kill client-side early rebuffering, improving the latency
This commit is contained in:
Jakob Ketterl
parent
93d4e629d1
commit
0b2c457030
@ -181,11 +181,6 @@ sdrs = {
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# ==== Misc settings ====
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client_audio_buffer_size = 5
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# increasing client_audio_buffer_size will:
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# - also increase the latency
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# - decrease the chance of audio underruns
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iq_port_range = [
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4950,
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4960,
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@ -1115,7 +1115,6 @@ function on_ws_recv(evt) {
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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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window.audio_buffering_fill_to = config['client_audio_buffer_size'];
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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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@ -1258,7 +1257,7 @@ function on_ws_recv(evt) {
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audio_prepare(audio_data);
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audio_buffer_current_size_debug += audio_data.length;
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audio_buffer_all_size_debug += audio_data.length;
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if (!(ios || is_chrome) && (audio_initialized === 0 && audio_prepared_buffers.length > audio_buffering_fill_to)) audio_init();
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if (!(ios || is_chrome) && (audio_initialized === 0)) audio_init();
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break;
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case 3:
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// secondary FFT
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@ -1536,12 +1535,8 @@ var audio_buffer_maximal_length_sec = 3; //actual number of samples are calculat
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var audio_buffer_decrease_to_on_overrun_sec = 2.2;
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var audio_flush_interval_ms = 500; //the interval in which audio_flush() is called
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var audio_prepared_buffers = Array();
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var audio_rebuffer;
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var audio_buffers = [];
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var audio_last_output_buffer;
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var audio_buffering = false;
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//var audio_buffering_fill_to=4; //on audio underrun we wait until this n*audio_buffer_size samples are present
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//tnx to the hint from HA3FLT, now we have about half the response time! (original value: 10)
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function gain_ff(gain_value, data) //great! solved clicking! will have to move to sdr.js
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{
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@ -1551,24 +1546,12 @@ function gain_ff(gain_value, data) //great! solved clicking! will have to move t
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}
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function audio_prepare(data) {
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//audio_rebuffer.push(sdrjs.ConvertI16_F(data));//no resampling
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//audio_rebuffer.push(audio_resampler.process(sdrjs.ConvertI16_F(data)));//resampling without ADPCM
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if (audio_compression === "none")
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audio_rebuffer.push(audio_resampler.process(gain_ff(volume, sdrjs.ConvertI16_F(data))));//resampling without ADPCM
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else if (audio_compression === "adpcm")
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audio_rebuffer.push(audio_resampler.process(gain_ff(volume, sdrjs.ConvertI16_F(audio_codec.decode(data))))); //resampling & ADPCM
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else return;
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//console.log("prepare",data.length,audio_rebuffer.remaining());
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while (audio_rebuffer.remaining()) {
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audio_prepared_buffers.push(audio_rebuffer.take());
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audio_buffer_current_count_debug++;
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}
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if (audio_buffering && audio_prepared_buffers.length > audio_buffering_fill_to) {
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console.log("buffers now: " + audio_prepared_buffers.length.toString());
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audio_buffering = false;
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var buffer = data;
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if (audio_compression === "adpcm") {
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//resampling & ADPCM
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buffer = audio_codec.decode(buffer);
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}
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audio_buffers.push(audio_resampler.process(gain_ff(volume, sdrjs.ConvertI16_F(buffer))));
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}
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if (!AudioBuffer.prototype.copyToChannel) { //Chrome 36 does not have it, Firefox does
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@ -1579,45 +1562,50 @@ if (!AudioBuffer.prototype.copyToChannel) { //Chrome 36 does not have it, Firefo
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}
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}
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var silence = new Float32Array(4096);
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function audio_onprocess(e) {
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if (audio_buffering) {
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e.outputBuffer.copyToChannel(silence, 0);
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return;
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}
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if (audio_prepared_buffers.length === 0) {
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audio_buffer_progressbar_update();
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audio_buffering = true;
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e.outputBuffer.copyToChannel(silence, 0);
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var total = 0;
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var out = new Float32Array(audio_buffer_size);
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while (audio_buffers.length) {
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var b = audio_buffers.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 > audio_buffer_size) {
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var tokeep = b.slice(0, audio_buffer_size - total);
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out.set(tokeep, total);
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var tobuffer = b.slice(audio_buffer_size - total, b.length);
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audio_buffers.unshift(tobuffer);
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break;
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} else {
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var buf = audio_prepared_buffers.shift();
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e.outputBuffer.copyToChannel(buf, 0);
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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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var audio_buffer_progressbar_update_disabled = false;
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var audio_buffer_total_average_level = 0;
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var audio_buffer_total_average_level_length = 0;
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var audio_overrun_cnt = 0;
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var audio_underrun_cnt = 0;
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function audio_buffers_total_length() {
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return audio_buffers.map(function(b){ return b.length; }).reduce(function(a, b){ return a + b; }, 0);
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}
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function audio_buffer_progressbar_update() {
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if (audio_buffer_progressbar_update_disabled) return;
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var audio_buffer_value = (audio_prepared_buffers.length * audio_buffer_size) / audio_context.sampleRate;
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var audio_buffer_value = audio_buffers_total_length() / audio_context.sampleRate;
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audio_buffer_total_average_level_length++;
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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);
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var overrun = audio_buffer_value > audio_buffer_maximal_length_sec;
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var underrun = audio_prepared_buffers.length === 0;
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var underrun = audio_buffers.length === 0;
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var text = "buffer";
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if (overrun) {
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text = "overrun";
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console.log("audio overrun, " + (++audio_overrun_cnt).toString());
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}
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if (underrun) {
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text = "underrun";
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console.log("audio underrun, " + (++audio_underrun_cnt).toString());
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}
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if (overrun || underrun) {
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audio_buffer_progressbar_update_disabled = true;
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@ -1633,12 +1621,12 @@ function audio_buffer_progressbar_update() {
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function audio_flush() {
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var flushed = false;
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var we_have_more_than = function (sec) {
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return sec * audio_context.sampleRate < audio_prepared_buffers.length * audio_buffer_size;
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return sec * audio_context.sampleRate < audio_buffers_total_length();
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};
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if (we_have_more_than(audio_buffer_maximal_length_sec)) while (we_have_more_than(audio_buffer_decrease_to_on_overrun_sec)) {
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if (!flushed) audio_buffer_progressbar_update();
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flushed = true;
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audio_prepared_buffers.shift();
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audio_buffers.shift();
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}
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}
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@ -1690,13 +1678,11 @@ function audio_preinit() {
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else if (audio_context.sampleRate > 44100 * 4)
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audio_buffer_size = 4096 * 4;
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if (!audio_rebuffer) {
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audio_rebuffer = new sdrjs.Rebuffer(audio_buffer_size, sdrjs.REBUFFER_FIXED);
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audio_last_output_buffer = new Float32Array(audio_buffer_size);
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//we send our setup packet
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// TODO this should be moved to another stage of initialization
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parsehash();
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if (!audio_resampler) {
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audio_calculate_resampling(audio_context.sampleRate);
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audio_resampler = new sdrjs.RationalResamplerFF(audio_client_resampling_factor, 1);
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}
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@ -65,7 +65,6 @@ class OpenWebRxReceiverClient(Client):
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"fft_compression",
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"max_clients",
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"start_mod",
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"client_audio_buffer_size",
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"start_freq",
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"center_freq",
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"mathbox_waterfall_colors",
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