refactor all the audio stuff into classes and a separate file
This commit is contained in:
parent
91b8c55de9
commit
13d7686258
@ -28,6 +28,7 @@
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<script src="static/lib/jquery-3.2.1.min.js"></script>
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<script src="static/lib/jquery.nanoscroller.js"></script>
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<script src="static/lib/bookmarks.js"></script>
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<script src="static/lib/AudioEngine.js"></script>
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<link rel="stylesheet" type="text/css" href="static/lib/nanoscroller.css" />
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<link rel="stylesheet" type="text/css" href="static/css/openwebrx.css" />
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<meta charset="utf-8">
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@ -209,7 +210,7 @@
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</div>
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</div>
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</div>
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<div id="openwebrx-big-grey" onclick="iosPlayButtonClick();">
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<div id="openwebrx-big-grey" onclick="playButtonClick();">
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<div id="openwebrx-play-button-text">
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<img id="openwebrx-play-button" src="static/gfx/openwebrx-play-button.png" />
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<br /><br />Start OpenWebRX
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214
htdocs/lib/AudioEngine.js
Normal file
214
htdocs/lib/AudioEngine.js
Normal file
@ -0,0 +1,214 @@
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// this controls if the new AudioWorklet API should be used if available.
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// the engine will still fall back to the ScriptProcessorNode if this is set to true but not available in the browser.
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var useAudioWorklets = true;
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function AudioEngine(maxBufferLength, audioReporter) {
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this.audioReporter = audioReporter;
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this.resetStats();
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var ctx = window.AudioContext || window.webkitAudioContext;
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if (!ctx) {
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return;
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}
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this.audioContext = new ctx();
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this.allowed = this.audioContext.state === 'running';
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this.started = false;
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this.audioCodec = new sdrjs.ImaAdpcm();
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this.compression = 'none';
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this.setupResampling();
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this.resampler = new sdrjs.RationalResamplerFF(this.resamplingFactor, 1);
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this.maxBufferSize = maxBufferLength * this.getSampleRate();
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}
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AudioEngine.prototype.start = function(callback) {
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var me = this;
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if (me.resamplingFactor === 0) return; //if failed to find a valid resampling factor...
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if (me.started) {
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if (callback) callback(false);
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return;
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}
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me.audioContext.resume().then(function(){
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me.allowed = me.audioContext.state === 'running';
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if (!me.allowed) {
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if (callback) callback(false);
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return;
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}
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me.started = true;
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me.gainNode = me.audioContext.createGain();
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me.gainNode.connect(me.audioContext.destination);
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if (useAudioWorklets && me.audioContext.audioWorklet) {
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me.audioContext.audioWorklet.addModule('static/lib/AudioProcessor.js').then(function(){
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me.audioNode = new AudioWorkletNode(me.audioContext, 'openwebrx-audio-processor', {
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numberOfInputs: 0,
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numberOfOutputs: 1,
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outputChannelCount: [1],
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processorOptions: {
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maxBufferSize: me.maxBufferSize
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}
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});
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me.audioNode.connect(me.gainNode);
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me.audioNode.port.addEventListener('message', function(m){
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var json = JSON.parse(m.data);
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if (typeof(json.buffersize) !== 'undefined') {
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me.audioReporter(json);
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}
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});
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me.audioNode.port.start();
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if (callback) callback(true, 'AudioWorklet');
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});
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} else {
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me.audioBuffers = [];
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if (!AudioBuffer.prototype.copyToChannel) { //Chrome 36 does not have it, Firefox does
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AudioBuffer.prototype.copyToChannel = function (input, channel) //input is Float32Array
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{
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var cd = this.getChannelData(channel);
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for (var i = 0; i < input.length; i++) cd[i] = input[i];
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}
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}
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var bufferSize;
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if (me.audioContext.sampleRate < 44100 * 2)
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bufferSize = 4096;
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else if (me.audioContext.sampleRate >= 44100 * 2 && me.audioContext.sampleRate < 44100 * 4)
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bufferSize = 4096 * 2;
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else if (me.audioContext.sampleRate > 44100 * 4)
