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chromium-webrtc-autoroll b6b7d80ea4 Roll chromium_revision fbd3997e87..03cca1960d (887673:887795) 
Change log: fbd3997e87..03cca1960d
Full diff: fbd3997e87..03cca1960d

Changed dependencies
* src/base: 65424b398e..47943882fb
* src/build: 068753b025..79538b13ec
* src/buildtools/third_party/libunwind/trunk: c8d0fb870a..c0776fc649
* src/ios: 2d15772da6..86d93e1021
* src/testing: 73587aa74e..dd9d32d6cd
* src/third_party: b4d7e91e70..94411bea33
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/edf4e0e88f..12d96cda9c
* src/tools: 979eedbf1e..4e0ef8c556
DEPS diff: fbd3997e87..03cca1960d/DEPS

No update to Clang.

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

Change-Id: I59a003cee2141e4a2401f25f4f2a83fa28693ab6
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/220903
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@{#34173}
2021-05-31 18:47:33 +00:00
api
Move proxies into pc/.
2021-05-27 09:56:42 +00:00
audio
Split VideoReceiveStream2 init into worker / network steps.
2021-05-31 17:10:23 +00:00
build_overrides
Roll chromium + fix: blacklist -> ignorelist for sanitizers suppressions
2021-05-27 16:16:01 +00:00
call
Split VideoReceiveStream2 init into worker / network steps.
2021-05-31 17:10:23 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Update BitBuffer methods to style guide
2021-05-18 11:10:27 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Introduce CreateDataChannelOrError
2021-05-26 09:43:29 +00:00
g3doc
add srtp docs
2021-05-26 09:39:19 +00:00
logging
Revert "Deprecate microsecond timestamps in RTC event log."
2021-05-24 13:11:10 +00:00
media
red: assign payload type 63 to audio/RED for opus
2021-05-31 13:35:44 +00:00
modules
Add ClippingPredictorLevelBuffer circular buffer.
2021-05-31 14:04:54 +00:00
net/dcsctp
dcsctp: Optimize SACK generation
2021-05-31 16:30:21 +00:00
p2p
Remove cricket::DtlsTransportState.
2021-05-21 21:45:29 +00:00
pc
Move PostTask for DeliverRtcp from PeerConnection to Call.
2021-05-31 10:34:09 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Proxy: solve event tracing with compile time strings.
2021-05-28 06:53:29 +00:00
rtc_tools
Convert to NTP time using the real clock.
2021-05-21 19:29:32 +00:00
sdk
Move proxies into pc/.
2021-05-27 09:56:42 +00:00
stats
Support for map of string keys to uint64_t / double values in RTCStats
2021-05-25 20:38:08 +00:00
system_wrappers
Make Clock::ConvertTimestampToNtpTime pure virtual
2021-05-21 09:55:14 +00:00
test
RtpFrameReferenceFinder return frames directly instead of via callback.
2021-05-26 15:47:03 +00:00
tools_webrtc
Make generate_license.py compatible with Python 3.
2021-05-31 07:05:46 +00:00
video
Split VideoReceiveStream2 init into worker / network steps.
2021-05-31 17:10:23 +00:00
.clang-format
Add IncludeBlocks to clang-format.
2021-02-03 16:29:07 +00:00
.git-blame-ignore-revs
…
.gitignore
Add .cache to .gitignore.
2021-01-20 15:01:07 +00:00
.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
AUTHORS
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
BUILD.gn
Switch from check_targets to no_check_targets in .gn
2021-05-20 10:42:21 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
DEPS
Roll chromium_revision fbd3997e87..03cca1960d (887673:887795)
2021-05-31 18:47:33 +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
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
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +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
…
README.md
Move style guide and abseil-in-webrtc into g3doc subfolder
2021-05-13 14:43:10 +00:00
WATCHLISTS
Add hta@ to rtc_base/ and api/ WATCHLISTS.
2021-01-06 09:43:34 +00:00
webrtc_lib_link_test.cc
Deprecate PeerConnectionFactory::CreatePeerConnection
2021-05-10 08:47:48 +00:00
webrtc.gni
Turn on the RTC_ENABLE_WIN_WGC build flag.
2021-05-10 20:16:52 +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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