for isolating scripts + isolate low_bandwidth_audio_test.py.

2017-10-02 16:57:09 +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 "Reland of Add full stack tests for MediaCodec encoder (moved from Rietveld)."

2017-09-29 13:48:29 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

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

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +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

Set RTPVideoHeader picture id in PayloadRouter if forced fallback for VP8 is enabled.

2017-10-06 13:41:14 +00:00

DEPS

Roll chromium_revision c6405066cc..961817f7c1 (507388:507941)

2017-10-11 11:42:19 +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

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +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

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C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%