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bufferSize = 4096 * 4;
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function audio_onprocess(e) {
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var total = 0;
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var out = new Float32Array(bufferSize);
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while (me.audioBuffers.length) {
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var b = me.audioBuffers.shift();
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var newLength = total + b.length;
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// not enough space to fit all data, so splice and put back in the queue
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if (newLength > bufferSize) {
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var tokeep = b.subarray(0, bufferSize - total);
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out.set(tokeep, total);
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var tobuffer = b.subarray(bufferSize - total, b.length);
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me.audioBuffers.unshift(tobuffer);
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break;
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} else {
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out.set(b, total);
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}
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total = newLength;
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}
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e.outputBuffer.copyToChannel(out, 0);
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}
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//on Chrome v36, createJavaScriptNode has been replaced by createScriptProcessor
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var method = 'createScriptProcessor';
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if (me.audioContext.createJavaScriptNode) {
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method = 'createJavaScriptNode';
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}
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me.audioNode = me.audioContext[method](bufferSize, 0, 1);
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me.audioNode.onaudioprocess = audio_onprocess;
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me.audioNode.connect(me.gainNode);
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if (callback) callback(true, 'ScriptProcessorNode');
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}
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setInterval(me.reportStats.bind(me), 1000);
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});
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}
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AudioEngine.prototype.isAllowed = function() {
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return this.allowed;
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}
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AudioEngine.prototype.reportStats = function() {
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var stats = {}
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if (this.audioNode.port) {
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this.audioNode.port.postMessage(JSON.stringify({cmd:'getBuffers'}));
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} else {
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stats.buffersize = this.getBuffersize();
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}
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stats.audioRate = this.stats.audioSamples;
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var elapsed = new Date() - this.stats.startTime;
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stats.audioByteRate = this.stats.audioBytes * 1000 / elapsed
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this.audioReporter(stats);
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// sample rate is just measuring the last seconds
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this.stats.audioSamples = 0;
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}
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AudioEngine.prototype.resetStats = function() {
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this.stats = {
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startTime: new Date(),
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audioBytes: 0,
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audioSamples: 0
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};
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}
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AudioEngine.prototype.setupResampling = function() { //both at the server and the client
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var output_range_max = 12000;
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var output_range_min = 8000;
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var targetRate = this.audioContext.sampleRate;
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var i = 1;
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while (true) {
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var audio_server_output_rate = Math.floor(targetRate / i);
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if (audio_server_output_rate < output_range_min) {
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this.resamplingFactor = 0;
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this.outputRate = 0;
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divlog('Your audio card sampling rate (' + targetRate + ') is not supported.<br />Please change your operating system default settings in order to fix this.', 1);
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break;
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} else if (audio_server_output_rate >= output_range_min && audio_server_output_rate <= output_range_max) {
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this.resamplingFactor = i;
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this.outputRate = audio_server_output_rate;
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break; //okay, we're done
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}
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i++;
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}
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}
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AudioEngine.prototype.getOutputRate = function() {
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return this.outputRate;
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}
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AudioEngine.prototype.getSampleRate = function() {
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return this.audioContext.sampleRate;
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}
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AudioEngine.prototype.pushAudio = function(data) {
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if (!this.audioNode) return;
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this.stats.audioBytes += data.byteLength;
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var buffer;
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if (this.compression === "adpcm") {
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//resampling & ADPCM
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buffer = this.audioCodec.decode(new Uint8Array(data));
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} else {
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buffer = new Int16Array(data);
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}
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buffer = this.resampler.process(sdrjs.ConvertI16_F(buffer));
