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chromium-webrtc-autoroll caf570f4c9 Roll chromium_revision d7e5a784b1..57028bb44e (879377:879509) 
Change log: d7e5a784b1..57028bb44e
Full diff: d7e5a784b1..57028bb44e

Changed dependencies
* src/base: e28a2b686b..a7bc7d4cf5
* src/build: c5571d5c12..2e396ccf2f
* src/ios: 46df6f13a4..5c99369d2f
* src/testing: 71d097df54..5aa5920dae
* src/third_party: b7cb6b22a3..38602aa0b0
* src/third_party/androidx: RjeRyfAqXQf6VfPsfovn5_088ot2hkVyL1FaGWMsxI0C..XlmeAzahzz9NitTuWACJceeLF4HYYTJOmA4X__MlSbYC
* src/third_party/depot_tools: 3da91715d3..7522924749
* src/tools: 2a89d35475..5143fd04ca
DEPS diff: d7e5a784b1..57028bb44e/DEPS

No update to Clang.

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

Change-Id: I08fb3e468f49c4bc0915c756dd37706ec21070b2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217542
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@{#33930}
2021-05-05 20:50:17 +00:00
api
Refactor the PlatformThread API.
2021-05-05 09:59:07 +00:00
audio
Use Timestamp to represent packet receive timestamps
2021-05-04 13:16:54 +00:00
build_overrides
Roll chromium_revision 34f3c82122..2dffe06711 (867171:871492)
2021-04-12 18:25:58 +00:00
call
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
common_audio
Avoid undefined behavior in a division operation.
2021-04-23 07:49:24 +00:00
common_video
Reland "Remove Invoke from VideoChannel::FillBitrateInfo."
2021-05-03 15:16:34 +00:00
data
…
docs
Create a VideoFrameTrackingId RTP header extension.
2021-03-25 17:25:18 +00:00
examples
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +00:00
g3doc
Add conceptual documentation for Audio - Mixer
2021-04-22 08:59:38 +00:00
logging
Remove use of istream in RTC event log parser.
2021-03-31 13:21:58 +00:00
media
Update video tests to clear network interface pointer consistently.
2021-05-03 15:45:35 +00:00
modules
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
net/dcsctp
dcsctp: Handle starting timer from timer callback
2021-05-05 13:13:03 +00:00
p2p
Remove unused setter for Port::socket_factory()
2021-05-04 15:07:48 +00:00
pc
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +00:00
resources
Disable high-pass filtering of the AEC reference
2021-02-23 07:06:11 +00:00
rtc_base
Refactor the PlatformThread API.
2021-05-05 09:59:07 +00:00
rtc_tools
Replace more instances of rtc::RefCountedObject with make_ref_counted.
2021-04-27 17:01:59 +00:00
sdk
Refactor the PlatformThread API.
2021-05-05 09:59:07 +00:00
stats
Simplify reference counting implementation of PendingTaskSafetyFlag.
2021-04-21 07:04:01 +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
Refactor the PlatformThread API.
2021-05-05 09:59:07 +00:00
tools_webrtc
crc32c: Point the licensing script to the LICENSE file
2021-05-03 16:46:30 +00:00
video
Store RtpPacketReceived::arrival_time as Timestamp.
2021-05-05 16:22:33 +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
Enable GN check on //net.
2021-05-03 14:23:09 +00:00
.vpython
Update six library version
2021-04-26 16:39:07 +00:00
abseil-in-webrtc.md
Polish the "Using Abseil in WebRTC" docs
2020-10-16 13:42:00 +00:00
AUTHORS
Revert "Ensure method which updates UI is called in main thread"
2021-04-30 09:26:03 +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 d7e5a784b1..57028bb44e (879377:879509)
2021-05-05 20:50:17 +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 mbonadei@ as owner of .pylintrc / .vpython.
2021-04-26 16:34:57 +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
Fix incorrect link in README.md
2021-04-20 10:58:08 +00:00
style-guide.md
Enhance the readability of the style guide.
2021-04-23 15:35:56 +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
Add GetTopLeft to WgcCaptureSource to facilitate cursor capture.
2021-04-23 16:46:49 +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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