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chromium-webrtc-autoroll 529c06498c Roll chromium_revision abd2dc6530..3a98cccc77 (790286:790410) 
Change log: abd2dc6530..3a98cccc77
Full diff: abd2dc6530..3a98cccc77

Changed dependencies
* src/base: c9c839d525..ad8cbea254
* src/build: e62a42e414..1ee11394cf
* src/ios: ad94e7d7d5..0bb13ae251
* src/testing: f126fc6578..6b481f35d3
* src/third_party: 6dcf344d93..0b665927c2
* src/tools: 2bf18d4320..bdb9f16475
DEPS diff: abd2dc6530..3a98cccc77/DEPS

No update to Clang.

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

Change-Id: I5dd60d40abb35e5ba468e1c164d6376f7042621b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/179826
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@{#31776}
2020-07-21 16:39:09 +00:00
api
Add list of local_addresses for network stats object
2020-07-21 16:17:02 +00:00
audio
Remove unused critical section includes.
2020-07-16 13:52:28 +00:00
build_overrides
Rename sanitizers suppression files.
2020-07-03 18:29:58 +00:00
call
Do not propage RTPFragmentationHeader into rtp_rtcp
2020-07-21 14:37:08 +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
…
docs
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
examples
Implemented Android Demo Application for VoIP API
2020-07-21 16:34:22 +00:00
logging
Search and replace gendered terms according to style guide:
2020-06-12 14:12:54 +00:00
media
Delete unused enum values for DataChannelType
2020-07-17 08:28:20 +00:00
modules
Do not propage RTPFragmentationHeader into rtp_rtcp
2020-07-21 14:37:08 +00:00
p2p
Reland "Implement packets_(sent | received) for RTCTransportStats"
2020-07-10 11:50:59 +00:00
pc
Revert "sdp: parse and serialize b=TIAS"
2020-07-17 21:12:17 +00:00
resources
iSAC API wrapper unit test fix
2020-02-27 14:27:23 +00:00
rtc_base
Rename CriticalSection to RecursiveCriticalSection.
2020-07-17 09:19:50 +00:00
rtc_tools
Migrate rtc_tools/, sdk/, and system_wrappers/ to webrtc::Mutex.
2020-07-10 08:02:05 +00:00
sdk
Add presubmit test to use RTC_OBJC_TYPE on RTC_OBJC_EXPORT types.
2020-07-17 09:57:50 +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 list of local_addresses for network stats object
2020-07-21 16:17:02 +00:00
tools_webrtc
Auto roller: send trooper notifications only when Commit-Queue+2.
2020-07-13 09:08:48 +00:00
video
Remove unused critical section includes.
2020-07-16 13:52:28 +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
Explicitly set presentation style to full screen, because,
2020-07-17 07:15:36 +00:00
BUILD.gn
Convert GN libs lists to frameworks
2020-07-06 10:08:09 +00:00
CODE_OF_CONDUCT.md
…
codereview.settings
…
common_types.h
Replaces OverheadObserver with simple getter.
2020-05-07 17:33:45 +00:00
DEPS
Roll chromium_revision abd2dc6530..3a98cccc77 (790286:790410)
2020-07-21 16:39:09 +00:00
ENG_REVIEW_OWNERS
…
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
Add presubmit test to use RTC_OBJC_TYPE on RTC_OBJC_EXPORT types.
2020-07-17 09:57:50 +00:00
pylintrc
…
README.chromium
Add CPEPrefix.
2020-07-13 11:42:07 +00:00
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 CI bots.
2020-07-15 17:50:55 +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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