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this.stats.audioSamples += buffer.length;
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if (this.audioNode.port) {
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// AudioWorklets supported
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this.audioNode.port.postMessage(buffer);
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} else {
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// silently drop excess samples
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if (this.getBuffersize() + buffer.length <= this.maxBufferSize) {
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this.audioBuffers.push(buffer);
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}
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}
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}
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AudioEngine.prototype.setCompression = function(compression) {
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this.compression = compression;
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}
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AudioEngine.prototype.setVolume = function(volume) {
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this.gainNode.gain.value = volume;
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}
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AudioEngine.prototype.getBuffersize = function() {
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// only available when using ScriptProcessorNode
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if (!this.audioBuffers) return 0;
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return this.audioBuffers.map(function(b){ return b.length; }).reduce(function(a, b){ return a + b; }, 0);
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}
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@ -1,9 +1,8 @@
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class OwrxAudioProcessor extends AudioWorkletProcessor {
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constructor(options){
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super(options);
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this.maxLength = options.processorOptions.maxLength;
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// initialize ringbuffer, make sure it aligns with the expected buffer size of 128
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this.bufferSize = Math.round(sampleRate * this.maxLength / 128) * 128
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this.bufferSize = Math.round(options.processorOptions.maxBufferSize / 128) * 128
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this.audioBuffer = new Float32Array(this.bufferSize);
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this.inPos = 0;
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this.outPos = 0;
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@ -30,16 +30,12 @@ function arrayBufferToString(buf) {
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var bandwidth;
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var center_freq;
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var audio_buffer_current_size_debug = 0;
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var audio_buffer_current_count_debug = 0;
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var fft_size;
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var fft_fps;
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var fft_compression = "none";
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var fft_codec = new sdrjs.ImaAdpcm();
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var audio_compression = "none";
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var waterfall_setup_done = 0;
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var secondary_fft_size;
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var audio_allowed;
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var rx_photo_state = 1;
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function e(what) {
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@ -108,7 +104,7 @@ function style_value(of_what, which) {
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}
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function updateVolume() {
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gainNode.gain.value = parseFloat(e("openwebrx-panel-volume").value) / 100;
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audioEngine.setVolume(parseFloat(e("openwebrx-panel-volume").value) / 100);
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}
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function toggleMute() {
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@ -406,8 +402,8 @@ function Demodulator_default_analog(offset_frequency, subtype) {
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this.subtype = subtype;
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this.filter = {
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min_passband: 100,
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high_cut_limit: (audio_server_output_rate / 2) - 1, //audio_context.sampleRate/2,
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low_cut_limit: (-audio_server_output_rate / 2) + 1 //-audio_context.sampleRate/2
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high_cut_limit: (audioEngine.getOutputRate() / 2) - 1,
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low_cut_limit: (-audioEngine.getOutputRate() / 2) + 1
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};
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//Subtypes only define some filter parameters and the mod string sent to server,
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//so you may set these parameters in your custom child class.
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@ -689,7 +685,8 @@ function scale_px_from_freq(f, range) {
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function get_visible_freq_range() {
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var out = {};
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var fcalc = function (x) {
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return Math.round(((-zoom_offset_px + x) / canvases[0].clientWidth) * bandwidth) + (center_freq - bandwidth / 2);
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var canvasWidth = canvas_container.clientWidth * zoom_levels[zoom_level];
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return Math.round(((-zoom_offset_px + x) / canvasWidth) * bandwidth) + (center_freq - bandwidth / 2);
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};
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out.start = fcalc(0);
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out.center = fcalc(canvas_container.clientWidth / 2);
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@ -1063,30 +1060,8 @@ function resize_waterfall_container(check_init) {
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}
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var audio_server_output_rate = 11025;
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var audio_client_resampling_factor = 4;
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function audio_calculate_resampling(targetRate) { //both at the server and the client
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var output_range_max = 12000;
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var output_range_min = 8000;
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var i = 1;
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while (true) {
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audio_server_output_rate = Math.floor(targetRate / i);
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if (audio_server_output_rate < output_range_min) {
