platform-external-webrtc

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chromium-webrtc-autoroll 5245501eaf Roll chromium_revision 8538f0b743..acc73a9128 (691247:691348)

Change log: 8538f0b743..acc73a9128
Full diff: 8538f0b743..acc73a9128

Changed dependencies
* src/build: 6fd0308a6c..b3dc85ce23
* src/ios: 1c9086b5ea..fff61d7294
* src/testing: 866f47eb20..d4c8dcbb65
* src/third_party: 756c67f7d7..38590684a8
* src/third_party/googletest/src: 6a3d632f40..eb56ee5a28
* src/tools: d4a68e5f06..d133065acb
DEPS diff: 8538f0b743..acc73a9128/DEPS

No update to Clang.

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

Change-Id: I7449a72cf5eadf0afbdec0d2812096b713d64a26
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/150821
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@{#28994}

2019-08-28 22:34:07 +00:00

api

Add support for RTCTransportStats.selectedCandidatePairChanges

2019-08-28 13:22:08 +00:00

audio

Removes TransportSequenceNumberAllocator

2019-08-28 08:08:37 +00:00

build_overrides

Remove crbug.com/904400 workaround.

2019-03-15 18:36:23 +00:00

call

Removes TransportSequenceNumberAllocator

2019-08-28 08:08:37 +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

Implement JNI and objc implementation for Ice Candidate Pair Change event surfacing

2019-08-12 23:58:50 +00:00

logging

Delete deprecated rtc_event_log header

2019-08-07 10:58:17 +00:00

media

Fix the bug and reland "Make min video target bitrate configurable."

2019-08-27 11:12:12 +00:00

modules

Fix wrong-import-order pylint errors in quality_assessment.signal_processing module.

2019-08-28 14:48:28 +00:00

p2p

Add support for RTCTransportStats.selectedCandidatePairChanges

2019-08-28 13:22:08 +00:00

Only create a datagram RTP transport if the datagram transport should be used for RTP.

2019-08-28 21:15:18 +00:00

resources

Use the AEC3 high-pass filter for the whole APM

2019-08-23 20:04:10 +00:00

rtc_base

Reland "Delete mac_utils.h and mac_utils.cc"

2019-08-28 09:30:32 +00:00

rtc_tools

Remove rtcbot.

2019-08-19 09:26:08 +00:00

sdk

Revert "Detect leaks of TextureBufferImpl objects."

2019-08-28 12:35:04 +00:00

stats

Add support for RTCTransportStats.selectedCandidatePairChanges

2019-08-28 13:22:08 +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

Fix simulcast tests and PC framework for conference mode support

2019-08-28 13:23:13 +00:00

tools_webrtc

Add license for android_ndk

2019-08-28 14:57:28 +00:00

video

Fix simulcast tests and PC framework for conference mode support

2019-08-28 13:23:13 +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 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

Migrate WebRTC test infra to ABSL_FLAG.

2019-07-19 06:54:04 +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

Format almost everything.

2019-07-08 13:45:15 +00:00

DEPS

Roll chromium_revision 8538f0b743..acc73a9128 (691247:691348)

2019-08-28 22:34:07 +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

Remove myself from OWNERS in a few places.

2019-06-10 07:57:46 +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

Official web site: http://www.webrtc.org
Master source code repo: https://webrtc.googlesource.com/src
Samples and reference apps: https://github.com/webrtc
Mailing list: http://groups.google.com/group/discuss-webrtc
Continuous build: http://build.chromium.org/p/client.webrtc
Coding style guide
Code of conduct

Languages

C++ 88.6%

C 3.3%

Java 3%

Objective-C++ 1.9%

Python 1.9%

Other 1%