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<h1><a href="//webrtc.github.io/samples/" title="WebRTC samples homepage">WebRTC samples</a> <span>Web Audio input</span></h1>
<audio autoplay controls></audio>
<div id="options">
<input type="checkbox" id="renderLocally" disabled><label for="renderLocally">Add local audio to output</label>
</div>
<div id="buttons">
<button id="start">Start</button>
<button id="stop" disabled>Stop</button>
</div>
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<p>Capture microphone input and stream it to a peer with processing applied to the audio.</p>
<p>The audio stream is:</p>
<ul>
<li>Recorded using <a href="http://www.html5audio.org/2012/09/live-audio-input-comes-to-googles-chrome-canary.html" title="Live audio input comes to Google's Chrome Canary">live audio input</a>.</li>
<li>Filtered using an HP filter with fc=1500 Hz.</li>
<li>Encoded using <a href="http://www.opus-codec.org/" title="Opus Codec">Opus</a>.</li>
<li>Transmitted (in loopback) to a remote peer using <a href="http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcpeerconnection-interface" title="RTCPeerConnection Interface">RTCPeerConnection</a> where it is decoded.</li>
<li>Finally, the received remote stream is used as source to an <audio> element and played out locally. </li>
</ul>
<p>Press any key to add an effect to the transmitted audio while talking.</p>
<p>Please note that:</p>
<ul>
<li>Linux is currently not supported.</li>
<li>Sample rate and channel configuration must be the same for input and
output sides on Windows.</li>
<li>Only the default microphone device can be used for capturing.</li>
</ul>
<p>For more information, see <a href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/webrtc-integration.html" title="Example 3: Capture microphone input and stream it out to a peer with a processing effect applied to the audio">WebRTC integration with the Web Audio API</a>.</p>
<a href="https://github.com/webrtc/samples/tree/gh-pages/src/content/peerconnection/webaudio-input" title="View source for this page on GitHub" id="viewSource">View source on GitHub</a>
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