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audio_client_resampling_factor = audio_server_output_rate = 0;
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divlog("Your audio card sampling rate (" + targetRate.toString() + ") is not supported.<br />Please change your operating system default settings in order to fix this.", 1);
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}
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if (audio_server_output_rate >= output_range_min && audio_server_output_rate <= output_range_max) break; //okay, we're done
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i++;
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}
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audio_client_resampling_factor = i;
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console.log("audio_calculate_resampling() :: " + audio_client_resampling_factor.toString() + ", " + audio_server_output_rate.toString());
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}
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var debug_ws_data_received = 0;
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var debug_ws_time_start = 0;
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var debug_ws_time_start;
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var max_clients_num = 0;
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var client_num = 0;
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var currentprofile;
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@ -1096,7 +1071,7 @@ var COMPRESS_FFT_PAD_N = 10; //should be the same as in csdr.c
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function on_ws_recv(evt) {
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if (typeof evt.data === 'string') {
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// text messages
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debug_ws_data_received += evt.data.length / 1000;
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debug_ws_data_received += evt.data.length;
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if (evt.data.substr(0, 16) === "CLIENT DE SERVER") {
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divlog("Server acknowledged WebSocket connection.");
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@ -1106,19 +1081,20 @@ function on_ws_recv(evt) {
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switch (json.type) {
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case "config":
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var config = json['value'];
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window.waterfall_colors = config['waterfall_colors'];
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window.waterfall_min_level_default = config['waterfall_min_level'];
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window.waterfall_max_level_default = config['waterfall_max_level'];
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window.waterfall_auto_level_margin = config['waterfall_auto_level_margin'];
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waterfall_colors = config['waterfall_colors'];
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waterfall_min_level_default = config['waterfall_min_level'];
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waterfall_max_level_default = config['waterfall_max_level'];
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waterfall_auto_level_margin = config['waterfall_auto_level_margin'];
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waterfallColorsDefault();
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window.starting_mod = config['start_mod'];
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window.starting_offset_frequency = config['start_offset_freq'];
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starting_mod = config['start_mod'];
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starting_offset_frequency = config['start_offset_freq'];
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bandwidth = config['samp_rate'];
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center_freq = config['center_freq'] + config['lfo_offset'];
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fft_size = config['fft_size'];
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fft_fps = config['fft_fps'];
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audio_compression = config['audio_compression'];
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var audio_compression = config['audio_compression'];
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audioEngine.setCompression(audio_compression);
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divlog("Audio stream is " + ((audio_compression === "adpcm") ? "compressed" : "uncompressed") + ".");
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fft_compression = config['fft_compression'];
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divlog("FFT stream is " + ((fft_compression === "adpcm") ? "compressed" : "uncompressed") + ".");
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@ -1129,20 +1105,14 @@ function on_ws_recv(evt) {
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mathbox_waterfall_history_length = config['mathbox_waterfall_history_length'];
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waterfall_init();
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audio_preinit();
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initialize_demodulator();
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bookmarks.loadLocalBookmarks();
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if (audio_allowed) {
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if (audio_initialized) {
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initialize_demodulator();
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} else {
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audio_init();
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}
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}
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waterfall_clear();
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currentprofile = config['profile_id'];
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$('#openwebrx-sdr-profiles-listbox').val(currentprofile);
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break;
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case "secondary_config":
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var s = json['value'];
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@ -1222,7 +1192,7 @@ function on_ws_recv(evt) {
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}
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} else if (evt.data instanceof ArrayBuffer) {
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// binary messages
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debug_ws_data_received += evt.data.byteLength / 1000;
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debug_ws_data_received += evt.data.byteLength;
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var type = new Uint8Array(evt.data, 0, 1)[0];
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var data = evt.data.slice(1);
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@ -1247,15 +1217,7 @@ function on_ws_recv(evt) {
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break;
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case 2:
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// audio data
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var audio_data;
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if (audio_compression === "adpcm") {
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audio_data = new Uint8Array(data);
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} else {
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audio_data = new Int16Array(data);
