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chromium-webrtc-autoroll 95a19cae2c Roll chromium_revision d2e8a52a3b..6631cffcc0 (797613:797748) 
Change log: d2e8a52a3b..6631cffcc0
Full diff: d2e8a52a3b..6631cffcc0

Changed dependencies
* src/base: 967db096e8..fa5f8b8ea9
* src/build: ede7a7363b..7198da4c17
* src/ios: ab7afc883c..e534542864
* src/testing: 91e579faae..a606aff6b4
* src/third_party: f38fe6f9ed..869519aac6
* src/third_party/perfetto: dbee93ac20..78d7b4b825
* src/tools: 85bd28b7d3..e8243b6b8e
DEPS diff: d2e8a52a3b..6631cffcc0/DEPS

No update to Clang.

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

Change-Id: I9fdf9aa77115bc80445e26c09416721bc1dffa8c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/181603
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@{#31930}
2020-08-13 18:37:56 +00:00
api
Delete VideoCodec::plType member
2020-08-13 14:18:16 +00:00
audio
Log audio network adaptor and DSCP in AudioSendStream.
2020-08-13 14:05:46 +00:00
build_overrides
set perfetto flag to default value of false
2020-07-22 10:14:53 +00:00
call
Log audio network adaptor and DSCP in AudioSendStream.
2020-08-13 14:05:46 +00:00
common_audio
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
common_video
Move function PrintVideoFrame to the test file where it is used.
2020-08-06 15:20:19 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Disable switch camera and route change buttons when processing
2020-08-11 13:18:39 +00:00
logging
Add missing tests for DTLS state logging in RTC event log.
2020-08-07 11:49:43 +00:00
media
Move H264::Profile to h264_profile_level_id.h
2020-08-12 12:10:24 +00:00
modules
Delete legacy MockEncodedImageCallback::OnEncodedFrame
2020-08-12 14:41:00 +00:00
p2p
Delete left-over TODO comment
2020-08-07 11:36:51 +00:00
pc
Reland "peerconnection: prefer spec names for signaling state"
2020-08-11 15:44:00 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Update settings for balanced degradation.
2020-08-10 10:59:17 +00:00
rtc_tools
Use non-deprecated EncodedImageCallback::OnEncodedImage in rtc_tools
2020-08-12 09:17:15 +00:00
sdk
Reland "Implement transceiver.stop()"
2020-08-11 10:46:23 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
test
Prepare for deleting VideoCodec::plType
2020-08-11 14:20:59 +00:00
tools_webrtc
Fix clang revision regexp in chromium autoroller to match new format
2020-08-07 09:22:03 +00:00
video
[Adaptation] Extract adaptation constraints to their own files
2020-08-13 13:55:03 +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
Use absl_deps in order to preapre to the Abseil component build release.
2020-06-08 12:59:40 +00:00
AUTHORS
Changed AndroidVideoDecoder to also handle IllegalArgumentException and IllegalStateException during the init of the decoder and fallback to a software decoder
2020-08-05 09:41:49 +00:00
BUILD.gn
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +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 H264::Profile to h264_profile_level_id.h
2020-08-12 12:10:24 +00:00
DEPS
Roll chromium_revision d2e8a52a3b..6631cffcc0 (797613:797748)
2020-08-13 18:37:56 +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
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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +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
Inclusive language in PRESUBMIT.py.
2020-07-22 10:01:23 +00:00
pylintrc
…
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
Fix link in documentation. (take 2)
2020-04-16 11:08:43 +00:00
style-guide.md
C++ style: We don't allow designated initializers
2020-06-03 09:11:09 +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
Revert "Support AVX2/FMA intrinsics in Audio Resampler module"
2020-07-30 17:35:30 +00:00
whitespace.txt
Trigger CI bots.
2020-07-15 17:50:55 +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 here 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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