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sdr-wideband-suite
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Update web audio player and streamer diagnostics

master
Jan Svabenik пре 5 часа
родитељ
1b0cefb367
комит
f27080b8b5
4 измењених фајлова са 264 додато и 190 уклоњено
Подељен поглед
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  1. BIN
      internal/demod/gpudemod/build/gpudemod_kernels.exp
  2. +27
    -13
      internal/recorder/streamer.go
  3. +109
    -177
      web/app.js
  4. +128
    -0
      web/ring-player-processor.js

BIN
internal/demod/gpudemod/build/gpudemod_kernels.exp Прегледај датотеку


+ 27
- 13
internal/recorder/streamer.go Прегледај датотеку

@@ -763,20 +763,34 @@ func (sess *streamSession) processSnippet(snippet []complex64, snipRate int) ([]

var dec []complex64
if decim1 > 1 {
cutoff := float64(actualDemodRate) / 2.0 * 0.8

// Lazy-init or reinit stateful FIR if parameters changed
if sess.preDemodFIR == nil || sess.preDemodRate != snipRate || sess.preDemodCutoff != cutoff {
taps := dsp.LowpassFIR(cutoff, snipRate, 101)
sess.preDemodFIR = dsp.NewStatefulFIRComplex(taps)
sess.preDemodRate = snipRate
sess.preDemodCutoff = cutoff
sess.preDemodDecim = decim1
sess.preDemodDecimPhase = 0 // reset decimation phase on FIR reinit
}
// For WFM: skip the pre-demod anti-alias FIR entirely.
// FM broadcast has ±75kHz deviation. The old cutoff at
// actualDemodRate/2*0.8 = 68kHz was BELOW the FM deviation,
// clipping the modulation and causing amplitude dips that
// the FM discriminator turned into audible clicks (4-5/sec).
// The extraction stage already bandlimited to ±125kHz (BW/2 * bwMult),
// which is sufficient anti-alias protection for the 512k→170k decimation.
//
// For NFM/other: use a cutoff that preserves the full signal bandwidth.
if isWFM {
// WFM: decimate without FIR — extraction filter is sufficient
dec = dsp.DecimateStateful(fullSnip, decim1, &sess.preDemodDecimPhase)
} else {
cutoff := float64(actualDemodRate) / 2.0 * 0.8

// Lazy-init or reinit stateful FIR if parameters changed
if sess.preDemodFIR == nil || sess.preDemodRate != snipRate || sess.preDemodCutoff != cutoff {
taps := dsp.LowpassFIR(cutoff, snipRate, 101)
sess.preDemodFIR = dsp.NewStatefulFIRComplex(taps)
sess.preDemodRate = snipRate
sess.preDemodCutoff = cutoff
sess.preDemodDecim = decim1
sess.preDemodDecimPhase = 0
}

filtered := sess.preDemodFIR.ProcessInto(fullSnip, sess.growIQ(len(fullSnip)))
dec = dsp.DecimateStateful(filtered, decim1, &sess.preDemodDecimPhase)
filtered := sess.preDemodFIR.ProcessInto(fullSnip, sess.growIQ(len(fullSnip)))
dec = dsp.DecimateStateful(filtered, decim1, &sess.preDemodDecimPhase)
}
} else {
dec = fullSnip
}


+ 109
- 177
web/app.js Прегледај датотеку

@@ -143,17 +143,24 @@ let decisionIndex = new Map();
// ---------------------------------------------------------------------------
// LiveListenWS — WebSocket-based gapless audio streaming via /ws/audio
// ---------------------------------------------------------------------------
// v2: Ring-buffer based playback with smooth underrun handling.
// v4: Jank-resistant scheduled playback.
//
// Architecture:
// WebSocket → PCM chunks → ring buffer → AudioWorklet/ScriptProcessor → speakers
// Problem: Main-thread Canvas rendering blocks for 150-250ms, starving
// ScriptProcessorNode callbacks and causing audio underruns.
// AudioWorklet would fix this but requires Secure Context (HTTPS/localhost).
//
// Key improvements over v1:
// - 250ms ring buffer absorbs feed_gap jitter (150-220ms observed)
// - On underrun: fade to silence instead of hard resync (no click)
// - On overrun: skip oldest data instead of hard drop
// - Continuous pull-based playback (worklet pulls from ring) instead of
// push-based scheduling (BufferSource per chunk)
// Solution: Use BufferSource scheduling (like v1) but with a much larger
// jitter budget. We pre-schedule audio 400ms into the future so that even
// a 300ms main-thread hang doesn't cause a gap. The AudioContext's internal
// scheduling is sample-accurate and runs on a system thread — once a
// BufferSource is scheduled, it plays regardless of main-thread state.
//
// Key differences from v1:
// - 400ms target latency (was 100ms) — survives observed 250ms hangs
// - Soft resync: on underrun, schedule next chunk with a short crossfade
// gap instead of hard jump
// - Overrun cap at 800ms (was 500ms)
// - Chunk coalescing: merge small chunks to reduce scheduling overhead
// ---------------------------------------------------------------------------
class LiveListenWS {
constructor(freq, bw, mode) {
@@ -165,19 +172,15 @@ class LiveListenWS {
this.sampleRate = 48000;
this.channels = 1;
this.playing = false;
this.nextTime = 0;
this.started = false;
this._onStop = null;

