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refactor all the audio stuff into classes and a separate file

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This commit is contained in:
Jakob Ketterl 2019-10-20 18:53:23 +02:00
parent 91b8c55de9
commit 13d7686258
5 changed files with 305 additions and 308 deletions
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3
htdocs/index.html
View File

@ -28,6 +28,7 @@
<script src="static/lib/jquery-3.2.1.min.js"></script>
<script src="static/lib/jquery.nanoscroller.js"></script>
<script src="static/lib/bookmarks.js"></script>
<script src="static/lib/AudioEngine.js"></script>
<link rel="stylesheet" type="text/css" href="static/lib/nanoscroller.css" />
<link rel="stylesheet" type="text/css" href="static/css/openwebrx.css" />
<meta charset="utf-8">
@ -209,7 +210,7 @@
</div>
</div>
</div>
<div id="openwebrx-big-grey" onclick="iosPlayButtonClick();">
<div id="openwebrx-big-grey" onclick="playButtonClick();">
<div id="openwebrx-play-button-text">
<img id="openwebrx-play-button" src="static/gfx/openwebrx-play-button.png" />
<br /><br />Start OpenWebRX

214
htdocs/lib/AudioEngine.js Normal file
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@ -0,0 +1,214 @@
// this controls if the new AudioWorklet API should be used if available.
// the engine will still fall back to the ScriptProcessorNode if this is set to true but not available in the browser.
var useAudioWorklets = true;
function AudioEngine(maxBufferLength, audioReporter) {
this.audioReporter = audioReporter;
this.resetStats();
var ctx = window.AudioContext || window.webkitAudioContext;
if (!ctx) {
return;
}
this.audioContext = new ctx();
this.allowed = this.audioContext.state === 'running';
this.started = false;
this.audioCodec = new sdrjs.ImaAdpcm();
this.compression = 'none';
this.setupResampling();
this.resampler = new sdrjs.RationalResamplerFF(this.resamplingFactor, 1);
this.maxBufferSize = maxBufferLength * this.getSampleRate();
}
AudioEngine.prototype.start = function(callback) {
var me = this;
if (me.resamplingFactor === 0) return; //if failed to find a valid resampling factor...
if (me.started) {
if (callback) callback(false);
return;
}
me.audioContext.resume().then(function(){
me.allowed = me.audioContext.state === 'running';
if (!me.allowed) {
if (callback) callback(false);
return;
}
me.started = true;
me.gainNode = me.audioContext.createGain();
me.gainNode.connect(me.audioContext.destination);
if (useAudioWorklets && me.audioContext.audioWorklet) {
me.audioContext.audioWorklet.addModule('static/lib/AudioProcessor.js').then(function(){
me.audioNode = new AudioWorkletNode(me.audioContext, 'openwebrx-audio-processor', {
numberOfInputs: 0,
numberOfOutputs: 1,
outputChannelCount: [1],
processorOptions: {
maxBufferSize: me.maxBufferSize
}
});
me.audioNode.connect(me.gainNode);
me.audioNode.port.addEventListener('message', function(m){
var json = JSON.parse(m.data);
if (typeof(json.buffersize) !== 'undefined') {
me.audioReporter(json);
}
});
me.audioNode.port.start();
if (callback) callback(true, 'AudioWorklet');
});
} else {
me.audioBuffers = [];
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];
}
}
var bufferSize;
if (me.audioContext.sampleRate < 44100 * 2)
bufferSize = 4096;
else if (me.audioContext.sampleRate >= 44100 * 2 && me.audioContext.sampleRate < 44100 * 4)
bufferSize = 4096 * 2;
else if (me.audioContext.sampleRate > 44100 * 4)
bufferSize = 4096 * 4;
function audio_onprocess(e) {
var total = 0;
var out = new Float32Array(bufferSize);
while (me.audioBuffers.length) {
var b = me.audioBuffers.shift();
var newLength = total + b.length;
// not enough space to fit all data, so splice and put back in the queue
if (newLength > bufferSize) {
var tokeep = b.subarray(0, bufferSize - total);
out.set(tokeep, total);
