webrtc_m130

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Patrik Höglund 3bb1194fff Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'"

This reverts commit d3070f43b19f503246be4ebad425d87568a71ce0.

Reason for revert: Need to re-enable h264 tests.

Original change's description:
> Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'
> 
> This doesn't change behavior at the moment because Chromium's
> 'proprietary_codecs' is already conditional on 'is_chrome_branded'
> but this guards WebRTC's default from upstream changes like
> https://chromium-review.googlesource.com/c/chromium/src/+/835010/6/build/config/features.gni
> 
> TBR=phoglund@webrtc.org
> 
> Bug: webrtc:8675
> Change-Id: Ic2ae311b5fc70a4d1ac1aefe4cc27574e4fcee40
> Reviewed-on: https://webrtc-review.googlesource.com/36321
> Commit-Queue: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
> Reviewed-by: Henrik Boström <hbos@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21452}

TBR=phoglund@webrtc.org,oprypin@webrtc.org,hbos@webrtc.org

# Not skipping CQ checks because original CL landed > 1 day ago.

Bug: webrtc:8675
Change-Id: I97e5b25fb638e9d4731ac9610f9f6009a3789578
Reviewed-on: https://webrtc-review.googlesource.com/62380
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Oleh Prypin <oprypin@webrtc.org>
Commit-Queue: Patrik Höglund <phoglund@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22476}

2018-03-16 14:34:21 +00:00

api

Adding a smooth transition from the startup phase parameter set in AEC3

2018-03-15 13:38:16 +00:00

audio

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

build_overrides

Add phoglund@ to various OWNERS and remove kjellander@

2017-10-19 09:21:12 +00:00

call

Fix race in DegradedCall::DestroyVideoSendStream

2018-03-16 10:23:47 +00:00

common_audio

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

common_video

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

data

…

examples

Final name changing of MediaStreamInterface.label() to id().

2018-03-14 20:30:52 +00:00

infra

Revert "Temporarily disable ios_api_framework."

2018-03-14 12:09:52 +00:00

logging

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

media

Delete encoder caching in WebRtcVideoSendStream.

2018-03-16 13:20:21 +00:00

modules

Delete payload_type from VCMEncoderDatabase and vcm::VideoSender.

2018-03-16 13:43:01 +00:00

ortc

Removed new calls on RtpTransportControllerSend.

2018-02-22 12:54:43 +00:00

p2p

Add configurable threshold for writability state update.

2018-03-13 18:54:03 +00:00

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

resources

Adding FourPeople_1280x720_30.yuv.

2018-02-12 15:55:00 +00:00

rtc_base

Remove unnecessary copies from AsyncInvoke

2018-03-16 13:09:18 +00:00

rtc_tools

Remove the public_deps to fileutils from test_support.

2018-03-16 09:06:27 +00:00

sdk

Fix podspec iOS version.

2018-03-16 09:30:27 +00:00

stats

Stats traversal algorithm added.

2018-03-12 10:54:09 +00:00

style-guide

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

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

system_wrappers

Fixing issues that affect newest versions of NDK.

2018-02-27 10:30:43 +00:00

test

Refactor payload types constants in CallTest

2018-03-16 11:57:47 +00:00

tools_webrtc

Revert "Reland "Split perf-test-specific resources in low_bandwidth_audio_test""

2018-03-15 13:52:47 +00:00

video

Delete payload_type from VCMEncoderDatabase and vcm::VideoSender.

2018-03-16 13:43:01 +00:00

.clang-format

Tune ObjC clang-format configuration

2017-05-11 09:14:18 +00:00

.git-blame-ignore-revs

…

.gitignore

Reland "Make it possible to run video_quality_loopback_test in swarming."

2018-01-23 13:03:17 +00:00

.gn

Re-enabling 'gn check' on //examples/*.

2018-02-19 15:07:45 +00:00

.vpython

vpython: Specify dependency on pywin32

2018-02-14 13:56:39 +00:00

AUTHORS

Expose a link-local network interfaces enumeration option

2018-02-06 19:12:04 +00:00

BUILD.gn

Removing definition of FEATURE_ENABLE_VOICEMAIL.

2018-02-19 15:51:24 +00:00

CODE_OF_CONDUCT.md

Add code of conduct to WebRTC repo

2017-05-16 12:09:13 +00:00

codereview.settings

Make Gerrit the default for WebRTC changes

2017-09-29 01:38:07 +00:00

common_types.cc

Delete the VideoCodec::plName string.

2018-03-06 11:17:41 +00:00

common_types.h

Reland "Delete VideoCodec::plName"

2018-03-13 09:32:28 +00:00

DEPS

Roll chromium_revision e0e02de5a7..ce851e47bd (543578:543685)

2018-03-16 14:25:41 +00:00

LICENSE

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

license_template.txt

…

LICENSE_THIRD_PARTY

Remove custom MD5 / SHA-1 implementations.

2018-02-19 15:03:35 +00:00

native-api.md

Remove legacy VoiceEngine.

2018-01-12 11:31:52 +00:00

OWNERS

Add back OWNERS entris that went missing

2018-03-06 10:31:11 +00:00

PATENTS

Moving src/webrtc into src/.

2017-09-15 04:25:06 +00:00

presubmit_test_mocks.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

presubmit_test.py

Using Change.BugsFromDescription to read CL bugs in PRESUBMIT checks.

2017-10-13 03:48:26 +00:00

PRESUBMIT.py

Presubmit: Fix bad file path in help text

2018-03-14 14:12:45 +00:00

pylintrc

Removing invalid-name from disabled pylint checks.

2017-10-11 08:06:49 +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

Style guide: State what version of C++ we should use

2018-03-15 10:32:16 +00:00

typedefs.h

Move FALLTHROUGH macro to a separate header, and give it an RTC_ prefix

2018-02-05 11:24:59 +00:00

WATCHLISTS

Move remaining traces of VoiceEngine

2018-01-17 13:27:47 +00:00

webrtc.gni

Revert "Add 'is_chrome_branded' guard to the default of 'rtc_use_h264'"

2018-03-16 14:34:21 +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

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Java 2.9%

C 2.2%

Objective-C++ 2%

Python 1.3%

Other 1%