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chromium-webrtc-autoroll 70767cba42 Roll chromium_revision d65ce76c39..95ebb2b7ff (695395:695497) 
Change log: d65ce76c39..95ebb2b7ff
Full diff: d65ce76c39..95ebb2b7ff

Changed dependencies
* src/base: 33f39abb40..18d2a7bbf9
* src/build: f02b4d31e1..4ab78ab2d7
* src/ios: 1668c95d24..2dcc2d0a13
* src/testing: 73d164100d..3d7e946f5b
* src/third_party: f4e6f18b5a..5fa2200e5a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/369f46e374..050abd8dd5
* src/third_party/depot_tools: dbca865e55..cc6f585f05
* src/third_party/googletest/src: 3a45039862..33a0d4f6d7
* src/third_party/harfbuzz-ng/src: bbad1b8298..170b5dd856
* src/tools: e038a4025e..3f2e054770
DEPS diff: d65ce76c39..95ebb2b7ff/DEPS

No update to Clang.

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

Change-Id: I466a301558b148e744ca36f101b9822250cef5e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152500
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@{#29146}
2019-09-11 06:42:12 +00:00
api
Make NetworkStateEstimator injectable in RemoteBitrateEstimator
2019-09-10 18:47:36 +00:00
audio
Deprecate SingleThreadedTaskQueueForTesting class.
2019-09-03 10:31:30 +00:00
build_overrides
Remove crbug.com/904400 workaround.
2019-03-15 18:36:23 +00:00
call
Add option to enable retransmission for all temporal layers in the constructor for rtp_sender_video.
2019-09-09 15:39:23 +00:00
common_audio
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
common_video
Add new FrameRateEstimator utility class for more precis FPS estimation.
2019-08-14 12:15:06 +00:00
data
Remove old data files.
2018-10-05 14:40:21 +00:00
docs
Fixing some typos.
2019-09-10 10:03:50 +00:00
examples
Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM
2019-09-09 08:10:41 +00:00
logging
Delete deprecated rtc_event_log_factory_interface.h
2019-09-05 08:57:36 +00:00
media
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
modules
Make NetworkStateEstimator injectable in RemoteBitrateEstimator
2019-09-10 18:47:36 +00:00
p2p
Use obfuscated IPs in logging in p2p/ and pc/.
2019-09-06 21:51:56 +00:00
pc
Revert "Reland "Refactor SCTP data channels to use DataChannelTransportInterface.""
2019-09-10 17:52:36 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Revert "Removes string support in field trial parser."
2019-09-10 14:39:55 +00:00
rtc_tools
Revert "Always create output_dir in setup_apprtc.py."
2019-09-04 18:19:12 +00:00
sdk
Adds setAudio[Track/Record]StateCallback interfaces to the Java ADM
2019-09-09 08:10:41 +00:00
stats
Add qualityLimitationResolutionChanges stat
2019-09-09 15:22:57 +00:00
style-guide
…
system_wrappers
Move code related to VideoCodingModule to its own build target
2019-09-10 12:34:38 +00:00
test
Revert "Adds peer scenario connection interface."
2019-09-10 18:19:48 +00:00
tools_webrtc
Add license for android_ndk
2019-08-28 14:57:28 +00:00
video
Removes decoder thread fallback from VideoReceiveStream.
2019-09-10 16:27:48 +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
Change apprtc_webrtc_browsertest resource dir to avoid MAX_PATH.
2019-09-04 18:49:28 +00:00
.gn
Switch to compiling WebRTC -std=c++14 by default
2019-09-09 19:24:16 +00:00
.vpython
…
abseil-in-webrtc.md
Allowing buffering a LNTF (loss notification) feedback message in RTCPSender
2019-06-03 16:28:34 +00:00
AUTHORS
Revert "Implemented screen enumeration and selection for desktop capture under X11 using the X Resize and Rotate entension version 1.5."
2019-09-06 05:36:23 +00:00
BUILD.gn
Make the RtpHeaderParserImpl available to tests and tools only.
2019-08-29 15:56:40 +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
Format almost everything.
2019-07-08 13:45:15 +00:00
DEPS
Roll chromium_revision d65ce76c39..95ebb2b7ff (695395:695497)
2019-09-11 06:42:12 +00:00
ENG_REVIEW_OWNERS
…
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
…
presubmit_test.py
…
PRESUBMIT.py
Move docs about native code development into a repo directory.
2019-09-10 09:42:26 +00:00
pylintrc
…
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
Add saza to audio watchlists
2019-09-03 14:55:43 +00:00
webrtc.gni
Remove rtc_use_lto GN arg.
2019-08-20 14:00:49 +00:00
whitespace.txt
…

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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