webrtc_m130

Go to file

Autoroller 0110d3e0ca Roll chromium_revision de566f6b16..07493032b5 (512797:512803)

Change log: de566f6b16..07493032b5
Full diff: de566f6b16..07493032b5

Changed dependencies:
* src/base: f073360234..e8fcacc606
* src/testing: 9648a9d6ce..1d0ff6c8e8
* src/third_party: 6251859a70..2609167cd8
DEPS diff: de566f6b16..07493032b5/DEPS

No update to Clang.

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

Change-Id: I907a6f1a92ffa02b8d920ad06fc75903406c6e72
Reviewed-on: https://webrtc-review.googlesource.com/17284
Commit-Queue: WebRTC Buildbot <buildbot@webrtc.org>
Reviewed-by: WebRTC Buildbot <buildbot@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20500}

2017-10-31 13:13:02 +00:00

api

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

audio

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

common_audio

Replacing undefined left shifts with multiplication.

2017-10-31 09:43:02 +00:00

common_video

Use $rtc_libyuv_dir in common_video/BUILD.gn, not hard-encoded "libyuv"

2017-10-16 16:05:37 +00:00

data

WebRTC: Replace ProjectRootPath by ResourcePath

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

examples

Fix/suppress new warnings introduced in Chromium roll.

2017-10-30 16:10:29 +00:00

infra

Temporarily remove linux_ubsan from commit queue

2017-10-27 09:49:17 +00:00

logging

Prevent unbounded memory consumption through RtcEventLogImpl::config_history_

2017-10-13 10:47:26 +00:00

media

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

modules

New PacedSender constructor with injected PacketQueue

2017-10-31 11:39:22 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

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

p2p

Fix clang style warnings in p2p/base/packettransport files

2017-10-30 19:51:49 +00:00

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

resources

Fix of integer overflow in WebRtcAecm_ProcessBlock / ApmTest.Process

2017-10-23 14:25:37 +00:00

rtc_base

Adding RTC_ prefixed LOG macros.

2017-10-27 08:24:37 +00:00

rtc_tools

Estimate RTP clock frequency and plot capture-send delay.

2017-10-26 08:42:54 +00:00

sdk

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +00:00

stats

Added RTCMediaStreamTrackStats.jitterBufferDelay for audio

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

system_wrappers

Remove references to and implementation of GetHistogramName().

2017-10-30 19:20:49 +00:00

test

Allow injection of custom network models in place of FakeNetworkPipe.

2017-10-26 11:11:25 +00:00

tools_webrtc

MB: Add Android Perf (swarming)

2017-10-23 13:02:07 +00:00

video

Fix typo in VideoSendTiming header extension structure

2017-10-31 11:20:22 +00:00

voice_engine

Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API

2017-10-31 12:35:42 +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 de566f6b16..07493032b5 (512797:512803)

2017-10-31 13:13:02 +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

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +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

Enable cpplint in pc/

2017-10-30 18:08:29 +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

Remove QUIC transport/data channel

2017-10-24 16:14:18 +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%