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chromium-webrtc-autoroll c242d12da2 Roll chromium_revision ca551bc6c0..407aa3d664 (757925:758038) 
Change log: ca551bc6c0..407aa3d664
Full diff: ca551bc6c0..407aa3d664

Changed dependencies
* src/base: 2e77bcfa24..f336b75408
* src/build: 896bf0603d..670c801549
* src/ios: 3c562ab19c..6c6a2d6653
* src/testing: ace718049c..3b24ca9ded
* src/third_party: 71665461cc..8def2539d9
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/c72a211d00..e9a5bd1297
* src/tools: 67dc293003..c53efa33fd
DEPS diff: ca551bc6c0..407aa3d664/DEPS

No update to Clang.

TBR=chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com,
BUG=None

Change-Id: Icf019ccbb346823be9d913a3f22b1a05e5e3c05e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/173342
Reviewed-by: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Commit-Queue: Autoroller <chromium-webrtc-autoroll@webrtc-ci.iam.gserviceaccount.com>
Cr-Commit-Position: refs/heads/master@{#31045}
2020-04-09 22:33:24 +00:00
api
AEC3: Remove deprecated parameter
2020-04-09 12:25:05 +00:00
audio
[InsertableStreams] Don't include the header in the transformable payload.
2020-04-02 11:20:59 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
Revert "[InsertableStreams] Fix simulcast: set frame transformer for all streams"
2020-04-08 10:00:51 +00:00
common_audio
Changed fft4g to be built as C++
2020-03-20 12:10:16 +00:00
common_video
Remove wildcard ownership for build files.
2020-02-19 14:05:46 +00:00
data
…
docs
doc: describe native turnserver scope more clearly
2020-04-01 19:48:20 +00:00
examples
Chromium refactor: Replace "resources_dirs" with "sources"
2020-03-25 19:32:15 +00:00
logging
Replace DataSize and DataRate factories with newer versions
2020-02-18 16:09:50 +00:00
media
Apply performance-for-range-copy fixes.
2020-04-03 11:36:52 +00:00
modules
AEC3: Remove parameters for the legacy filter naming
2020-04-08 07:34:08 +00:00
p2p
Temporarily increase DTLS buffer size to 2.
2020-04-03 02:48:44 +00:00
pc
Handle missing tcptype on TCP ICE candidates.
2020-04-07 21:17:29 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Adds tiny rtc::StringFormat utility
2020-04-09 18:13:11 +00:00
rtc_tools
Total packet rate plots for event_log_visualizer.
2020-04-01 13:14:24 +00:00
sdk
Migrate to modern selector syntax for ObjcVideoEncoderFactory.
2020-04-08 19:04:49 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
style-guide
…
system_wrappers
Do not trigger -Wimplicit-int-float-conversion
2020-04-09 11:43:14 +00:00
test
Restore call's final stats collection in PC level framework
2020-04-09 11:21:04 +00:00
tools_webrtc
Add voip_unittests to gn_isolate_map.pyl
2020-04-03 09:52:33 +00:00
video
[InsertableStreams] Fix IsKeyFrame() for video receiver delegate.
2020-04-07 20:22:42 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .clangd to .gitignore
2019-10-28 12:27:50 +00:00
.gn
Reenable libaom decoder by default
2020-03-18 18:04:41 +00:00
.vpython
Add source-side perf upload script for WebRTC.
2019-11-18 14:37:01 +00:00
abseil-in-webrtc.md
Fix typo in abseil-in-webrtc.md.
2019-12-18 14:27:34 +00:00
AUTHORS
Fix bad frees in error paths of WebRtcAecm_Create
2020-03-21 23:01:47 +00:00
BUILD.gn
Revert "Preparation for ReceiveStatisticsProxy lock reduction."
2020-04-07 19:50:20 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
Don't add webrtc-reviews@ to CC, it can be added globally on Gerrit
2018-10-25 08:19:53 +00:00
common_types.h
Reland "Remove PlayoutDelayOracle and make RtpSenderVideo guarantee delivery"
2020-02-07 08:23:58 +00:00
DEPS
Roll chromium_revision ca551bc6c0..407aa3d664 (757925:758038)
2020-04-09 22:33:24 +00:00
ENG_REVIEW_OWNERS
…
LICENSE
…
license_template.txt
…
native-api.md
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
OWNERS
Remove phoglund as root owner.
2020-03-30 12:15:56 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
Use source_sets in component builds and static_library in release builds.
2019-10-17 21:17:18 +00:00
PRESUBMIT.py
Fix public_deps presubmit and gn format fighting each other.
2020-01-30 11:22:46 +00:00
pylintrc
…
README.chromium
…
README.md
…
style-guide.md
Add guidance to style guide how to reference a bug in a TODO
2019-12-11 11:55:52 +00:00
WATCHLISTS
Remove benwright@webrtc.org from WATCHLISTS
2020-01-31 18:46:52 +00:00
webrtc_lib_link_test.cc
Rewrite the lib link test to just be a binary.
2019-10-18 07:42:20 +00:00
webrtc.gni
Make transient suppression optionally excludable via defines
2020-04-02 11:44:07 +00:00
whitespace.txt
Whitespace change
2020-01-28 07:53:15 +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.

Authoritative list of directories that contain the native API header files.

More info

Description
The idea is to make CMake build for WebRTC m130 version - for audio processing module
Readme BSD-3-Clause 446 MiB
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