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}
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audio_prepare(audio_data);
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audio_buffer_current_size_debug += audio_data.length;
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if (!(ios || is_chrome) && (audio_initialized === 0)) audio_init();
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audioEngine.pushAudio(data);
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break;
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case 3:
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// secondary FFT
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@ -1498,8 +1460,13 @@ function on_ws_opened() {
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ws.send("SERVER DE CLIENT client=openwebrx.js type=receiver");
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divlog("WebSocket opened to " + ws.url);
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debug_ws_data_received = 0;
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debug_ws_time_start = new Date().getTime();
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debug_ws_time_start = new Date();
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reconnect_timeout = false;
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ws.send(JSON.stringify({
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"type": "dspcontrol",
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"action": "start",
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"params": {"output_rate": audioEngine.getOutputRate()}
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}));
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}
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var was_error = 0;
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@ -1517,114 +1484,11 @@ function divlog(what, is_error) {
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nano.nanoScroller({scroll: 'bottom'});
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}
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var audio_context;
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var audio_initialized = 0;
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var gainNode;
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var volumeBeforeMute = 100.0;
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var mute = false;
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var audio_resampler;
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var audio_codec = new sdrjs.ImaAdpcm();
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var audio_node;
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// Optimalise these if audio lags or is choppy:
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var audio_buffer_size;
|
||||
var audio_buffer_maximal_length_sec = 1; //actual number of samples are calculated from sample rate
|
||||
var audio_buffer_decrease_to_on_overrun_sec = 0.8;
|
||||
var audio_flush_interval_ms = 500; //the interval in which audio_flush() is called
|
||||
|
||||
var audio_buffers = [];
|
||||
var audio_last_output_buffer;
|
||||
|
||||
function audio_prepare(data) {
|
||||
if (!audio_node) return;
|
||||
var buffer = data;
|
||||
if (audio_compression === "adpcm") {
|
||||
//resampling & ADPCM
|
||||
buffer = audio_codec.decode(buffer);
|
||||
}
|
||||
buffer = audio_resampler.process(sdrjs.ConvertI16_F(buffer));
|
||||
if (audio_node.port) {
|
||||
// AudioWorklets supported
|
||||
audio_node.port.postMessage(buffer);
|
||||
} else {
|
||||
audio_buffers.push(buffer);
|
||||
}
|
||||
}
|
||||
|
||||
if (!AudioBuffer.prototype.copyToChannel) { //Chrome 36 does not have it, Firefox does
|
||||
AudioBuffer.prototype.copyToChannel = function (input, channel) //input is Float32Array
|
||||
{
|
||||
var cd = this.getChannelData(channel);
|
||||
for (var i = 0; i < input.length; i++) cd[i] = input[i];
|
||||
}
|
||||
}
|
||||
|
||||
function audio_onprocess(e) {
|
||||
var total = 0;
|
||||
var out = new Float32Array(audio_buffer_size);
|
||||
while (audio_buffers.length) {
|
||||
var b = audio_buffers.shift();
|
||||
var newLength = total + b.length;
|
||||
// not enough space to fit all data, so splice and put back in the queue
|
||||
if (newLength > audio_buffer_size) {
|
||||
var tokeep = b.subarray(0, audio_buffer_size - total);
|
||||
out.set(tokeep, total);
|
||||
var tobuffer = b.subarray(audio_buffer_size - total, b.length);
|
||||
audio_buffers.unshift(tobuffer);
|
||||
break;
|
||||
} else {
|
||||
out.set(b, total);
|
||||
}
|
||||
total = newLength;
|
||||
}
|
||||
|
||||
e.outputBuffer.copyToChannel(out, 0);
|
||||
|
||||
if (!audio_buffers.length) {
|
||||
audio_buffer_progressbar_update();
|
||||
}
|
||||
}
|
||||
|
||||
var audio_buffer_total_average_level = 0;
|
||||
var audio_buffer_total_average_level_length = 0;
|
||||
|
||||
function audio_buffers_total_length() {
|
||||
return audio_buffers.map(function(b){ return b.length; }).reduce(function(a, b){ return a + b; }, 0);
|
||||
}
|
||||
|
||||
function audio_buffer_progressbar_update(reportedValue) {
|
||||
var audio_buffer_value = reportedValue;
|
||||
if (typeof(audio_buffer_value) === 'undefined') {
|
||||
audio_buffer_value = audio_buffers_total_length();
|
||||
}
|
||||
audio_buffer_value /= audio_context.sampleRate;
|
||||
audio_buffer_total_average_level_length++;
|
||||
audio_buffer_total_average_level = (audio_buffer_total_average_level * ((audio_buffer_total_average_level_length - 1) / audio_buffer_total_average_level_length)) + (audio_buffer_value / audio_buffer_total_average_level_length);
|
||||
var overrun = audio_buffer_value > audio_buffer_maximal_length_sec;
|
||||
var underrun = audio_buffer_value === 0;
|
||||
var text = "buffer";
|
||||
if (overrun) {
|
||||
text = "overrun";
|
||||
}
|
||||
if (underrun) {
|
||||
text = "underrun";
|
||||
}
|
||||
progressbar_set(e("openwebrx-bar-audio-buffer"), audio_buffer_value, "Audio " + text + " [" + (audio_buffer_value).toFixed(1) + " s]", overrun || underrun);
|
||||
}
|
||||
|
||||
|
||||
function audio_flush() {
|
||||
var flushed = false;
|
||||
var we_have_more_than = function (sec) {
|
||||
return sec * audio_context.sampleRate < audio_buffers_total_length();
|
||||
};
|
||||
if (we_have_more_than(audio_buffer_maximal_length_sec)) while (we_have_more_than(audio_buffer_decrease_to_on_overrun_sec)) {
|
||||
if (!flushed) audio_buffer_progressbar_update();
|
||||
flushed = true;
|
||||
audio_buffers.shift();
|
||||
}
|
||||
}
|
||||
|
||||
function webrx_set_param(what, value) {
|
||||
var params = {};
|
||||
@ -1632,11 +1496,10 @@ function webrx_set_param(what, value) {
|
||||
ws.send(JSON.stringify({"type": "dspcontrol", "params": params}));
|
||||
}
|
||||
|
||||
var starting_mute = false;
|
||||
var starting_offset_frequency;
|
||||
var starting_mod;
|
||||
|
||||
function parsehash() {
|
||||
function parseHash() {
|
||||
var h;
|
||||
if (h = window.location.hash) {
|
||||
h.substring(1).split(",").forEach(function (x) {
|
||||
@ -1657,106 +1520,21 @@ function parsehash() {
|
||||
}
|
||||
}
|
||||
|
||||
function audio_preinit() {
|
||||
try {
|
||||
var ctx = window.AudioContext || window.webkitAudioContext;
|
||||
audio_context = new ctx();
|
||||
}
|
||||
catch (e) {
|
||||
divlog('Your browser does not support Web Audio API, which is required for WebRX to run. Please upgrade to a HTML5 compatible browser.', 1);
|
||||
return;
|
||||
}
|
||||
function onAudioStart(success, apiType){
|
||||
divlog('Web Audio API succesfully initialized, using ' + apiType + ' API, sample rate: ' + audioEngine.getSampleRate() + " Hz");
|
||||
|
||||
//we send our setup packet
|
||||
// TODO this should be moved to another stage of initialization
|
||||
parsehash();
|
||||
// canvas_container is set after waterfall_init() has been called. we cannot initialize before.