var tobuffer = b.subarray(bufferSize - total, b.length);
me.audioBuffers.unshift(tobuffer);
break;
} else {
out.set(b, total);
}
total = newLength;
}
e.outputBuffer.copyToChannel(out, 0);
}
//on Chrome v36, createJavaScriptNode has been replaced by createScriptProcessor
var method = 'createScriptProcessor';
if (me.audioContext.createJavaScriptNode) {
method = 'createJavaScriptNode';
}
me.audioNode = me.audioContext[method](bufferSize, 0, 1);
me.audioNode.onaudioprocess = audio_onprocess;
me.audioNode.connect(me.gainNode);
if (callback) callback(true, 'ScriptProcessorNode');
}
setInterval(me.reportStats.bind(me), 1000);
});
}
AudioEngine.prototype.isAllowed = function() {
return this.allowed;
}
AudioEngine.prototype.reportStats = function() {
var stats = {}
if (this.audioNode.port) {
this.audioNode.port.postMessage(JSON.stringify({cmd:'getBuffers'}));
} else {
stats.buffersize = this.getBuffersize();
}
stats.audioRate = this.stats.audioSamples;
var elapsed = new Date() - this.stats.startTime;
stats.audioByteRate = this.stats.audioBytes * 1000 / elapsed
this.audioReporter(stats);
// sample rate is just measuring the last seconds
this.stats.audioSamples = 0;
}
AudioEngine.prototype.resetStats = function() {
this.stats = {
startTime: new Date(),
audioBytes: 0,
audioSamples: 0
};
}
AudioEngine.prototype.setupResampling = function() { //both at the server and the client
var output_range_max = 12000;
var output_range_min = 8000;
var targetRate = this.audioContext.sampleRate;
var i = 1;
while (true) {
var audio_server_output_rate = Math.floor(targetRate / i);
if (audio_server_output_rate < output_range_min) {
this.resamplingFactor = 0;
this.outputRate = 0;
divlog('Your audio card sampling rate (' + targetRate + ') is not supported.<br />Please change your operating system default settings in order to fix this.', 1);
break;
} else if (audio_server_output_rate >= output_range_min && audio_server_output_rate <= output_range_max) {
this.resamplingFactor = i;
this.outputRate = audio_server_output_rate;
break; //okay, we're done
}
i++;
}
}
AudioEngine.prototype.getOutputRate = function() {
return this.outputRate;
}
AudioEngine.prototype.getSampleRate = function() {
return this.audioContext.sampleRate;
}
AudioEngine.prototype.pushAudio = function(data) {
if (!this.audioNode) return;
this.stats.audioBytes += data.byteLength;
var buffer;
if (this.compression === "adpcm") {
//resampling & ADPCM
buffer = this.audioCodec.decode(new Uint8Array(data));
} else {
buffer = new Int16Array(data);
}
buffer = this.resampler.process(sdrjs.ConvertI16_F(buffer));
this.stats.audioSamples += buffer.length;
if (this.audioNode.port) {
// AudioWorklets supported
this.audioNode.port.postMessage(buffer);
} else {
// silently drop excess samples
if (this.getBuffersize() + buffer.length <= this.maxBufferSize) {
this.audioBuffers.push(buffer);
}
}
}
AudioEngine.prototype.setCompression = function(compression) {
this.compression = compression;
}
AudioEngine.prototype.setVolume = function(volume) {
this.gainNode.gain.value = volume;
}
AudioEngine.prototype.getBuffersize = function() {
// only available when using ScriptProcessorNode
if (!this.audioBuffers) return 0;
return this.audioBuffers.map(function(b){ return b.length; }).reduce(function(a, b){ return a + b; }, 0);
}

3
htdocs/lib/AudioProcessor.js
View File

@ -1,9 +1,8 @@
class OwrxAudioProcessor extends AudioWorkletProcessor {
constructor(options){
super(options);
this.maxLength = options.processorOptions.maxLength;
// initialize ringbuffer, make sure it aligns with the expected buffer size of 128
this.bufferSize = Math.round(sampleRate * this.maxLength / 128) * 128
this.bufferSize = Math.round(options.processorOptions.maxBufferSize / 128) * 128
this.audioBuffer = new Float32Array(this.bufferSize);
this.inPos = 0;
this.outPos = 0;

