platform-external-webrtc

Files

alessiob 36e6a8f1bd WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/).

The unit test ConversationalSpeechTest.MultiEndCallWavReaderAdaptorSine uses CreateSineWavFile() and writes temporary wav files that are used for test (deleted only if the test passes).

BUG=webrtc:7218

Review-Url: https://codereview.webrtc.org/2774423005
Cr-Commit-Position: refs/heads/master@{#17608}

2017-04-10 07:53:53 +00:00

api

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

audio

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

base

Revert of Delete class ScopedPtrCollection. Replaced with vector of unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2808463002/ )

2017-04-10 07:31:33 +00:00

call

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

common_audio

iSAC fix entropy coder: Recently added DCHECK could in fact trigger

2017-03-13 12:28:47 +00:00

common_video

Delete unneeded includes of deprecated system_wrappers include files.

2017-03-30 07:31:15 +00:00

examples

Use new RTCCameraVideoCapturer in AppRTCMobile.

2017-04-05 19:17:48 +00:00

logging

Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )

2017-04-07 07:59:12 +00:00

media

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

modules

WavReaderAdaptor is a simple adaptor of the existing class WavReader from webrtc/common_audio/wav_file.h. The adaptor was mainly needed to use dependency injection and easily test the MultiEndCall class (see https://codereview.webrtc.org/2761853002/).

2017-04-10 07:53:53 +00:00

ortc

Disable the ORTC integration tests on TSan.

2017-03-28 20:45:18 +00:00

p2p

Fix compilation issues of std::unique_ptr

2017-04-09 16:09:06 +00:00

Fix compilation issues of std::unique_ptr

2017-04-09 16:09:06 +00:00

sdk

Move AndroidVideoTrackSourceObserver from API to src

2017-04-06 11:31:26 +00:00

stats

Move RTCOutboundRTPStreamStats.roundTripTime to inbound, don't collect.

2017-03-01 09:02:45 +00:00

system_wrappers

Delete system_wrappers logging facility.

2017-04-06 10:53:06 +00:00

test

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

tools

Resolve cyclic dependency between audio network adaptor and event log api

2017-04-06 12:59:10 +00:00

video

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

voice_engine

Added the GetSources() to the RtpReceiverInterface and implemented

2017-04-07 17:57:22 +00:00

.gitignore

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BUILD.gn

Reland of Loosening the coupling between WebRTC and //third_party/protobuf (patchset #1 id:1 of https://codereview.webrtc.org/2786363002/ )

2017-04-07 07:59:12 +00:00

call.h

Moved call.h and most of api/call/* into call/

2016-12-07 12:53:04 +00:00

codereview.settings

…

common_types.cc

Reject XR TargetBitrate items with unsupported layer indices

2016-12-06 14:09:00 +00:00

common_types.h

Revert of Supporting 48kHz PCM file. (patchset #1 id:1 of https://codereview.webrtc.org/2790493004/ )

2017-03-31 23:32:28 +00:00

config.cc

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

config.h

Adding "adapter" ORTC objects on top of ChannelManager/BaseChannel/etc.

2017-02-26 02:15:09 +00:00

DEPS

Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )

2017-04-05 10:02:20 +00:00

LICENSE

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LICENSE_THIRD_PARTY

Delete unused spreadsort implementation.

2016-12-05 11:03:26 +00:00

OWNERS

Cleanup unused rules in webrtc/DEPS + add kjellander to OWNERS for it

2016-11-29 10:52:22 +00:00

PATENTS

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PRESUBMIT.py

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README.chromium

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typedefs.h

Make WARN_UNUSED_RESULT a no-op on gcc

2017-04-06 11:32:27 +00:00

video_decoder.h

Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )

2017-04-05 10:02:20 +00:00

video_encoder.h

Reland of Move video_encoder.h and video_decoder.h to /api and create GN targets for them (patchset #1 id:1 of https://codereview.webrtc.org/2794033002/ )

2017-04-05 10:02:20 +00:00

video_frame.h

Move VideoFrame and related declarations to webrtc/api/video.

2017-01-10 15:44:26 +00:00

video_receive_stream.h

Add |protected_by_flexfec| flag to VideoReceiveStream::Config.

2017-03-09 14:13:20 +00:00

video_send_stream.h

Add number of quality adapt changes to VideoSendStream stats.

2017-04-04 12:51:49 +00:00

webrtc.gni

Reland of Setting is_component_build to false by default (patchset #1 id:1 of https://codereview.webrtc.org/2731703004/ )

2017-03-04 03:41:59 +00:00

whitespace.txt

Moving whitespace file up by one folder

2017-01-24 13:19:42 +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.