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henrika@webrtc.org 7a5615bc84 New WebAudio-WebRTC demo.

Capture microphone input and stream it out to a peer with a processing effect applied to the audio.

The audio stream is: 

o Recorded using live-audio input.
o Filtered using an HP filter with fc=1500 Hz.
o Encoded using Opus.
o Transmitted (in loopback) to remote peer using RTCPeerConnection where it is decoded.
o Finally, the received remote stream is used as source to an <audio> tag and played out locally.

Press any key to add an effect to the transmitted audio while talking.

Please note that: 

o Linux is currently not supported.
o Sample rate and channel configuration must be the same for input and output sides on Windows.
o Only the Default microphone device can be used for capturing.

R=phoglund@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/1256004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@4006 4adac7df-926f-26a2-2b94-8c16560cd09d

2013-05-13 09:29:13 +00:00

data

ADM no longer reads PCM files from data/audio_device. Now uses the resource folder instead.

2012-08-20 09:44:59 +00:00

samples/js

New WebAudio-WebRTC demo.

2013-05-13 09:29:13 +00:00

third_party

Fix gflags compile error on x86 Android

2013-04-02 11:07:04 +00:00

tools

Updating perf