375
htdocs/openwebrx.js
View File

@ -30,16 +30,12 @@ function arrayBufferToString(buf) {
var bandwidth;
var center_freq;
var audio_buffer_current_size_debug = 0;
var audio_buffer_current_count_debug = 0;
var fft_size;
var fft_fps;
var fft_compression = "none";
var fft_codec = new sdrjs.ImaAdpcm();
var audio_compression = "none";
var waterfall_setup_done = 0;
var secondary_fft_size;
var audio_allowed;
var rx_photo_state = 1;
function e(what) {
@ -108,7 +104,7 @@ function style_value(of_what, which) {
}
function updateVolume() {
gainNode.gain.value = parseFloat(e("openwebrx-panel-volume").value) / 100;
audioEngine.setVolume(parseFloat(e("openwebrx-panel-volume").value) / 100);
}
function toggleMute() {
@ -406,8 +402,8 @@ function Demodulator_default_analog(offset_frequency, subtype) {
this.subtype = subtype;
this.filter = {
min_passband: 100,
high_cut_limit: (audio_server_output_rate / 2) - 1, //audio_context.sampleRate/2,
low_cut_limit: (-audio_server_output_rate / 2) + 1 //-audio_context.sampleRate/2
high_cut_limit: (audioEngine.getOutputRate() / 2) - 1,
low_cut_limit: (-audioEngine.getOutputRate() / 2) + 1
};
//Subtypes only define some filter parameters and the mod string sent to server,
//so you may set these parameters in your custom child class.
@ -689,7 +685,8 @@ function scale_px_from_freq(f, range) {
function get_visible_freq_range() {
var out = {};
var fcalc = function (x) {
return Math.round(((-zoom_offset_px + x) / canvases[0].clientWidth) * bandwidth) + (center_freq - bandwidth / 2);
var canvasWidth = canvas_container.clientWidth * zoom_levels[zoom_level];
return Math.round(((-zoom_offset_px + x) / canvasWidth) * bandwidth) + (center_freq - bandwidth / 2);
};
out.start = fcalc(0);
out.center = fcalc(canvas_container.clientWidth / 2);
@ -1063,30 +1060,8 @@ function resize_waterfall_container(check_init) {
}
var audio_server_output_rate = 11025;
var audio_client_resampling_factor = 4;
function audio_calculate_resampling(targetRate) { //both at the server and the client
var output_range_max = 12000;
var output_range_min = 8000;
var i = 1;
while (true) {
audio_server_output_rate = Math.floor(targetRate / i);
if (audio_server_output_rate < output_range_min) {
audio_client_resampling_factor = audio_server_output_rate = 0;
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);
}
if (audio_server_output_rate >= output_range_min && audio_server_output_rate <= output_range_max) break; //okay, we're done
i++;
}
audio_client_resampling_factor = i;
console.log("audio_calculate_resampling() :: " + audio_client_resampling_factor.toString() + ", " + audio_server_output_rate.toString());
}
var debug_ws_data_received = 0;
var debug_ws_time_start = 0;
var debug_ws_time_start;
var max_clients_num = 0;
var client_num = 0;
var currentprofile;
@ -1096,7 +1071,7 @@ var COMPRESS_FFT_PAD_N = 10; //should be the same as in csdr.c
function on_ws_recv(evt) {
if (typeof evt.data === 'string') {
// text messages
debug_ws_data_received += evt.data.length / 1000;
debug_ws_data_received += evt.data.length;
if (evt.data.substr(0, 16) === "CLIENT DE SERVER") {
divlog("Server acknowledged WebSocket connection.");
@ -1106,19 +1081,20 @@ function on_ws_recv(evt) {
switch (json.type) {
case "config":
var config = json['value'];
window.waterfall_colors = config['waterfall_colors'];
window.waterfall_min_level_default = config['waterfall_min_level'];
window.waterfall_max_level_default = config['waterfall_max_level'];
window.waterfall_auto_level_margin = config['waterfall_auto_level_margin'];
waterfall_colors = config['waterfall_colors'];
waterfall_min_level_default = config['waterfall_min_level'];
