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chromium-webrtc-autoroll acc1123a80 Roll chromium_revision 598b095453..f629a87d19 (787511:787611) 
Change log: 598b095453..f629a87d19
Full diff: 598b095453..f629a87d19

Changed dependencies
* src/base: fb172975a5..abd5405c28
* src/build: eca94691a0..9d828da174
* src/third_party: 0008680207..05a05f9845
* src/third_party/depot_tools: 0be4724e2f..271a78ac35
* src/tools: 2e25d4684d..84642d7d96
DEPS diff: 598b095453..f629a87d19/DEPS

No update to Clang.

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

Change-Id: I768f0891f781f758500d98ad91cb7d8c7c01875d
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179189
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@{#31711}
2020-07-13 02:52:43 +00:00
api
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
audio
webrtc::AudioSendStream: Add lock annotation to audio_level_
2020-07-06 17:05:25 +00:00
build_overrides
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
call
[Adaptation] Move AdaptationConstraints to VideoStreamAdapter
2020-07-09 13:06:56 +00:00
common_audio
Replace slave -> helper, master -> reference
2020-06-29 12:18:05 +00:00
common_video
Migrate common_video/ and examples/ to webrtc::Mutex.
2020-07-07 13:33:27 +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
Add missing dependencies to java targets.
2020-07-10 09:17:00 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Migrate leftovers in media/ and modules/ to webrtc::Mutex.
2020-07-10 08:27:45 +00:00
modules
Check old_vector_size prior to copying in RTPFragmentationHeader::Resize
2020-07-10 17:35:53 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
peerconnection: remove old helper function
2020-07-10 12:35:59 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Migrate rtc_base to webrtc::Mutex.
2020-07-08 20:38:54 +00:00
rtc_tools
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
sdk
Add missing dependencies to java targets.
2020-07-10 09:17:00 +00:00
stats
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
style-guide
…
system_wrappers
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
test
Add jitter buffer delay metric to pc level tests
2020-07-10 11:30:39 +00:00
tools_webrtc
Revert "Test luci.notifier."
2020-07-11 10:05:11 +00:00
video
Migrate VideoStreamDecoderImpl to webrtc::Mutex.
2020-07-10 11:02:28 +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
Relanding: Fix data channel message integrity violation
2020-07-07 03:06:24 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +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
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision 598b095453..f629a87d19 (787511:787611)
2020-07-13 02:52:43 +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
…
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.
2020-06-30 14:03:10 +00:00
pylintrc
…
README.chromium
…
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
Add target_sdk_version to rtc_test.
2020-07-11 08:25:04 +00:00
whitespace.txt
Trigger bots.
2020-06-15 08:40:44 +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
Languages
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C 2.2%
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Python 1.3%
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