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chromium-webrtc-autoroll db01a82ff2 Roll chromium_revision 5371070da9..ca2293bd17 (863745:863854) 
Change log: 5371070da9..ca2293bd17
Full diff: 5371070da9..ca2293bd17

Changed dependencies
* src/base: 3bbd658686..3f7efbf9e1
* src/build: a8fc514621..2e22b6d06b
* src/ios: a532c657e9..56d5aa0f85
* src/testing: abdf5bd723..f83b428d45
* src/third_party: b4d3774855..f2ef9d89aa
* src/third_party/androidx: ZvXzCmnbaONNhr6d8cbYTREJiadk9QBhBCm1foVA0nAC..xIdmDAvhfStt7ky1enCv_j6NNXqdhajuI8563NJ3QekC
* src/third_party/perfetto: a575cd672d..371d8955a1
* src/tools: 09f0df6f24..c624ebd1f0
DEPS diff: 5371070da9..ca2293bd17/DEPS

No update to Clang.

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

Change-Id: I9d97fc3adae487f95fb120393026ec2914f8abf4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/212321
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@{#33493}
2021-03-17 18:44:33 +00:00
api
Reland "Reland "Enable quality scaling when allowed""
2021-03-16 15:13:52 +00:00
audio
Add remote-outbound stats for audio streams
2021-03-12 20:39:50 +00:00
build_overrides
Allow webrtc mac cross compile
2021-03-10 18:42:58 +00:00
call
Reland "[Battery]: Delay start of TaskQueuePacedSender."
2021-03-17 16:10:02 +00:00
common_audio
Remove from chromium build targets that are not compatible with it.
2021-02-01 13:46:19 +00:00
common_video
Delete rtc::Callback0 and friends.
2021-02-16 12:41:35 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
doc: add M88/M89 release notes
2021-03-01 16:16:03 +00:00
examples
Expose addIceCandidate with completion handler.
2021-03-11 16:15:44 +00:00
g3doc
Introduce WebRTC documentation structure and how-to
2021-03-12 12:07:52 +00:00
logging
Replace RTC_DEPRECATED with ABSL_DEPRECATED
2021-02-22 12:53:23 +00:00
media
sctp: Finish sending partial messages before sending stream reset events
2021-03-17 14:52:51 +00:00
modules
AV1: Use Default TX type for encoding
2021-03-17 18:43:28 +00:00
p2p
Allow port 53 as a TURN port.
2021-03-15 12:22:01 +00:00
pc
Disable more flaky PeerConnectionIntegrationTests on Windows
2021-03-17 14:54:41 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Reland "Fix problem with ipv4 over ipv6 on Android"
2021-03-15 10:50:31 +00:00
rtc_tools
Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
2021-02-25 10:48:55 +00:00
sdk
Expose enableImplicitRollback in iOS SDK.
2021-03-17 09:05:35 +00:00
stats
[Stats] Add minimum RTCReceivedRtpStreamStats with jitter and packetsLost
2021-03-11 11:58:58 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Don't use SystemTimeNanos() for current wallclock time on WINUWP
2021-02-23 14:53:22 +00:00
test
Remove NTP timestamp from PacketBuffer::Packet.
2021-03-12 15:19:35 +00:00
tools_webrtc
Avoid two consecutive version updates.
2021-03-16 08:08:52 +00:00
video
Reland "Reland "Enable quality scaling when allowed""
2021-03-16 15:13:52 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore new format cleanup.
2019-07-11 16:18:51 +00:00
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Rename PlayoutDelay --> VideoPlayoutDelay, move to api/video/video_timing.h
2020-09-07 08:37:14 +00:00
.vpython
Reland "Add protobuf-py2_py3 3.13.0 to .vpython."
2020-11-20 07:52:26 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Roll src/third_party/libjpeg_turbo/ fa0de0767..7b4981b65 (2 commits)
2021-02-24 08:40:13 +00:00
BUILD.gn
Move SequenceChecker header to API: step 1, move header only
2021-02-08 11:49:58 +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
DEPS
Roll chromium_revision 5371070da9..ca2293bd17 (863745:863854)
2021-03-17 18:44:33 +00:00
DIR_METADATA
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
ENG_REVIEW_OWNERS
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
LICENSE
…
license_template.txt
…
native-api.md
Make the remote_bitrate_estimator build target private
2020-11-26 12:21:22 +00:00
OWNERS
Move metadata in OWNERS files to DIR_METADATA files.
2021-02-08 19:09:33 +00:00
PATENTS
…
presubmit_test_mocks.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
presubmit_test.py
Reformat python files checked by pylint (part 1/2).
2020-10-30 10:13:11 +00:00
PRESUBMIT.py
Add deprecation section to webrtc style guide
2021-02-22 13:34:40 +00:00
pylintrc
Undo enforcing of PEP-8 pylint changes for method and function names.
2020-11-10 18:26:25 +00:00
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
README.md
doc: move bug reporting instructions to the repository
2020-10-21 14:47:49 +00:00
style-guide.md
Add deprecation section to webrtc style guide
2021-02-22 13:34:40 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +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
build: improve rtc_include_tests documentation
2021-03-05 13:54:20 +00:00
whitespace.txt
Whitespace update
2021-02-14 19:14:44 +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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