webrtc_m130

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Alessio Bazzica 86f25d3bcd Create links to single experiments in APM-QA exported results.

For each single experiment, a URL is defined by adding a specific anchor.
A URL can be copied using the button beneath the score of the experiment
one would like to share.

This CL also includes a few optimizations and fixes:
- JS and CSS are minified
- Dialog close event listener added, this fixes a small bug preventing
  the play out audio to stop when pressing ESC instead of using the close
  button
- Snackbar notifications added
- Simple unit test for the export module

BUG=webrtc:7218
Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2

TBR=

Change-Id: Iad00ce69094a5968ee0520d105d59656cfafa4e2
Reviewed-on: https://webrtc-review.googlesource.com/7960
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20266}

2017-10-13 07:48:47 +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

Fix fat build for ios.

2017-10-11 09:13:39 +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

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

2017-10-06 13:00:14 +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

Updating RTCDeviceType with new iOS devices.

2017-10-11 10:06:59 +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

Revert "Remove sent framerate and bitrate calculations from MediaOptimization."

2017-10-11 12:59:15 +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

…

.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

Languages

C++ 90.3%

Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%