platform-external-webrtc

Files

tina.legrand@webrtc.org d0d41498a3 Adding AUDIO application as default for Opus stereo

The Opus audio codec targets applications for pure conversations as well as other types of audio (e.g. music), and there are two different settings to use for this (VoIP and AUDIO). In the current implementation of Opus in WebRTC we use VoIP only.

I this CL I have changed default setting to AUDIO in the case of stereo, and kept VoIP as default in case of mono.

Next step is to add an API to choose application mode.

BUG=issue1239

Review URL: https://webrtc-codereview.appspot.com/1007006

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3319 4adac7df-926f-26a2-2b94-8c16560cd09d

2012-12-20 09:23:10 +00:00

build

Replace the last occurrence of .s with .h

2012-12-05 07:12:15 +00:00

common_audio

Porting ARM optimization from Android to ios.

2012-11-17 00:22:46 +00:00

common_video

Making I420VideoFrame ref-counted

2012-12-19 00:07:57 +00:00

modules

Adding AUDIO application as default for Opus stereo

2012-12-20 09:23:10 +00:00

system_wrappers

Reformatted sort.

2012-12-19 14:55:24 +00:00

test

Will now only require near-perfect PSNR and SSIM.

2012-12-13 10:15:06 +00:00

tools

Changed so that frame_cutter takes and argument where one can specify in which interval the frames should be deleted between the first frame to cut and the last frame to cut. This can for example be used to decrease the frame rate.

2012-12-13 14:46:40 +00:00

video_engine

Make protection method, filename and resolution configurable for FullStackTest.

2012-12-19 12:45:16 +00:00

voice_engine

Disabled flaky test.

2012-12-14 10:40:05 +00:00

common_types.h

Add number of inserted samples to NetEq statistics.

2012-12-14 00:06:18 +00:00

engine_configurations.h

First pass of MediaCodecDecoder which uses Android MediaCodec API.

2012-12-04 06:38:19 +00:00

LICENSE

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

LICENSE_THIRD_PARTY

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

PATENTS

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

README.chromium

Move src/ -> webrtc/

2012-10-22 18:19:23 +00:00

typedefs.h

Move src/ -> webrtc/

2012-10-22 18:19:23 +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.