henrika@webrtc.org | 7a5615b | 2013-05-13 09:29:13 +0000 | [diff] [blame] | 1 | <!DOCTYPE html> |
| 2 | <html> |
| 3 | <head> |
| 4 | <meta charset="utf-8"> |
| 5 | <title>Audio effects with WebAudio in WebRTC</title> |
| 6 | <script type="text/javascript" src="../../base/adapter.js"></script> |
| 7 | <script> |
| 8 | var audioElement; |
| 9 | var buttonStart; |
| 10 | var buttonStop; |
| 11 | var localStream; |
| 12 | var pc1, pc2; |
| 13 | var display; |
| 14 | |
| 15 | var webAudio; |
| 16 | |
| 17 | // WebAudio helper class which takes care of the WebAudio related parts. |
| 18 | |
| 19 | function WebAudio() { |
| 20 | this.context = new webkitAudioContext(); |
| 21 | this.soundBuffer = null; |
| 22 | } |
| 23 | |
| 24 | WebAudio.prototype.start = function() { |
| 25 | this.filter = this.context.createBiquadFilter(); |
| 26 | this.filter.type = this.filter.HIGHPASS; |
| 27 | this.filter.frequency.value = 1500; |
| 28 | } |
| 29 | |
| 30 | WebAudio.prototype.applyFilter = function(stream) { |
| 31 | this.mic = this.context.createMediaStreamSource(stream); |
| 32 | this.mic.connect(this.filter); |
| 33 | this.peer = this.context.createMediaStreamDestination(); |
| 34 | this.filter.connect(this.peer); |
| 35 | return this.peer.stream; |
| 36 | } |
| 37 | |
| 38 | WebAudio.prototype.renderLocally = function(enabled) { |
| 39 | if (enabled) { |
| 40 | this.mic.connect(this.context.destination); |
| 41 | } else { |
| 42 | this.mic.disconnect(0); |
| 43 | this.mic.connect(this.filter); |
| 44 | } |
| 45 | } |
| 46 | |
| 47 | WebAudio.prototype.stop = function() { |
| 48 | this.mic.disconnect(0); |
| 49 | this.filter.disconnect(0); |
| 50 | mic = null; |
| 51 | peer = null; |
| 52 | } |
| 53 | |
| 54 | WebAudio.prototype.addEffect = function() { |
| 55 | var effect = this.context.createBufferSource(); |
| 56 | effect.buffer = this.soundBuffer; |
| 57 | if (this.peer) { |
| 58 | effect.connect(this.peer); |
| 59 | effect.start(0); |
| 60 | } |
| 61 | } |
| 62 | |
| 63 | WebAudio.prototype.loadCompleted = function() { |
| 64 | this.soundBuffer = this.context.createBuffer(this.request.response, true); |
| 65 | } |
| 66 | |
| 67 | WebAudio.prototype.loadSound = function(url) { |
| 68 | this.request = new XMLHttpRequest(); |
| 69 | this.request.open('GET', url, true); |
| 70 | this.request.responseType = 'arraybuffer'; |
| 71 | this.request.onload = this.loadCompleted.bind(this); |
| 72 | this.request.send(); |
| 73 | } |
| 74 | |
| 75 | // Global methods. |
| 76 | |
| 77 | function trace(txt) { |
| 78 | display.innerHTML += txt + "<br>"; |
| 79 | } |
| 80 | |
| 81 | function logEvent(e) { |
| 82 | console.log(e.type + ':' + e.target + ':' + e.target.id + ':muted=' + |
| 83 | e.target.muted); |
| 84 | } |
| 85 | |
| 86 | $ = function(id) { |
| 87 | return document.getElementById(id); |
| 88 | }; |
| 89 | |
| 90 | function start() { |
| 91 | webAudio.start(); |
| 92 | var constraints = {audio:true, video:false}; |
| 93 | getUserMedia(constraints, gotStream, gotStreamFailed); |
| 94 | buttonStart.disabled = true; |
| 95 | buttonStop.disabled = false; |
| 96 | } |
