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chromium-webrtc-autoroll 86f8b3bd5d Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093) 
Change log: 48f88caf2e..619b1dc518
Full diff: 48f88caf2e..619b1dc518

Changed dependencies
* src/base: 41fb4d35eb..4c0433c9cd
* src/ios: 2927c5e09c..b61081dc76
* src/third_party: 62faa6561e..93cfdd52ab
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/43030745c1..dd686e35e9
* src/tools: 55dcdc7de2..af10189827
DEPS diff: 48f88caf2e..619b1dc518/DEPS

No update to Clang.

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

Change-Id: I78dff6e9d91d1195a9ece36ae2c2af7801a4b0d4
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/144704
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@{#28499}
2019-07-05 22:46:33 +00:00
api
Remove deprecation notice for VideoFrame::ntp_time_ms
2019-07-05 18:37:41 +00:00
audio
Introduce PC level audio quality test.
2019-07-05 11:41:38 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
common_audio
Initialize signal processing function pointers statically
2019-06-28 14:20:03 +00:00
common_video
Extend structures to store updated version of the dependency descriptor
2019-06-27 12:10:38 +00:00
crypto
Adding new top-level directory crypto/
2019-03-08 00:35:05 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
examples
Delete RtcEventLogFactory factory as now unused
2019-06-27 10:03:22 +00:00
logging
Delete RtcEventLogFactory factory as now unused
2019-06-27 10:03:22 +00:00
media
Remove trial WebRTC-SimulcastMaxLayers and make its behavior default
2019-07-05 14:55:46 +00:00
modules
Wire up new PacedSender code path.
2019-07-05 15:38:59 +00:00
p2p
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
2019-06-28 19:12:14 +00:00
pc
Reduce kMaxSimulcastStreams to 3
2019-07-04 15:54:36 +00:00
resources
Cleanup of resources from removed remote bitrate estimate test framework.
2019-06-18 10:22:01 +00:00
rtc_base
Make rtc_base/base_java public.
2019-07-03 14:44:46 +00:00
rtc_tools
Switch unpack_aecdump to ABSL_FLAG.
2019-07-04 14:19:40 +00:00
sdk
Android: Add error callback for GL_OUT_OF_MEMORY in EglRenderer
2019-07-05 13:04:17 +00:00
stats
[getStats] Implement "media-source" audio levels, fixing Chrome bug.
2019-07-04 08:13:45 +00:00
style-guide
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Sanity-check field trial string at initialization.
2019-06-11 14:11:06 +00:00
test
Add ability to create EmulatedNetworkNode from BuiltInNetworkBehaviorConfig
2019-07-05 12:43:17 +00:00
tools_webrtc
Remove android_tools deps
2019-07-02 06:48:30 +00:00
video
Disable VP9 2nd profile test for ios arm64
2019-07-03 14:48:16 +00:00
.clang-format
…
.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
Remove last mention of ortc from the codebase.
2019-05-25 07:28:05 +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
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
2019-06-03 16:28:34 +00:00
AUTHORS
Stun server should return XOR-MAPPED-ADDRESS/MAPPED-ADDRESS correctly
2019-06-28 19:12:14 +00:00
BUILD.gn
Move network emulation framework under test/network
2019-07-05 11:08:42 +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
Move kRtpCsrcSize from common_types.h to rtp_headers.h
2019-05-10 09:43:39 +00:00
DEPS
Roll chromium_revision 48f88caf2e..619b1dc518 (674992:675093)
2019-07-05 22:46:33 +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
Delete unused I420 "codec"
2018-12-18 12:30:58 +00:00
OWNERS
Add juberti@ to webrtc root owners
2019-05-17 18:11:58 +00:00
PATENTS
…
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
Convert file objects to strings, before passing to PresubmitNotifyResult
2019-06-26 11:31:21 +00:00
pylintrc
Fixing py lint errors
2018-07-23 15:28:48 +00:00
README.chromium
…
README.md
…
style-guide.md
Remove rule that discourages passing optional by const reference
2019-02-05 11:58:05 +00:00
WATCHLISTS
Remove myself from OWNERS in a few places.
2019-06-10 07:57:46 +00:00
webrtc.gni
Specify min_sdk_version for unittest apks also in GN configs
2019-06-26 07:18:15 +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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