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chromium-webrtc-autoroll 56d89da9cc Roll chromium_revision e25e764221..a87779d34b (694813:695071) 
Change log: e25e764221..a87779d34b
Full diff: e25e764221..a87779d34b

Changed dependencies
* src/base: 2de3a55fc4..2593ef8132
* src/build: ccaf07df5c..2d9fa32455
* src/buildtools: 74cfb57006..fce87d1a32
* src/buildtools/linux64: git_revision:152c5144ceed9592c20f0c8fd55769646077569b..git_revision:ad9e442d92dcd9ee73a557428cfc336b55cbd533
* src/buildtools/mac: git_revision:152c5144ceed9592c20f0c8fd55769646077569b..git_revision:ad9e442d92dcd9ee73a557428cfc336b55cbd533
* src/buildtools/win: git_revision:152c5144ceed9592c20f0c8fd55769646077569b..git_revision:ad9e442d92dcd9ee73a557428cfc336b55cbd533
* src/ios: fe8022e34d..6c7a089224
* src/testing: 5b7605a491..0fb5737633
* src/third_party: 13752878f8..82bfbbfe6a
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/3c6c057c3f..884c81e170
* src/third_party/depot_tools: efce0d1b76..e5641be5fe
* src/third_party/googletest/src: 3f05f651ae..3a45039862
* src/third_party/libvpx/source/libvpx: 305a5283c5..5a0242ba5c
* src/tools: 3a469b7fc1..7fdcd44406
DEPS diff: e25e764221..a87779d34b/DEPS

No update to Clang.

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

Change-Id: I4506d80579919afd51bed348040332266623daf2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/152300
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@{#29128}
2019-09-10 08:46:59 +00:00
api
Add placeholder target to move rtc_error out of the main API target.
2019-09-10 08:41:39 +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
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
AEC3: Downmix multichannel signals before delay estimation
2019-09-10 08:16:07 +00:00
p2p
Use obfuscated IPs in logging in p2p/ and pc/.
2019-09-06 21:51:56 +00:00
pc
Reland "Refactor SCTP data channels to use DataChannelTransportInterface."
2019-09-09 21:58:36 +00:00
resources
Use the AEC3 high-pass filter for the whole APM
2019-08-23 20:04:10 +00:00
rtc_base
Implement stable rate support in SimulcastRateAllocator
2019-09-06 11:17:30 +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
Add style guide rule about paired .h and .cc files
2018-03-14 13:02:35 +00:00
system_wrappers
Add helper functions to convert between integer milliseconds and fixed-point seconds.
2019-08-16 14:49:46 +00:00
test
Adds peer scenario connection interface.
2019-09-10 08:10:37 +00:00
tools_webrtc
Add license for android_ndk
2019-08-28 14:57:28 +00:00
video
MultiCodecReceiveTest: fix for flaky test.
2019-09-10 07:57:59 +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
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
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 e25e764221..a87779d34b (694813:695071)
2019-09-10 08:46:59 +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
…
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
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
Disable RunPythonTests on rtc_tools.
2019-08-06 12:48:33 +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
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
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
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Readme 255 MiB
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