platform-external-webrtc

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chromium-webrtc-autoroll 2923d768f0 Roll chromium_revision c049034cc7..6c5bbd86c3 (671848:671957)

Change log: c049034cc7..6c5bbd86c3
Full diff: c049034cc7..6c5bbd86c3

Changed dependencies
* src/build: a703fac545..fa4e56cd54
* src/ios: 11fdd4a29d..ce91b95565
* src/testing: c0b90eb408..1cd8c31743
* src/third_party: caeb94df80..ea1355f5cf
* src/third_party/catapult: https://chromium.googlesource.com/catapult.git/+log/304e5101f3..8db999ee35
* src/third_party/depot_tools: dd2737e2b5..8812b932c9
* src/third_party/nasm: c8b248039e..f564874f49
* src/tools: 87ce84be27..bcea6a313c
DEPS diff: c049034cc7..6c5bbd86c3/DEPS

No update to Clang.

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

Change-Id: If68e5119fc88235edd4147ae94f45cb06c2921e7
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/143660
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@{#28359}

2019-06-25 05:37:44 +00:00

api

Reland "Cleanup of RTP references in GoogCC implementation."

2019-06-24 09:10:52 +00:00

audio

Avoid triggering a false error logging when using encryptor and sending DTX.

2019-06-24 10:55:06 +00:00

build_overrides

Remove crbug.com/904400 workaround.

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

call

Switch from RtpPacketSender to RtpPacketPacer interface usage.

2019-06-24 10:46:06 +00:00

common_audio

Refactor WavWriter to use FileWrapper rather than PlatformFile

2019-06-14 10:18:28 +00:00

common_video

Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.

2019-06-20 10:24:29 +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

Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest

2019-06-18 17:10:06 +00:00

logging

Minor rtc_event_log_impl cleanup.

2019-06-20 11:57:54 +00:00

media

Add plumbing of RtpPacketInfos to each VideoFrame as input for SourceTracker.

2019-06-20 10:24:29 +00:00

modules

Removes unused PacketFeedbackComparator

2019-06-24 11:02:35 +00:00

p2p

Add logging and edit the field trial name for piggyback ICE check

2019-06-19 22:35:47 +00:00

[getStats] Make remote-inbound-rtp.ssrc match outbound-rtp.ssrc.

2019-06-24 15:26:02 +00:00

resources

Cleanup of resources from removed remote bitrate estimate test framework.

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

rtc_base

BalancedDegradationSettings: Add option to configure QP thresholds.

2019-06-24 09:32:51 +00:00

rtc_tools

Switch frame_analyzer to ABSL_FLAG.

2019-06-25 05:20:52 +00:00

sdk

Revert "Raise IllegalStateException for calls to retain() or release() on zero ref count"

2019-06-19 08:50:53 +00:00

stats

Implement QualityLimitationReasonTracker and expose "reason".

2019-05-28 16:23:55 +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

Improvements to scenario video stats for scenario tests.

2019-06-24 15:42:42 +00:00

tools_webrtc

Roll chromium_revision 6ae0f0cd4c..bf62d746a4 (669703:669828) + fix AndroidManifest

2019-06-18 17:10:06 +00:00

video

BalancedDegradationSettings: Add option to configure QP thresholds.

2019-06-24 09:32:51 +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 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

Import proto_library.gni when rtc_enable_protobuf is true

2019-02-27 09:56:42 +00:00

BUILD.gn

Cleanup of resources from removed remote bitrate estimate test framework.

2019-06-18 10:22:01 +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

Move kRtpCsrcSize from common_types.h to rtp_headers.h

2019-05-10 09:43:39 +00:00

DEPS

Roll chromium_revision c049034cc7..6c5bbd86c3 (671848:671957)

2019-06-25 05:37:44 +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

Removes legacy bitrate controller.

2019-06-11 13:16:05 +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

Do not use libevent when targeting wasm.

2019-06-20 09:40:57 +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%