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Oleh Prypin cdab13d9a3 Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788) 
Change log: c27b32b2fd..1a597bc4e4
Full diff: c27b32b2fd..1a597bc4e4

Changed dependencies
* src/base: 5b5c6b774e..1161fb86d1
* src/build: ae9b018ef6..e148b4cffc
* src/ios: 472ec5538f..3f7730ca8d
* src/testing: 73be5bfc61..bbb6ca1af1
* src/third_party: 778cfca8c4..a8a90adc37
* src/third_party/boringssl/src: https://boringssl.googlesource.com/boringssl.git/+log/1a51a5b4a6..70fe610556
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/78b1c61fff..11e283fea2
* src/third_party/depot_tools: 484d74fa21..61d0c29253
* src/third_party/freetype/src: 1e7a8f30c2..a6feefdfef
* src/tools: 662d7014c7..a02273b65b
DEPS diff: c27b32b2fd..1a597bc4e4/DEPS

Clang version changed 352138:353250
Details: c27b32b2fd..1a597bc4e4/tools/clang/scripts/update.py

Bug: None
Change-Id: I471a19927cefe25b3b2aa893cb9385a38fc9d5bf
Reviewed-on: https://webrtc-review.googlesource.com/c/121702
Commit-Queue: Artem Titarenko <artit@webrtc.org>
Reviewed-by: Artem Titarenko <artit@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26581}
2019-02-07 08:39:57 +00:00
api
Delete MediaTransport method SetNetworkChangeCallback
2019-02-06 17:46:19 +00:00
audio
Propagate base minimum delay to audio_receiver_stream
2019-02-06 11:07:42 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Propagate base minimum delay to audio_receiver_stream
2019-02-06 11:07:42 +00:00
common_audio
Stop using Googletest legacy APIs.
2019-01-31 13:23:33 +00:00
common_video
Always add/rewrite VUI and set max_num_reorder_frames to 0.
2019-02-04 09:18:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Remove unused file.
2019-02-06 10:24:07 +00:00
logging
Add events processing to GetIceEvents.
2019-02-06 14:14:09 +00:00
media
Reland: Remove dead code from stream_params.h
2019-02-06 19:18:15 +00:00
modules
Move RtpSenderVideo tests to separate file.
2019-02-06 18:00:39 +00:00
p2p
Remove TCPPort incoming_only_ member.
2019-02-04 19:46:30 +00:00
pc
Allow more than one registered network change callback in MediaTransport
2019-02-06 16:14:45 +00:00
resources
Reland of Opus multistream.
2019-01-29 12:16:19 +00:00
rtc_base
Implement encoder overshoot detector and rate adjuster.
2019-02-06 15:54:11 +00:00
rtc_tools
Do not build rtp_generator in Chromium builds.
2019-02-04 14:48:58 +00:00
sdk
Separate AndroidVideoTrackSource::OnFrameCaptured from adaptation
2019-02-06 14:33:59 +00:00
stats
Expose video freeze metrics in GetStats.
2019-02-04 09:58:08 +00:00
style-guide
…
system_wrappers
Makes Clock interface fully mutable.
2019-01-31 08:12:05 +00:00
test
Fix test FrameGenerator to work with a single file source
2019-02-06 11:59:03 +00:00
tools_webrtc
Pause rolling buildtools
2019-02-07 08:38:52 +00:00
video
Implement encoder overshoot detector and rate adjuster.
2019-02-06 15:54:11 +00:00
.clang-format
…
.git-blame-ignore-revs
…
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
…
.vpython
…
abseil-in-webrtc.md
Allow use of functions in absl/algorithms
2019-01-28 18:42:16 +00:00
AUTHORS
Add ',' between elements in RTCStatsReport::ToJson
2019-01-08 16:05:23 +00:00
BUILD.gn
Remove HAVE_WEBRTC_VOICE.
2019-02-06 18:39:45 +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
Delete constant RTP_PAYLOAD_NAME_SIZE
2019-01-25 07:59:52 +00:00
DEPS
Roll chromium_revision c27b32b2fd..1a597bc4e4 (629510:629788)
2019-02-07 08:39:57 +00:00
ENG_REVIEW_OWNERS
Enforce LGTM from owners of depends-on paths in DEPS via presubmit.
2018-09-28 12:49:54 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
…
PRESUBMIT.py
Add PRESUBMIT to enforce usage of new Googletest APIs.
2019-02-06 11:06:02 +00:00
pylintrc
…
README.chromium
…
README.md
…
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Override default manifest from Chromium in WebRTC.
2019-01-24 13:00:49 +00:00
whitespace.txt
…

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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