platform-external-webrtc

Go to file

chromium-webrtc-autoroll 34aee67c11 Roll chromium_revision 514a543362..3ae2445b34 (686198:686310)

Change log: 514a543362..3ae2445b34
Full diff: 514a543362..3ae2445b34

Changed dependencies
* src/base: 4def23584c..08d654afee
* src/build: a2984b015f..b5e8d0f2d1
* src/third_party: 1d6494df55..45dec31484
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/5d664a903a..1c632e376b
* src/third_party/depot_tools: d6186f9936..3c814957ee
* src/third_party/icu: 682a230923..9a5af81f58
* src/tools: 7bb19f3198..bd69de11b0
DEPS diff: 514a543362..3ae2445b34/DEPS

No update to Clang.

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

Change-Id: I86f196fac80dd35a5233f178ca506d3400ef29fb
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/148940
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@{#28839}

2019-08-13 04:36:37 +00:00

api

audioproc_f: input AEC dump as string, output audio to vector

2019-08-12 09:17:36 +00:00

audio

Let StreamStatistician::GetReceiveStreamDataCounters return counters by value

2019-08-12 12:36:00 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Make fake network degradation work also for sent audio

2019-08-12 15:20:18 +00:00

common_audio

Add RTC_ prefix to non-standard format specifier macro "PRIdNS"

2019-08-07 13:36:05 +00:00

common_video

Format almost everything.

2019-07-08 13:45:15 +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

Remove MediaTransport from Call.

2019-08-08 10:58:57 +00:00

modules

Use unit types in RoundRobingPacketQueue and PacedSender

2019-08-12 17:10:21 +00:00

p2p

Delete deprecated rtc_event_log header

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

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

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

resources

Cleanup of resources from removed remote bitrate estimate test framework.

2019-06-18 10:22:01 +00:00

rtc_base

Delete placeholder file rtc_base/function_view.h

2019-08-09 09:39:32 +00:00

rtc_tools

Add RTC_ prefix to non-standard format specifier macro "PRIdNS"

2019-08-07 13:36:05 +00:00

sdk

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

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

stats

Format almost everything.

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

style-guide

Add style guide rule about paired .h and .cc files

2018-03-14 13:02:35 +00:00

system_wrappers

Format almost everything.

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

test

Add frames_in_flight metric to catch not delivered frames

2019-08-12 16:13:50 +00:00

tools_webrtc

Sync download_tools.py with changes in gclient_utils module.

2019-08-02 08:15:15 +00:00

video

Use recommended min bitrate limit provided by encoder.

2019-08-09 14:54:09 +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 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 514a543362..3ae2445b34 (686198:686310)

2019-08-13 04:36:37 +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

Migrate WebRTC test infra to ABSL_FLAG.

2019-07-19 06:54:04 +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%