platform-external-webrtc

Files

BUG=webrtc:6424

Review-Url: https://codereview.webrtc.org/2374033002
Cr-Commit-Position: refs/heads/master@{#14519}

2016-10-05 08:45:59 +00:00

androidjunit

Format all Java in WebRTC.

2016-09-29 11:12:51 +00:00

api

RTCStatsMember<bool> and RTCStatsMember<std::vector<bool>> added.

2016-10-04 21:37:15 +00:00

audio

Restarting channel when swapping AudioReceiveStreams in WebrtcVoE.

2016-10-04 09:45:54 +00:00

base

Delete profiler.h.

2016-10-05 08:45:59 +00:00

build

Roll chromium_revision f86fb54ec3..dd442d4812 (420104:421425)

2016-09-28 07:45:12 +00:00

call

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

common_audio

Test RTC_DCHECK_IS_ON instead of checking DCHECK_ALWAYS_ON everywhere

2016-10-04 20:47:02 +00:00

common_video

Delete webrtc::VideoFrame::CreateEmptyFrame.

2016-10-05 06:27:37 +00:00

examples

Disable TCPChannelClientTest.testConnectIPv6

2016-09-30 07:20:12 +00:00

libjingle

GN: Change rtc_source_set targets --> rtc_static_library

2016-09-23 07:38:58 +00:00

logging

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +00:00

media

Delete left-over SignalRenderFrame signal.

2016-10-05 07:14:02 +00:00

modules

Delete webrtc::VideoFrame::CreateEmptyFrame.

2016-10-05 06:27:37 +00:00

p2p

Delete httpportallocator.h.

2016-10-05 07:46:49 +00:00

Delete unused class cricket::MediaSinkInterface, and mediasink.h.

2016-10-04 12:33:53 +00:00

sdk

Remove Crit::Scope lock by using atomic bool property.

2016-09-30 07:21:28 +00:00

stats

RTCStatsMember<bool> and RTCStatsMember<std::vector<bool>> added.

2016-10-04 21:37:15 +00:00

system_wrappers

Sort #includes that got unsorted when gmock.h and gtest.h moved to webrtc/test/

2016-10-01 05:29:53 +00:00

test

Delete webrtc::VideoFrame::CreateEmptyFrame.

2016-10-05 06:27:37 +00:00

tools

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +00:00

video

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

voice_engine

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +00:00

.gitignore

…

BUILD.gn

Delete profiler.h.

2016-10-05 08:45:59 +00:00

call.h

This reverts commit 8eb37a39e79fe1098d3503dcb8c8c2d196203fed. Chrome now have its own implementation of TaskQueues that is based on Chrome threads.

2016-09-01 08:17:43 +00:00

codereview.settings

…

common_types.cc

Reland of Ignore Camera and Flip bits in CVO when parsing video rotation (patchset #1 id:1 of https://codereview.webrtc.org/2300323002/ )

2016-09-08 10:25:05 +00:00

common_types.h

Enable the -Wundef warning for clang

2016-09-29 00:42:08 +00:00

common.gyp

Moving/renaming webrtc/common.h.

2016-09-07 14:34:45 +00:00

config.cc

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

config.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

DEPS

Enable the -Wundef warning for clang

2016-09-29 00:42:08 +00:00

engine_configurations.h

Reland of move all reference to carbon api (patchset #1 id:1 of https://codereview.webrtc.org/2317343003/ )

2016-09-09 10:19:54 +00:00

LICENSE

…

LICENSE_THIRD_PARTY

Delete talk directory, and references from build_ios_libs.sh.

2016-09-02 10:07:13 +00:00

OWNERS

OWNERS: Make everyone able to change *.gn,*.gni files.

2016-09-09 12:51:48 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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supplement.gypi

Cleanup MIPS specific link configuration

2016-06-20 05:39:25 +00:00

transport.h

…

typedefs.h

[MIPS] Fix build issue for mips64el

2016-06-23 10:52:35 +00:00

video_decoder.h

…

video_encoder.h

Add codec name to CodecSpecificInfo and get the codec name stats from there instead.

2016-09-01 07:21:18 +00:00

video_frame.h

Delete webrtc::VideoFrame::CreateEmptyFrame.

2016-10-05 06:27:37 +00:00

video_receive_stream.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

video_send_stream.h

Rename FecConfig to UlpfecConfig in config.h.

2016-10-05 06:28:43 +00:00

webrtc_examples.gyp

Reland of name AppRTCDemo on Android and iOS to AppRTCMobile (patchset #1 id:1 of https://codereview.webrtc.org/2358133003/ )

2016-09-27 08:10:52 +00:00

webrtc_tests.gypi

GYP: Remove targets inside include_tests==1 that are converted to GN.

2016-09-15 11:57:39 +00:00

webrtc.gyp

Moved RtcEventLog files from call/ to logging/

2016-10-04 01:31:32 +00:00

README.chromium

Name: WebRTC
URL: http://www.webrtc.org
Version: 90
License: BSD
License File: LICENSE

Description:
WebRTC provides real time voice and video processing
functionality to enable the implementation of 
PeerConnection/MediaStream.

Third party code used in this project is described 
in the file LICENSE_THIRD_PARTY.