Start OpenWebRX diff --git a/htdocs/lib/AudioEngine.js b/htdocs/lib/AudioEngine.js new file mode 100644 index 0000000..aa79e72 --- /dev/null +++ b/htdocs/lib/AudioEngine.js @@ -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.
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); +} diff --git a/htdocs/lib/AudioProcessor.js b/htdocs/lib/AudioProcessor.js index 525299f..0902c50 100644 --- a/htdocs/lib/AudioProcessor.js +++ b/htdocs/lib/AudioProcessor.js @@ -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; diff --git a/htdocs/openwebrx.js b/htdocs/openwebrx.js index 855018e..58ff2d5 100644 --- a/htdocs/openwebrx.js +++ b/htdocs/openwebrx.js @@ -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.
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(); + // 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 ======================= // ======================================================== diff --git a/owrx/connection.py b/owrx/connection.py index 4ee8734..b4b827b 100644 --- a/owrx/connection.py +++ b/owrx/connection.py @@ -157,6 +157,8 @@ class OpenWebRxReceiverClient(Client): self.sdr = next + self.startDsp() + # send initial config configProps = ( self.sdr.getProps()