|
||||
if (canvas_container) initialize_demodulator();
|
||||
|
||||
if (!audio_resampler) {
|
||||
audio_calculate_resampling(audio_context.sampleRate);
|
||||
audio_resampler = new sdrjs.RationalResamplerFF(audio_client_resampling_factor, 1);
|
||||
}
|
||||
//hide log panel in a second (if user has not hidden it yet)
|
||||
window.setTimeout(function () {
|
||||
if (typeof e("openwebrx-panel-log").openwebrxHidden === "undefined" && !was_error) {
|
||||
toggle_panel("openwebrx-panel-log");
|
||||
}
|
||||
}, 2000);
|
||||
|
||||
ws.send(JSON.stringify({
|
||||
"type": "dspcontrol",
|
||||
"action": "start",
|
||||
"params": {"output_rate": audio_server_output_rate}
|
||||
}));
|
||||
}
|
||||
|
||||
function audio_init() {
|
||||
if (is_chrome) audio_context.resume();
|
||||
if (starting_mute) toggleMute();
|
||||
|
||||
if (audio_client_resampling_factor === 0) return; //if failed to find a valid resampling factor...
|
||||
|
||||
audio_debug_time_start = (new Date()).getTime();
|
||||
audio_debug_time_last_start = audio_debug_time_start;
|
||||
audio_buffer_current_count_debug = 0;
|
||||
|
||||
if (audio_context.sampleRate < 44100 * 2)
|
||||
audio_buffer_size = 4096;
|
||||
else if (audio_context.sampleRate >= 44100 * 2 && audio_context.sampleRate < 44100 * 4)
|
||||
audio_buffer_size = 4096 * 2;
|
||||
else if (audio_context.sampleRate > 44100 * 4)
|
||||
audio_buffer_size = 4096 * 4;
|
||||
|
||||
//https://github.com/0xfe/experiments/blob/master/www/tone/js/sinewave.js
|
||||
audio_initialized = 1; // only tell on_ws_recv() not to call it again
|
||||
|
||||
// --- Resampling ---
|
||||
webrx_set_param("audio_rate", audio_context.sampleRate);
|
||||
|
||||
var finish = function() {
|
||||
divlog('Web Audio API succesfully initialized, using ' + audio_node.constructor.name + ', sample rate: ' + audio_context.sampleRate.toString() + " sps");
|
||||
initialize_demodulator();
|
||||
|
||||
//hide log panel in a second (if user has not hidden it yet)
|
||||
window.setTimeout(function () {
|
||||
if (typeof e("openwebrx-panel-log").openwebrxHidden === "undefined" && !was_error) {
|
||||
toggle_panel("openwebrx-panel-log");
|
||||
//animate(e("openwebrx-panel-log"),"opacity","",1,0,0.9,1000,60);
|
||||
//window.setTimeout(function(){toggle_panel("openwebrx-panel-log");e("openwebrx-panel-log").style.opacity="1";},1200)
|
||||
}
|
||||
}, 2000);
|
||||
};
|
||||
|
||||
gainNode = audio_context.createGain();
|
||||
gainNode.connect(audio_context.destination);
|
||||
//Synchronise volume with slider
|
||||
updateVolume();
|
||||
|
||||
if (audio_context.audioWorklet) {
|
||||
audio_context.audioWorklet.addModule('static/lib/AudioProcessor.js').then(function(){
|
||||
audio_node = new AudioWorkletNode(audio_context, 'openwebrx-audio-processor', {
|
||||
numberOfInputs: 0,
|
||||
numberOfOutputs: 1,
|
||||
outputChannelCount: [1],
|
||||
processorOptions: {
|
||||
maxLength: audio_buffer_maximal_length_sec
|
||||
}
|
||||
});
|
||||
audio_node.connect(gainNode);
|
||||
window.setInterval(function(){
|
||||
audio_node.port.postMessage(JSON.stringify({cmd:'getBuffers'}));
|
||||
}, audio_flush_interval_ms);
|
||||
audio_node.port.addEventListener('message', function(m){
|
||||
var json = JSON.parse(m.data);
|
||||
if (typeof(json.buffersize) !== 'undefined') {
|
||||
audio_buffer_progressbar_update(json.buffersize);
|
||||
}
|
||||
});
|
||||
audio_node.port.start();
|
||||
finish();
|
||||
});
|
||||
} else {
|
||||
//on Chrome v36, createJavaScriptNode has been replaced by createScriptProcessor
|
||||
var createjsnode_function = (audio_context.createJavaScriptNode === undefined) ? audio_context.createScriptProcessor.bind(audio_context) : audio_context.createJavaScriptNode.bind(audio_context);
|
||||
audio_node = createjsnode_function(audio_buffer_size, 0, 1);
|
||||
audio_node.onaudioprocess = audio_onprocess;
|
||||
audio_node.connect(gainNode);
|
||||
window.setInterval(audio_flush, audio_flush_interval_ms);
