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chromium-webrtc-autoroll e19a6da043 Roll chromium_revision a77f654a3c..e7ecd1bfc2 (634608:634731) 
Change log: a77f654a3c..e7ecd1bfc2
Full diff: a77f654a3c..e7ecd1bfc2

Changed dependencies
* src/base: b85c191a4b..684b17e2c5
* src/build: 582ae4a19d..52fa227c5e
* src/ios: 4fc4adb870..18051ca0c7
* src/testing: 22c79184e4..6ae2f43c72
* src/third_party: 54ddfab62d..5e378b8f47
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/a74495a295..b4fdbb737c
* src/third_party/depot_tools: c0546b9817..e831004371
* src/tools: e22d14c764..8d9358aff8
DEPS diff: a77f654a3c..e7ecd1bfc2/DEPS

No update to Clang.

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

Change-Id: I0313f65ee3f9b53048acde8fca7e009b096f6304
Reviewed-on: https://webrtc-review.googlesource.com/c/123943
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@{#26821}
2019-02-22 18:29:35 +00:00
api
Add support for TransportSequenceNumberV2 in SDP negotiation
2019-02-22 12:47:04 +00:00
audio
Reland "Delete test/constants.h"
2019-02-19 08:51:20 +00:00
build_overrides
Disable hermetic toolchain when building on macOS 10.14.
2018-12-07 13:57:28 +00:00
call
Adds FakeNetworkPipeTest to rtc_unittests.
2019-02-22 15:05:39 +00:00
common_audio
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
common_video
Always add/rewrite VUI and set max_num_reorder_frames to 0.
2019-02-04 09:18:46 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Make sure RTC_SUPPORTS_METAL is set in AppRTCMobile.
2019-02-21 18:51:47 +00:00
logging
Add missing absl/memory/memory.h to rtc_event_generic_ack_received.cc
2019-02-22 11:14:38 +00:00
media
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
modules
Replace field trials with WebRtcKeyValueConfig in PacedSender
2019-02-22 13:08:18 +00:00
p2p
Allow IceConnectionState to become failed without ever connecting.
2019-02-22 10:17:36 +00:00
pc
Add support for TransportSequenceNumberV2 in SDP negotiation
2019-02-22 12:47:04 +00:00
resources
2nd reland of https://webrtc-review.googlesource.com/c/src/+/114883
2019-02-22 09:59:01 +00:00
rtc_base
Migrate stdlib task queue to TaskQueueBase interface
2019-02-21 14:36:07 +00:00
rtc_tools
Remove webrtc::ProtoString.
2019-02-16 11:11:45 +00:00
sdk
Synchronize replaceRegion calls.
2019-02-22 14:03:06 +00:00
stats
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
webrtc: Remove semicolons.
2019-02-20 16:02:59 +00:00
test
Rename "OnReceivedFrame" to "OnAssembledFrame"
2019-02-22 10:49:07 +00:00
tools_webrtc
Delete obsolete usage of FakeConstraints
2019-02-12 12:27:04 +00:00
video
Rename "OnReceivedFrame" to "OnAssembledFrame"
2019-02-22 10:49:07 +00:00
.clang-format
Tune ObjC clang-format configuration
2017-05-11 09:14:18 +00:00
.git-blame-ignore-revs
Let git-hyper-blame ignore format commit.
2018-06-20 09:26:44 +00:00
.gitignore
Add Visual Studio Code project folder to gitignore file.
2019-01-21 18:42:33 +00:00
.gn
Re-enable gtest absl pretty printers.
2018-08-13 13:54:05 +00:00
.vpython
Add vpython dependencies needed to run presubmit tests on LUCI
2018-05-18 08:10:25 +00:00
abseil-in-webrtc.md
Migrate gcd task queue implementation to TaskQueueBase interface
2019-02-20 17:08:53 +00:00
AUTHORS
[AndroidAudioRecord] Added audio format parameter to configure AudioRecord - JavaAudioDeviceModule
2019-02-20 15:08:53 +00:00
BUILD.gn
Adds FakeNetworkPipeTest to rtc_unittests.
2019-02-22 15:05:39 +00:00
CODE_OF_CONDUCT.md
Add code of conduct to WebRTC repo
2017-05-16 12:09:13 +00:00
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
Delete constant RTP_PAYLOAD_NAME_SIZE
2019-01-25 07:59:52 +00:00
DEPS
Roll chromium_revision a77f654a3c..e7ecd1bfc2 (634608:634731)
2019-02-22 18:29:35 +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
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
license_template.txt
Update template to follow chromium copyright style
2013-04-24 01:01:28 +00:00
native-api.md
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Clean up root OWNERS.
2018-11-09 14:23:59 +00:00
PATENTS
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
presubmit_test_mocks.py
Reland: Add presubmit check for changes in 3pp
2018-05-22 13:11:18 +00:00
presubmit_test.py
Fixing py lint errors
2018-07-23 15:28:48 +00:00
PRESUBMIT.py
Add PRESUBMIT to enforce usage of new Googletest APIs.
2019-02-06 11:06:02 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
Moving src/webrtc into src/.
2017-09-15 04:25:06 +00:00
README.md
Tell users where they can find the native API headers
2017-11-14 10:36:46 +00:00
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Modify pc/ WATCHLISTS definition
2019-01-31 22:09:16 +00:00
webrtc.gni
Add gn configs to remove the dependency to audio and video codecs.
2019-02-12 20:41:25 +00:00
whitespace.txt
Whitespace change
2018-02-23 10:34:16 +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 http://www.webrtc.org/native-code/development 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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