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chromium-webrtc-autoroll ac80481c48 Roll chromium_revision 57d80a05df..e8ae36dede (753397:753511) 
Change log: 57d80a05df..e8ae36dede
Full diff: 57d80a05df..e8ae36dede

Changed dependencies
* src/base: 802d73d836..7445f330eb
* src/build: c01b08f0db..b287793e9e
* src/ios: c7b9f2f144..24c774f954
* src/testing: d45b0a4f91..7355573ba5
* src/third_party: edd9c72dee..62270d226f
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/54f9ec7399..8351d2182b
* src/third_party/depot_tools: e2ac022f0b..de6c456466
* src/tools: 57bf5d27ad..9b0be5f02a
DEPS diff: 57d80a05df..e8ae36dede/DEPS

No update to Clang.

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

Change-Id: I6cfeb65319e5f89d9be3992bf3c40ce27f15b3b4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/171727
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@{#30892}
2020-03-26 04:41:59 +00:00
api
Make all RtpExtension uris constexpr rather than just const
2020-03-25 14:13:19 +00:00
audio
Switch low bw audio test to histograms.
2020-03-25 14:52:10 +00:00
build_overrides
Purge phoglund from most OWNERS files.
2020-03-09 14:08:30 +00:00
call
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +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
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Rename index.md to README.md to make it automatically show up
2020-03-06 10:43:51 +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
[Stats] Explicit RTP-RTX and RTP-FEC mappings. Unblocks simulcast stats.
2020-03-24 13:31:54 +00:00
modules
Implement AudioEncoder::GetFrameLengthRange() for all audio encoders.
2020-03-25 22:19:21 +00:00
p2p
remove deprecated fields in rtc::NetworkRoute
2020-03-25 06:44:06 +00:00
pc
Refactor ssl_stream_adapter API to show object ownership
2020-03-21 18:53:46 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Add interface_id to rtc::Network
2020-03-25 13:03:09 +00:00
rtc_tools
Chromium refactor: Replace "resources_dirs" with "sources"
2020-03-25 19:32:15 +00:00
sdk
Remove IP address logging from NetworkInformation
2020-03-25 09:52:27 +00:00
stats
Add jitterBufferTargetDelay as RTCNonStandardStatsMember to new GetStats API
2020-03-11 12:08:32 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add RTC_EXPORT to webrtc::Clock
2020-03-10 23:00:21 +00:00
test
Move calculation of target_encode_bitrate to DefaultVideoQualityAnalyzer
2020-03-25 11:38:47 +00:00
tools_webrtc
Greatly simplify flags for test binaries.
2020-03-25 09:56:07 +00:00
video
Fixes temporal rate allocation issues.
2020-03-25 11:20:47 +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
Fix 'all' build on non Android platforms.
2020-03-11 08:24:27 +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 57d80a05df..e8ae36dede (753397:753511)
2020-03-26 04:41:59 +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
Add #COMPONENT to WebRTC.
2019-10-08 12:20:39 +00:00
PATENTS
…
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
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
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
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
Reformat GN files.
2020-01-21 12:13:11 +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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Python 1.3%
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