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chromium-webrtc-autoroll 5fe0b372ba Roll chromium_revision 7e70585ca5..74f869d04b (871605:871745) 
Change log: 7e70585ca5..74f869d04b
Full diff: 7e70585ca5..74f869d04b

Changed dependencies
* src/base: 8fc5bf1d4a..ca7b938131
* src/build: 5d0017aeec..79006bea8b
* src/ios: 0b3d0c7d99..b10d9dc408
* src/testing: f3e0beba79..21d746be78
* src/third_party: fcd6536576..cd3c7ea8d4
* src/third_party/androidx: DkjJyAndtE7UV7h5rNQdFWztxKvMiWIhS0cXGk4pS-UC..IoB78nVutc7u1QVob6zeGRS6YrTtEaNyYCFV4iBH3bcC
* src/third_party/ffmpeg: 4fb42ae52e..280d5fd0df
* src/third_party/perfetto: 1b78913ea7..31ac7832bf
* src/tools: 7ad58b1ed2..55b8544079
DEPS diff: 7e70585ca5..74f869d04b/DEPS

No update to Clang.

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

Change-Id: I2a9ec3ad3e56a026b59cfc672c05ba5dbf12f5f1
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215103
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@{#33701}
2021-04-13 02:44:09 +00:00
api
Move threading documentation for API into g3doc structure
2021-04-12 13:38:49 +00:00
audio
Remove assoc_send_channel_lock_ from ChannelReceive.
2021-04-01 21:49:02 +00:00
build_overrides
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
call
Update WebRTC code version (2021-04-12T04:03:54).
2021-04-12 05:36:03 +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
Provide a default implementation of NV12BufferInterface::CropAndScale.
2021-03-22 11:09:36 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
addIceCandidate with callback into Android's SDK.
2021-04-12 07:04:54 +00:00
g3doc
Add module overview of ICE
2021-04-12 14:03:57 +00:00
logging
Remove use of istream in RTC event log parser.
2021-03-31 13:21:58 +00:00
media
sctp: Rename SctpTransport to UsrSctpTransport
2021-04-12 10:40:34 +00:00
modules
[Battery]: TaskQueuePacedSender not started by default.
2021-04-12 15:36:46 +00:00
net/dcsctp
dcsctp: Use correct field width for PPID
2021-04-12 09:28:48 +00:00
p2p
Remove MDNS message implementation
2021-04-12 22:24:56 +00:00
pc
srtp: compare key length to srtp policy key length
2021-04-12 07:57:03 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Revert "Add addr in error msg if stun sock sent with error"
2021-04-12 19:51:20 +00:00
rtc_tools
Reland "Enable use of rtc::SystemTimeNanos() provided by Chromium"
2021-02-25 10:48:55 +00:00
sdk
addIceCandidate with callback into Android's SDK.
2021-04-12 07:04:54 +00:00
stats
Remove RTCRemoteInboundRtpStreamStats duplicate members.
2021-04-08 09:06:24 +00:00
style-guide
Remove kwiberg@webrtc.org from OWNERS files
2020-12-04 15:11:26 +00:00
system_wrappers
Consolidate the different NTP clocks into one.
2021-04-08 13:54:04 +00:00
test
Remove MDNS message implementation
2021-04-12 22:24:56 +00:00
tools_webrtc
Add dcsctp_unittests to gn_isolate_map.
2021-03-31 10:33:17 +00:00
video
Remove unused members in tests.
2021-04-12 07:21:03 +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
Adds missing header to fix compilation error when compiling with use_custom_libcxx set to false.
2021-03-25 09:57:00 +00:00
BUILD.gn
Move RTC_ENABLE_WIN_WGC define to the top level BUILD.gn
2021-04-08 16:31:49 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision 7e70585ca5..74f869d04b (871605:871745)
2021-04-13 02:44:09 +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
g3doc.lua
Improve webrtc documentation infra. Preview at:
2021-03-30 10:29:30 +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
Add titovartem@webrtc.org as owner for /g3doc
2021-04-12 13:40:47 +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
sctp: Rename SctpTransport to UsrSctpTransport
2021-04-12 10:40:34 +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
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
whitespace.txt
Reland "Triggering CI."
2021-03-22 11:57:23 +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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