| 97 | |
| 98 | function stop() { |
| 99 | webAudio.stop(); |
| 100 | pc1.close(); |
| 101 | pc2.close(); |
| 102 | pc1 = null; |
| 103 | pc2 = null; |
| 104 | buttonStart.enabled = true; |
| 105 | buttonStop.enabled = false; |
| 106 | localStream.stop(); |
| 107 | } |
| 108 | |
| 109 | function gotStream(stream) { |
| 110 | audioTracks = stream.getAudioTracks(); |
| 111 | if (audioTracks.length == 1) { |
| 112 | console.log('gotStream({audio:true, video:false})'); |
| 113 | |
| 114 | var filteredStream = webAudio.applyFilter(stream); |
| 115 | |
| 116 | var servers = null; |
| 117 | pc1 = new webkitRTCPeerConnection(servers); |
| 118 | console.log('Created local peer connection object pc1'); |
| 119 | pc1.onicecandidate = iceCallback1; |
| 120 | pc2 = new webkitRTCPeerConnection(servers); |
| 121 | console.log('Created remote peer connection object pc2'); |
| 122 | pc2.onicecandidate = iceCallback2; |
| 123 | pc2.onaddstream = gotRemoteStream; |
| 124 | |
| 125 | pc1.addStream(filteredStream); |
| 126 | pc1.createOffer(gotDescription1); |
| 127 | |
| 128 | stream.onended = function() { |
| 129 | console.log('stream.onended'); |
| 130 | buttonStart.disabled = false; |
| 131 | buttonStop.disabled = true; |
| 132 | }; |
| 133 | |
| 134 | localStream = stream; |
| 135 | } else { |
| 136 | alert('The media stream contains an invalid amount of audio tracks.'); |
| 137 | stream.stop(); |
| 138 | } |
| 139 | } |
| 140 | |
| 141 | function gotStreamFailed(error) { |
| 142 | buttonStart.disabled = false; |
| 143 | buttonStop.disabled = true; |
| 144 | alert('Failed to get access to local media. Error code: ' + error.code); |
| 145 | } |
| 146 | |
| 147 | function forceOpus(sdp) { |
| 148 | // Remove all other codecs (not the video codecs though). |
| 149 | sdp = sdp.replace(/m=audio (\d+) RTP\/SAVPF.*\r\n/g, |
| 150 | 'm=audio $1 RTP/SAVPF 111\r\n'); |
| 151 | sdp = sdp.replace(/a=rtpmap:(?!111)\d{1,3} (?!VP8|red|ulpfec).*\r\n/g, ''); |
| 152 | return sdp; |
| 153 | } |
| 154 | |
| 155 | function gotDescription1(desc){ |
| 156 | console.log('Offer from pc1 \n' + desc.sdp); |
| 157 | var modifiedOffer = new RTCSessionDescription({type: 'offer', |
| 158 | sdp: forceOpus(desc.sdp)}); |
| 159 | pc1.setLocalDescription(modifiedOffer); |
| 160 | console.log('Offer from pc1 \n' + modifiedOffer.sdp); |
| 161 | pc2.setRemoteDescription(modifiedOffer); |
| 162 | pc2.createAnswer(gotDescription2); |
| 163 | } |
| 164 | |
| 165 | function gotDescription2(desc){ |
| 166 | pc2.setLocalDescription(desc); |
| 167 | console.log('Answer from pc2 \n' + desc.sdp); |
| 168 | pc1.setRemoteDescription(desc); |
| 169 | } |
| 170 | |
| 171 | function gotRemoteStream(e){ |
| 172 | attachMediaStream(audioElement, e.stream); |
| 173 | } |
| 174 | |
| 175 | function iceCallback1(event){ |
| 176 | if (event.candidate) { |
| 177 | pc2.addIceCandidate(new RTCIceCandidate(event.candidate)); |
| 178 | console.log('Local ICE candidate: \n' + event.candidate.candidate); |
| 179 | } |
| 180 | } |
| 181 | |
| 182 | function iceCallback2(event){ |
| 183 | if (event.candidate) { |