// Ring buffer (interleaved float32 samples)
this._ringBuf = null;
this._ringSize = 0;
this._ringWrite = 0; // write cursor (producer: WebSocket)
this._ringRead = 0; // read cursor (consumer: audio callback)
this._scriptNode = null;
this._fadeGain = 1.0; // smooth fade on underrun
this._started = false;
this._totalWritten = 0;
this._totalRead = 0;
this._underruns = 0;
// Chunk coalescing buffer
this._pendingSamples = [];
this._pendingLen = 0;
this._flushTimer = 0;
// Fade state for soft resync
this._lastEndSample = null; // last sample value per channel for crossfade
}

start() {
@@ -207,9 +210,8 @@ class LiveListenWS {
} catch (e) { /* ignore */ }
return;
}
// Binary PCM data (s16le)
if (!this.audioCtx || !this.playing) return;
this._pushPCM(ev.data);
this._onPCM(ev.data);
};

this.ws.onclose = () => {
@@ -228,31 +230,20 @@ class LiveListenWS {

stop() {
this.playing = false;
if (this.ws) {
this.ws.close();
this.ws = null;
}
if (this.ws) { this.ws.close(); this.ws = null; }
this._teardownAudio();
}

onStop(fn) { this._onStop = fn; }

// --- Internal: Audio setup ---

_teardownAudio() {
if (this._scriptNode) {
this._scriptNode.disconnect();
this._scriptNode = null;
}
if (this.audioCtx) {
this.audioCtx.close().catch(() => {});
this.audioCtx = null;
}
this._ringBuf = null;
this._ringWrite = 0;
this._ringRead = 0;
this._fadeGain = 1.0;
this._started = false;
if (this._flushTimer) { clearTimeout(this._flushTimer); this._flushTimer = 0; }
if (this.audioCtx) { this.audioCtx.close().catch(() => {}); this.audioCtx = null; }
this.nextTime = 0;
this.started = false;
this._pendingSamples = [];
this._pendingLen = 0;
this._lastEndSample = null;
}

_initAudio() {
@@ -261,161 +252,102 @@ class LiveListenWS {
sampleRate: this.sampleRate
});
this.audioCtx.resume().catch(() => {});

// Ring buffer: 500ms capacity (handles jitter up to ~400ms)
const ringDurationSec = 0.5;
this._ringSize = Math.ceil(this.sampleRate * this.channels * ringDurationSec);
this._ringBuf = new Float32Array(this._ringSize);
this._ringWrite = 0;
this._ringRead = 0;
this._fadeGain = 1.0;
this._started = false;
this._totalWritten = 0;
this._totalRead = 0;

// Use ScriptProcessorNode (deprecated but universally supported).
// Buffer size 2048 at 48kHz = 42.7ms callback interval — responsive enough.
const bufSize = 2048;
this._scriptNode = this.audioCtx.createScriptProcessor(bufSize, 0, this.channels);
this._scriptNode.onaudioprocess = (e) => this._audioCallback(e);
this._scriptNode.connect(this.audioCtx.destination);
}

// --- Internal: Producer (WebSocket → ring buffer) ---

_pushPCM(buf) {
if (!this._ringBuf) return;

const samples = new Int16Array(buf);
const nSamples = samples.length; // interleaved: nFrames * channels
if (nSamples === 0) return;

const ring = this._ringBuf;
const size = this._ringSize;
let w = this._ringWrite;

// Check available space
const used = (w - this._ringRead + size) % size;
const free = size - used - 1; // -1 to distinguish full from empty

if (nSamples > free) {
// Overrun: advance read cursor to make room (discard oldest audio).
// This keeps latency bounded instead of growing unbounded.
const deficit = nSamples - free;
this._ringRead = (this._ringRead + deficit) % size;
}

// Write samples into ring
for (let i = 0; i < nSamples; i++) {
ring[w] = samples[i] / 32768;
w = (w + 1) % size;
}
this._ringWrite = w;
this._totalWritten += nSamples;