|
||||
finish();
|
||||
}
|
||||
}
|
||||
|
||||
function initialize_demodulator() {
|
||||
@ -1772,12 +1550,6 @@ function initialize_demodulator() {
|
||||
var reconnect_timeout = false;
|
||||
|
||||
function on_ws_closed() {
|
||||
try {
|
||||
audio_node.disconnect();
|
||||
}
|
||||
catch (dont_care) {
|
||||
}
|
||||
audio_initialized = 0;
|
||||
if (reconnect_timeout) {
|
||||
// max value: roundabout 8 and a half minutes
|
||||
reconnect_timeout = Math.min(reconnect_timeout * 2, 512000);
|
||||
@ -2168,25 +1940,66 @@ function init_header() {
|
||||
});
|
||||
}
|
||||
|
||||
function audio_buffer_progressbar_update(buffersize) {
|
||||
var audio_buffer_value = buffersize / audioEngine.getSampleRate();
|
||||
var overrun = audio_buffer_value > audio_buffer_maximal_length_sec;
|
||||
var underrun = audio_buffer_value === 0;
|
||||
var text = "buffer";
|
||||
if (overrun) {
|
||||
text = "overrun";
|
||||
}
|
||||
if (underrun) {
|
||||
text = "underrun";
|
||||
}
|
||||
progressbar_set(e("openwebrx-bar-audio-buffer"), audio_buffer_value, "Audio " + text + " [" + (audio_buffer_value).toFixed(1) + " s]", overrun || underrun);
|
||||
}
|
||||
|
||||
function updateNetworkStats() {
|
||||
var elapsed = (new Date() - debug_ws_time_start) / 1000;
|
||||
var network_speed_value = (debug_ws_data_received / 1000) / elapsed;
|
||||
progressbar_set(e("openwebrx-bar-network-speed"), network_speed_value * 8 / 2000, "Network usage [" + (network_speed_value * 8).toFixed(1) + " kbps]", false);
|
||||
}
|
||||
|
||||
function audioReporter(stats) {
|
||||
if (typeof(stats.buffersize) !== 'undefined') {
|
||||
audio_buffer_progressbar_update(stats.buffersize);
|
||||
}
|
||||
|
||||
if (typeof(stats.audioByteRate) !== 'undefined') {
|
||||
var audio_speed_value = stats.audioByteRate * 8;
|
||||
progressbar_set(e("openwebrx-bar-audio-speed"), audio_speed_value / 500000, "Audio stream [" + (audio_speed_value / 1000).toFixed(0) + " kbps]", false);
|
||||
}
|
||||
|
||||
if (typeof(stats.audioRate) !== 'undefined') {
|
||||
var audio_max_rate = audioEngine.getSampleRate() * 1.25;
|
||||
var audio_min_rate = audioEngine.getSampleRate() * .25;
|
||||
progressbar_set(e("openwebrx-bar-audio-output"), stats.audioRate / audio_max_rate, "Audio output [" + (stats.audioRate / 1000).toFixed(1) + " ksps]", stats.audioRate > audio_max_rate || stats.audioRate < audio_min_rate);
|
||||
}
|
||||
}
|
||||
|
||||
var bookmarks;
|
||||
var audioEngine;
|
||||
|
||||
function openwebrx_init() {
|
||||
if (ios || is_chrome) e("openwebrx-big-grey").style.display = "table-cell";
|
||||
var opb = e("openwebrx-play-button-text");
|
||||
opb.style.marginTop = (window.innerHeight / 2 - opb.clientHeight / 2).toString() + "px";
|
||||
audioEngine = new AudioEngine(audio_buffer_maximal_length_sec, audioReporter);
|
||||
if (!audioEngine.isAllowed()) {
|
||||
e("openwebrx-big-grey").style.display = "table-cell";
|
||||
var opb = e("openwebrx-play-button-text");
|
||||
opb.style.marginTop = (window.innerHeight / 2 - opb.clientHeight / 2).toString() + "px";
|
||||
} else {
|
||||
audioEngine.start(onAudioStart);
|
||||
}
|
||||
init_rx_photo();
|
||||
open_websocket();
|
||||
setInterval(updateNetworkStats, 1000);
|
||||
secondary_demod_init();
|
||||
digimodes_init();
|
||||
place_panels(first_show_panel);
|
||||
window.setTimeout(function () {
|
||||
window.setInterval(debug_audio, 1000);
|
||||
}, 1000);
|
||||
window.addEventListener("resize", openwebrx_resize);
|
||||
check_top_bar_congestion();
|
||||
init_header();
|
||||
bookmarks = new BookmarkBar();
|
||||
|
||||
parseHash();
|
||||
}
|
||||
|
||||
function digimodes_init() {
|
||||
@ -2210,14 +2023,13 @@ function update_dmr_timeslot_filtering() {
|
||||
webrx_set_param("dmr_filter", filter);
|
||||
}
|
||||
|
||||
function iosPlayButtonClick() {
|
||||
function playButtonClick() {
|
||||
//On iOS, we can only start audio from a click or touch event.