waterfall_max_level_default = config['waterfall_max_level'];
waterfall_auto_level_margin = config['waterfall_auto_level_margin'];
waterfallColorsDefault();
window.starting_mod = config['start_mod'];
window.starting_offset_frequency = config['start_offset_freq'];
starting_mod = config['start_mod'];
starting_offset_frequency = config['start_offset_freq'];
bandwidth = config['samp_rate'];
center_freq = config['center_freq'] + config['lfo_offset'];
fft_size = config['fft_size'];
fft_fps = config['fft_fps'];
audio_compression = config['audio_compression'];
var audio_compression = config['audio_compression'];
audioEngine.setCompression(audio_compression);
divlog("Audio stream is " + ((audio_compression === "adpcm") ? "compressed" : "uncompressed") + ".");
fft_compression = config['fft_compression'];
divlog("FFT stream is " + ((fft_compression === "adpcm") ? "compressed" : "uncompressed") + ".");
@ -1129,20 +1105,14 @@ function on_ws_recv(evt) {
mathbox_waterfall_history_length = config['mathbox_waterfall_history_length'];
waterfall_init();
audio_preinit();
initialize_demodulator();
bookmarks.loadLocalBookmarks();
if (audio_allowed) {
if (audio_initialized) {
initialize_demodulator();
} else {
audio_init();
}
}
waterfall_clear();
currentprofile = config['profile_id'];
$('#openwebrx-sdr-profiles-listbox').val(currentprofile);
break;
case "secondary_config":
var s = json['value'];
@ -1222,7 +1192,7 @@ function on_ws_recv(evt) {
}
} else if (evt.data instanceof ArrayBuffer) {
// binary messages
debug_ws_data_received += evt.data.byteLength / 1000;
debug_ws_data_received += evt.data.byteLength;
var type = new Uint8Array(evt.data, 0, 1)[0];
var data = evt.data.slice(1);
@ -1247,15 +1217,7 @@ function on_ws_recv(evt) {
break;
case 2:
// audio data
var audio_data;
if (audio_compression === "adpcm") {
audio_data = new Uint8Array(data);
} else {
audio_data = new Int16Array(data);
}
audio_prepare(audio_data);
audio_buffer_current_size_debug += audio_data.length;
if (!(ios || is_chrome) && (audio_initialized === 0)) audio_init();
audioEngine.pushAudio(data);
break;
case 3:
// secondary FFT
@ -1498,8 +1460,13 @@ function on_ws_opened() {
ws.send("SERVER DE CLIENT client=openwebrx.js type=receiver");
divlog("WebSocket opened to " + ws.url);
debug_ws_data_received = 0;
debug_ws_time_start = new Date().getTime();
debug_ws_time_start = new Date();
reconnect_timeout = false;
ws.send(JSON.stringify({
"type": "dspcontrol",
"action": "start",
"params": {"output_rate": audioEngine.getOutputRate()}
}));
}
var was_error = 0;
@ -1517,114 +1484,11 @@ function divlog(what, is_error) {
nano.nanoScroller({scroll: 'bottom'});
}
var audio_context;
var audio_initialized = 0;
var gainNode;
var volumeBeforeMute = 100.0;
var mute = false;
var audio_resampler;
var audio_codec = new sdrjs.ImaAdpcm();
var audio_node;
// Optimalise these if audio lags or is choppy:
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();
if (!audio_resampler) {
audio_calculate_resampling(audio_context.sampleRate);
audio_resampler = new sdrjs.RationalResamplerFF(audio_client_resampling_factor, 1);
}
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();
// canvas_container is set after waterfall_init() has been called. we cannot initialize before.
if (canvas_container) 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";
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 =======================
// ========================================================

2
owrx/connection.py
View File

@ -157,6 +157,8 @@ class OpenWebRxReceiverClient(Client):
self.sdr = next
self.startDsp()
# send initial config
configProps = (
self.sdr.getProps()