platform-external-webrtc

Files

peah e0eae3cec6 This CL adds the basic framework for AEC3 in the audio processing module.

It will be followed by a number of other CLs that extends this framework.

BUG=webrtc:6018

Review-Url: https://codereview.webrtc.org/2567513003
Cr-Commit-Position: refs/heads/master@{#15593}

2016-12-14 09:16:28 +00:00

androidjunit

Format all Java in WebRTC.

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

api

Revert of Delete rtc::linked_ptr. Only use, in statstypes.h, replaced bu std::unique_ptr. (patchset #1 id:1 of https://codereview.webrtc.org/2567143003/ )

2016-12-14 08:06:38 +00:00

audio

Support external audio mixer in webrtc 2.

2016-12-13 22:06:35 +00:00

base

Delete unused class rtc::RegKey.

2016-12-14 08:30:33 +00:00

build

Move /webrtc/api/android files to /webrtc/sdk/android

2016-12-06 12:29:45 +00:00

call

Disable failing perf test on Android.

2016-12-09 14:15:22 +00:00

common_audio

Update common_audio/smoothing_filter.

2016-12-13 14:53:07 +00:00

common_video

Delete VideoFrame default constructor, and IsZeroSize method.

2016-12-13 14:08:39 +00:00

examples

Add a gtk3 port of peerconnection_client on Linux

2016-12-13 10:57:50 +00:00

libjingle

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

logging

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

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

media

Support external audio mixer in webrtc 2.

2016-12-13 22:06:35 +00:00

modules

This CL adds the basic framework for AEC3 in the audio processing module.

2016-12-14 09:16:28 +00:00

p2p

Fixing possible crash due to RefCountedChannel assignment operator.

2016-12-14 00:38:46 +00:00

Removing "crypto_required" from MediaContentDescription.

2016-12-13 19:29:16 +00:00

sdk

Delete VideoFrame default constructor, and IsZeroSize method.

2016-12-13 14:08:39 +00:00

stats

Collecting RTCIceCandidatePairStats.transport_id and improved unittests.

2016-11-30 09:50:36 +00:00

system_wrappers

Delete stl_util.h. Unused since cl https://codereview.webrtc.org/2447103002

2016-12-14 08:07:44 +00:00

test

Delete VideoFrame default constructor, and IsZeroSize method.

2016-12-13 14:08:39 +00:00

tools

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

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

video

Delete VideoFrame default constructor, and IsZeroSize method.

2016-12-13 14:08:39 +00:00

voice_engine

Delete voice_engine_configurations.h

2016-12-12 13:03:08 +00:00

.gitignore

…

BUILD.gn

Delete unused class rtc::RegKey.

2016-12-14 08:30:33 +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 Turn off error resilience for VP9 if no spatial or temporal layers are configured and NACK is enabl… (patchset #1 id:40001 of https://codereview.webrtc.org/2532053002/ )

2016-12-09 10:35:30 +00:00

config.cc

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

config.h

Keep all codec parameters in VideoReceiveStream::Decoder

2016-11-25 11:56:41 +00:00

DEPS

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

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

LICENSE

…

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

…

PRESUBMIT.py

…

README.chromium

…

transport.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

typedefs.h

Remove extra uses of basictypes.h.

2016-12-09 14:54:08 +00:00

video_decoder.h

Move VideoDecoder::Create() logic to separate internal video decoder factory

2016-12-01 08:27:35 +00:00

video_encoder.h

Move usage of QualityScaler to ViEEncoder.

2016-11-29 09:44:22 +00:00

video_frame.h

Delete VideoFrame default constructor, and IsZeroSize method.

2016-12-13 14:08:39 +00:00

video_receive_stream.h

Moved transport.h from webrtc/ to webrtc/api, created build target and updated WebRTC dependencies.

2016-11-28 15:02:19 +00:00

video_send_stream.h

Implement periodic bandwidth probing in application-limited region.

2016-11-28 21:11:24 +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.