|
||||
audio_init();
|
||||
audioEngine.start(onAudioStart);
|
||||
e("openwebrx-big-grey").style.opacity = 0;
|
||||
window.setTimeout(function () {
|
||||
e("openwebrx-big-grey").style.display = "none";
|
||||
}, 1100);
|
||||
audio_allowed = 1;
|
||||
}
|
||||
|
||||
var rt = function (s, n) {
|
||||
@ -2226,37 +2038,6 @@ var rt = function (s, n) {
|
||||
});
|
||||
};
|
||||
|
||||
var audio_debug_time_start = 0;
|
||||
var audio_debug_time_last_start = 0;
|
||||
|
||||
function debug_audio() {
|
||||
if (audio_debug_time_start === 0) return; //audio_init has not been called
|
||||
var time_now = (new Date()).getTime();
|
||||
var audio_debug_time_since_last_call = (time_now - audio_debug_time_last_start) / 1000;
|
||||
audio_debug_time_last_start = time_now; //now
|
||||
var audio_debug_time_taken = (time_now - audio_debug_time_start) / 1000;
|
||||
var kbps_mult = (audio_compression === "adpcm") ? 8 : 16;
|
||||
|
||||
var audio_speed_value = audio_buffer_current_size_debug * kbps_mult / audio_debug_time_since_last_call;
|
||||
progressbar_set(e("openwebrx-bar-audio-speed"), audio_speed_value / 500000, "Audio stream [" + (audio_speed_value / 1000).toFixed(0) + " kbps]", false);
|
||||
|
||||
var audio_output_value = (audio_buffer_current_count_debug * audio_buffer_size) / audio_debug_time_taken;
|
||||
var audio_max_rate = audio_context.sampleRate * 1.25;
|
||||
var audio_min_rate = audio_context.sampleRate * .25;
|
||||
progressbar_set(e("openwebrx-bar-audio-output"), audio_output_value / audio_max_rate, "Audio output [" + (audio_output_value / 1000).toFixed(1) + " ksps]", audio_output_value > audio_max_rate || audio_output_value < audio_min_rate);
|
||||
|
||||
// disable when audioworklets used
|
||||
if (audio_node && !audio_node.port) audio_buffer_progressbar_update();
|
||||
|
||||
var debug_ws_time_taken = (time_now - debug_ws_time_start) / 1000;
|
||||
var network_speed_value = debug_ws_data_received / debug_ws_time_taken;
|
||||
progressbar_set(e("openwebrx-bar-network-speed"), network_speed_value * 8 / 2000, "Network usage [" + (network_speed_value * 8).toFixed(1) + " kbps]", false);
|
||||
|
||||
audio_buffer_current_size_debug = 0;
|
||||
|
||||
if (waterfall_measure_minmax) waterfall_measure_minmax_print();
|
||||
}
|
||||
|
||||
// ========================================================
|
||||
// ======================= PANELS =======================
|
||||
// ========================================================
|
||||
|
@ -157,6 +157,8 @@ class OpenWebRxReceiverClient(Client):
|
||||
|
||||
self.sdr = next
|
||||
|
||||
self.startDsp()
|
||||
|
||||
# send initial config
|
||||
configProps = (
|
||||
self.sdr.getProps()
|
||||
|
Loading…
Reference in New Issue
Block a user