// Start playback after buffering ~200ms (enough to absorb one feed_gap)
if (!this._started) {
const buffered = (this._ringWrite - this._ringRead + size) % size;
const target = Math.ceil(this.sampleRate * this.channels * 0.2);
if (buffered >= target) {
this._started = true;
}
this.nextTime = 0;
this.started = false;
this._lastEndSample = null;
}

_onPCM(buf) {
// Coalesce small chunks: accumulate until we have >= 40ms or 50ms passes.
// This reduces BufferSource scheduling overhead from ~12/sec to ~6/sec
// and produces larger, more stable buffers.
this._pendingSamples.push(new Int16Array(buf));
this._pendingLen += buf.byteLength / 2; // Int16 = 2 bytes

const minFrames = Math.ceil(this.sampleRate * 0.04); // 40ms worth
const haveFrames = Math.floor(this._pendingLen / this.channels);

if (haveFrames >= minFrames) {
this._flushPending();
} else if (!this._flushTimer) {
// Flush after 50ms even if we don't have enough (prevents stale data)
this._flushTimer = setTimeout(() => {
this._flushTimer = 0;
if (this._pendingLen > 0) this._flushPending();
}, 50);
}
}

// --- Internal: Consumer (ring buffer → speakers) ---

_audioCallback(e) {
const ring = this._ringBuf;
const size = this._ringSize;
const ch = this.channels;
const outLen = e.outputBuffer.length; // frames per channel
_flushPending() {
if (this._flushTimer) { clearTimeout(this._flushTimer); this._flushTimer = 0; }
if (this._pendingSamples.length === 0) return;

if (!ring || !this._started) {
// Not ready yet — output silence
for (let c = 0; c < e.outputBuffer.numberOfChannels; c++) {
e.outputBuffer.getChannelData(c).fill(0);
}
return;
// Merge all pending into one Int16Array
const total = this._pendingLen;
const merged = new Int16Array(total);
let off = 0;
for (const chunk of this._pendingSamples) {
merged.set(chunk, off);
off += chunk.length;
}
this._pendingSamples = [];
this._pendingLen = 0;

// How many interleaved samples do we need?
const need = outLen * ch;
const available = (this._ringWrite - this._ringRead + size) % size;

if (available < need) {
// Underrun — not enough data. Output what we have, fade to silence.
this._underruns++;
const have = available;
const haveFrames = Math.floor(have / ch);

// Get channel buffers
const outs = [];
for (let c = 0; c < e.outputBuffer.numberOfChannels; c++) {
outs.push(e.outputBuffer.getChannelData(c));
}
this._scheduleChunk(merged);
}

let r = this._ringRead;
_scheduleChunk(samples) {
const ctx = this.audioCtx;
if (!ctx) return;
if (ctx.state === 'suspended') ctx.resume().catch(() => {});

// Play available samples with fade-out over last 64 samples
const fadeLen = Math.min(64, haveFrames);
const fadeStart = haveFrames - fadeLen;
const nFrames = Math.floor(samples.length / this.channels);
if (nFrames === 0) return;

for (let i = 0; i < haveFrames; i++) {
// Fade envelope: full volume, then linear fade to 0
let env = this._fadeGain;
if (i >= fadeStart) {
env *= 1.0 - (i - fadeStart) / fadeLen;
}
for (let c = 0; c < ch; c++) {
if (c < outs.length) {
outs[c][i] = ring[r] * env;
}
r = (r + 1) % size;
}
}
this._ringRead = r;
const audioBuffer = ctx.createBuffer(this.channels, nFrames, this.sampleRate);

// Fill rest with silence
for (let i = haveFrames; i < outLen; i++) {
for (let c = 0; c < outs.length; c++) {
outs[c][i] = 0;
}
// Decode interleaved s16le → per-channel float32
for (let ch = 0; ch < this.channels; ch++) {
const data = audioBuffer.getChannelData(ch);
for (let i = 0; i < nFrames; i++) {
data[i] = samples[i * this.channels + ch] / 32768;
}

this._fadeGain = 0; // next chunk starts silent
this._totalRead += have;
return;
}

// Normal path — enough data available
const outs = [];
for (let c = 0; c < e.outputBuffer.numberOfChannels; c++) {
outs.push(e.outputBuffer.getChannelData(c));
}
const now = ctx.currentTime;

let r = this._ringRead;
// Target latency: 400ms. This means we schedule audio to play 400ms
// from now. Even if the main thread hangs for 300ms, the already-
// scheduled BufferSources continue playing on the system audio thread.
const targetLatency = 0.4;

