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chromium-webrtc-autoroll 637cd3ebd7 Roll chromium_revision 8ef32b65bc..e58e865ec2 (653455:653566) 
Change log: 8ef32b65bc..e58e865ec2
Full diff: 8ef32b65bc..e58e865ec2

Changed dependencies
* src/base: c6eea90abe..7a3355bb35
* src/build: 5160e912c4..a2521a179a
* src/ios: 60068c91fa..d8a10fba64
* src/testing: ebe4d80d63..09217eda36
* src/third_party: 33fd4c6333..b7158829af
* src/tools: 7ff42423e0..6920ccde1f
DEPS diff: 8ef32b65bc..e58e865ec2/DEPS

No update to Clang.

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

Change-Id: If7aa36627557d52ae136990de56e17ffac94109e
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/134082
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@{#27741}
2019-04-24 14:40:10 +00:00
api
AEC3: Stationary init: fixing a typo when reading the json configuration
2019-04-24 07:42:37 +00:00
audio
Add RTP sequence number to TransportFeedbackObserver::AddPacket()
2019-04-23 11:02:56 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Add RTP sequence number to TransportFeedbackObserver::AddPacket()
2019-04-23 11:02:56 +00:00
common_audio
Add ability to play audio in circle for TestAudioDevice wav file capturer
2019-04-16 15:33:03 +00:00
common_video
Prepare to move SPS VUI rewriting out of H.264 packetizer.
2019-04-17 13:21:30 +00:00
crypto
Adding new top-level directory crypto/
2019-03-08 00:35:05 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Fix Screensharing extension build.
2019-04-24 09:19:40 +00:00
logging
Improving robustness of feedback matching code in event log parser.
2019-04-24 13:11:54 +00:00
media
Revert "Copy video frames metadata between encoded and plain frames in one place"
2019-04-24 12:56:52 +00:00
modules
Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader
2019-04-24 13:13:04 +00:00
p2p
Factor out the fake port allocator in build.
2019-04-23 20:47:38 +00:00
pc
Factor out the fake port allocator in build.
2019-04-23 20:47:38 +00:00
resources
AEC3: Avoid using filter output in suppression gain computation in non-linear mode
2019-04-15 16:08:41 +00:00
rtc_base
Delete unused class FileStream
2019-04-24 11:18:55 +00:00
rtc_tools
Add RTP sequence number to TransportFeedbackObserver::AddPacket()
2019-04-23 11:02:56 +00:00
sdk
Remove deprecated method from video decoder interface.
2019-04-24 09:18:35 +00:00
stats
Implement RTCOutboundRtpStreamStats.retransmitted[Bytes/Packets]Sent.
2019-04-17 13:04:50 +00:00
style-guide
…
system_wrappers
Delete unneeded direct includes of common_types.h
2019-04-01 07:18:13 +00:00
test
Revert "Copy video frames metadata between encoded and plain frames in one place"
2019-04-24 12:56:52 +00:00
tools_webrtc
Remove loopback video quality test from configs
2019-04-12 12:32:42 +00:00
video
Move frame_type member from RtpDepacketizer::ParsedPayload to RTPVideoHeader
2019-04-24 13:13:04 +00:00
.clang-format
…
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Bump iOS min supported version to 10.0
2019-03-07 13:08:17 +00:00
.vpython
…
abseil-in-webrtc.md
Ban absl::StrSplit and absl::StrJoin
2019-02-26 00:45:11 +00:00
AUTHORS
Import proto_library.gni when rtc_enable_protobuf is true
2019-02-27 09:56:42 +00:00
BUILD.gn
Replacing SequencedTaskChecker with SequenceChecker.
2019-04-09 12:28:04 +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
Move enum VideoType out of common_types.h
2019-04-08 09:47:54 +00:00
DEPS
Roll chromium_revision 8ef32b65bc..e58e865ec2 (653455:653566)
2019-04-24 14:40:10 +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
Removing niklas.enbom@webrtc.org from WATCHLIST and OWNERS
2019-04-23 14:51:24 +00:00
PATENTS
…
presubmit_test_mocks.py
…
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Add a presubmit check for absl/memory/memory.h inclusion for WrapUnique
2019-02-28 14:12:48 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
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
Removing niklas.enbom@webrtc.org from WATCHLIST and OWNERS
2019-04-23 14:51:24 +00:00
webrtc.gni
Testing no /DUNICODE assumptions with Win more configs bots.
2019-03-28 08:46:37 +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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