| 184 | pc1.addIceCandidate(new RTCIceCandidate(event.candidate)); |
| 185 | console.log('Remote ICE candidate: \n ' + event.candidate.candidate); |
| 186 | } |
| 187 | } |
| 188 | |
| 189 | function handleKeyDown(event) { |
| 190 | var keyCode = event.keyCode; |
| 191 | webAudio.addEffect(); |
| 192 | } |
| 193 | |
| 194 | function doMix(checkbox) { |
| 195 | webAudio.renderLocally(checkbox.checked); |
| 196 | } |
| 197 | |
| 198 | function onload() { |
| 199 | webAudio = new WebAudio(); |
| 200 | webAudio.loadSound('../sounds/Shamisen-C4.wav'); |
| 201 | |
| 202 | audioElement = $('audio'); |
| 203 | buttonStart = $('start'); |
| 204 | buttonStop = $('stop'); |
| 205 | display = $('display'); |
| 206 | |
| 207 | document.addEventListener('keydown', handleKeyDown, false); |
| 208 | |
| 209 | buttonStart.enabled = true; |
| 210 | buttonStop.disabled = true; |
| 211 | } |
| 212 | </script> |
| 213 | </head> |
| 214 | |
| 215 | <body onload='onload()'> |
| 216 | <h2>Capture microphone input and stream it out to a peer with a processing |
| 217 | effect applied to the audio.</h2> |
| 218 | <p>The audio stream is: <br><br> |
| 219 | o Recorded using <a href="http://www.html5audio.org/2012/09/live-audio-input-comes-to-googles-chrome-canary.html" |
| 220 | title="Live audio input comes to Google's Chrome Canary">live-audio |
| 221 | input.</a><br> |
| 222 | o Filtered using an HP filter with fc=1500 Hz.<br> |
| 223 | o Encoded using <a href="http://www.opus-codec.org/" title="Opus Codec"> |
| 224 | Opus.</a><br> |
| 225 | o Transmitted (in loopback) to remote peer using |
| 226 | <a href="http://dev.w3.org/2011/webrtc/editor/webrtc.html#rtcpeerconnection-interface" |
| 227 | title="RTCPeerConnection Interface">RTCPeerConnection</a> where it is decoded.<br> |
| 228 | o Finally, the received remote stream is used as source to an <audio> |
| 229 | tag and played out locally.<br> |
| 230 | <br>Press any key to add an effect to the transmitted audio while talking. |
| 231 | </p> |
| 232 | <p>Please note that: <br><br> |
| 233 | o Linux is currently not supported.<br> |
| 234 | o Sample rate and channel configuration must be the same for input and |
| 235 | output sides on Windows.<br> |
| 236 | o Only the Default microphone device can be used for capturing. |
| 237 | </p> |
| 238 | <p>For more information, see <a href="https://dvcs.w3.org/hg/audio/raw-file/tip/webaudio/webrtc-integration.html" |
| 239 | title="Example 3: Capture microphone input and stream it out to a peer with a processing effect applied to the audio"> |
| 240 | WebRTC integration with the Web Audio API.</a> |
| 241 | </p> |
| 242 | <style> |
| 243 | button { |
| 244 | font: 14px sans-serif; |
| 245 | padding: 8px; |
| 246 | } |
| 247 | </style> |
| 248 | <audio id="audio" autoplay controls></audio><br><br> |
| 249 | <button id="start" onclick="start()">Start</button> |
| 250 | <button id="stop" onclick="stop()">Stop</button><br><br> |
| 251 | Add local audio to output:<input id="mix" type="checkbox" onclick="doMix(this);"><br><br> |
| 252 | <pre id="display"></pre> |
| 253 | </body> |
| 254 | </html> |