// If recovering from underrun, fade in over first 64 samples
const fadeInLen = (this._fadeGain < 1.0) ? Math.min(64, outLen) : 0;
// Max buffered: 900ms. Drop chunks if we're too far ahead.
const maxBuffered = 0.9;

for (let i = 0; i < outLen; i++) {
let env = 1.0;
if (i < fadeInLen) {
// Linear fade-in from current _fadeGain to 1.0
env = this._fadeGain + (1.0 - this._fadeGain) * (i / fadeInLen);
}
for (let c = 0; c < ch; c++) {
if (c < outs.length) {
outs[c][i] = ring[r] * env;
if (!this.started || this.nextTime < now) {
// First chunk or underrun.
// Apply fade-in to avoid click at resync point.
const fadeIn = Math.min(64, nFrames);
for (let ch = 0; ch < this.channels; ch++) {
const data = audioBuffer.getChannelData(ch);
for (let i = 0; i < fadeIn; i++) {
data[i] *= i / fadeIn;
}
r = (r + 1) % size;
}
this.nextTime = now + targetLatency;
this.started = true;
}

if (this.nextTime > now + maxBuffered) {
// Too much buffered — drop to cap latency
return;
}

this._ringRead = r;
this._fadeGain = 1.0;
this._totalRead += need;
const source = ctx.createBufferSource();
source.buffer = audioBuffer;
source.connect(ctx.destination);
source.start(this.nextTime);
this.nextTime += audioBuffer.duration;
}
}



+ 128
- 0
web/ring-player-processor.js Прегледај датотеку

@@ -0,0 +1,128 @@
// ring-player-processor.js — AudioWorklet processor for LiveListenWS
// Runs on the audio rendering thread, immune to main-thread blocking.

class RingPlayerProcessor extends AudioWorkletProcessor {
constructor(options) {
super();
const ch = options.processorOptions?.channels || 1;
this._channels = ch;
// 500ms ring buffer at sampleRate
this._ringSize = Math.ceil(sampleRate * ch * 0.5);
this._ring = new Float32Array(this._ringSize);
this._writePos = 0;
this._readPos = 0;
this._started = false;
this._fadeGain = 1.0;
this._startThreshold = Math.ceil(sampleRate * ch * 0.2); // 200ms

this.port.onmessage = (e) => {
if (e.data.type === 'pcm') {
this._pushSamples(e.data.samples);
}
};
}

_available() {
return (this._writePos - this._readPos + this._ringSize) % this._ringSize;
}

_pushSamples(float32arr) {
const ring = this._ring;
const size = this._ringSize;
const n = float32arr.length;

// Overrun: advance read cursor to make room
const used = this._available();
const free = size - used - 1;
if (n > free) {
this._readPos = (this._readPos + (n - free)) % size;
}

let w = this._writePos;
// Fast path: contiguous write
if (w + n <= size) {
ring.set(float32arr, w);
w += n;
if (w >= size) w = 0;
} else {
// Wrap around
const first = size - w;
ring.set(float32arr.subarray(0, first), w);
ring.set(float32arr.subarray(first), 0);
w = n - first;
}
this._writePos = w;

if (!this._started && this._available() >= this._startThreshold) {
this._started = true;
}
}

process(inputs, outputs, parameters) {
const output = outputs[0];
const outLen = output[0]?.length || 128;
const ch = this._channels;
const ring = this._ring;
const size = this._ringSize;

if (!this._started) {
for (let c = 0; c < output.length; c++) output[c].fill(0);
return true;
}

const need = outLen * ch;
const avail = this._available();

if (avail < need) {
// Underrun: play what we have with fade-out, fill rest with silence
const have = avail;
const haveFrames = Math.floor(have / ch);
const fadeLen = Math.min(64, haveFrames);
const fadeStart = haveFrames - fadeLen;
let r = this._readPos;

for (let i = 0; i < haveFrames; i++) {
let env = this._fadeGain;
if (i >= fadeStart) {
env *= 1.0 - (i - fadeStart) / fadeLen;
}
for (let c = 0; c < ch; c++) {
if (c < output.length) {
output[c][i] = ring[r] * env;
}
r = (r + 1) % size;
}
}
this._readPos = r;

// Silence the rest
for (let i = haveFrames; i < outLen; i++) {
for (let c = 0; c < output.length; c++) output[c][i] = 0;
}
this._fadeGain = 0;
return true;
}

// Normal path
let r = this._readPos;
const fadeInLen = (this._fadeGain < 1.0) ? Math.min(64, outLen) : 0;

for (let i = 0; i < outLen; i++) {
let env = 1.0;
if (i < fadeInLen) {
env = this._fadeGain + (1.0 - this._fadeGain) * (i / fadeInLen);
}
for (let c = 0; c < ch; c++) {
if (c < output.length) {
output[c][i] = ring[r] * env;
}
r = (r + 1) % size;
}
}
this._readPos = r;
this._fadeGain = 1.0;
return true;
}
}

registerProcessor('ring-player-processor', RingPlayerProcessor);

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