webrtc_m130

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Alessio Bazzica 849030dab8 Optionally copy clean speech input files under _cache with APM-QA.

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Bug: webrtc:7494
Change-Id: I41c5cfc6fd57aefaf246816c0ba4094947b9e767
Reviewed-on: https://webrtc-review.googlesource.com/13123
Reviewed-by: Alessio Bazzica <alessiob@webrtc.org>
Commit-Queue: Alessio Bazzica <alessiob@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20343}

2017-10-19 08:51:44 +00:00

api

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

audio

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

build_overrides

Use hermetic toolchain on Mac, except for local iOS builds

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

call

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

common_audio

Adding NOLINT for typedefs.h and common_types.h

2017-09-15 13:03:51 +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

Replace use of codec name strings with constants.

2017-10-19 07:28:12 +00:00

infra

Remove Rietveld CQ config.

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

logging

Prevent unbounded memory consumption through RtcEventLogImpl::config_history_

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

media

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

modules

Optionally copy clean speech input files under _cache with APM-QA.

2017-10-19 08:51:44 +00:00

ortc

Disable flaky test OrtcFactoryIntegrationTest.BasicTwoWayAudioVideoRtpSendersAndReceivers.

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

p2p

Rewrite WebRtcSession ICE tests as PeerConnection tests

2017-10-13 18:42:34 +00:00

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

resources

Fixing WebRTC after moving from src/webrtc to src/

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

rtc_base

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +00:00

rtc_tools

Partial revert of "Update scripts to use DEPS-pinned depot_tools"

2017-10-18 13:06:58 +00:00

sdk

Replace use of codec name strings with constants.

2017-10-19 07:28:12 +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

Separate AEC3 config from AudioProcessing::Config.

2017-10-19 08:19:52 +00:00

tools_webrtc

MB: Remove WebRTC Perf Tests from FYI waterfall.

2017-10-18 18:26:04 +00:00

video

Update aggregating interval in getStats for receive side.

2017-10-16 17:19:07 +00:00

voice_engine

Revert "BWE allocation strategy allows controlling of bitrate allocation with WEBRTC external logic."

2017-10-17 19:59:04 +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 65e1b24a78..f4ecd4bed3 (508708:508787)

2017-10-16 11:00:27 +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

GN rtc_* templates: Forward global "visibility"

2017-10-16 11:04:07 +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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