@blockcast/mmt-render 0.1.0-main.5dd8358ae902

package/dist/audio/audio-render-pipeline.js

@@ -0,0 +1,372 @@
+var __typeError = (msg) => {
+  throw TypeError(msg);
+};
+var __accessCheck = (obj, member, msg) => member.has(obj) || __typeError("Cannot " + msg);
+var __privateGet = (obj, member, getter) => (__accessCheck(obj, member, "read from private field"), getter ? getter.call(obj) : member.get(obj));
+var __privateAdd = (obj, member, value) => member.has(obj) ? __typeError("Cannot add the same private member more than once") : member instanceof WeakSet ? member.add(obj) : member.set(obj, value);
+var __privateSet = (obj, member, value, setter) => (__accessCheck(obj, member, "write to private field"), setter ? setter.call(obj, value) : member.set(obj, value), value);
+var __privateMethod = (obj, member, method) => (__accessCheck(obj, member, "access private method"), method);
+var __privateWrapper = (obj, member, setter, getter) => ({
+  set _(value) {
+    __privateSet(obj, member, value, setter);
+  },
+  get _() {
+    return __privateGet(obj, member, getter);
+  }
+});
+
+// src/audio/audio-render-pipeline.ts
+import { MmtpDecodePipeline } from "@blockcast/mmt-container/mmtp-decode-pipeline";
+
+// src/pacing/frame-pacer.ts
+var _frameCount, _anchorWallMs, _anchored;
+var FrameCountPacer = class {
+  constructor(opts) {
+    __privateAdd(this, _frameCount, 0);
+    __privateAdd(this, _anchorWallMs, 0);
+    __privateAdd(this, _anchored, false);
+    if (!(opts.frameDurationMs > 0)) {
+      throw new Error("frameDurationMs must be > 0");
+    }
+    this.frameDurationMs = opts.frameDurationMs;
+    this.thresholdMs = opts.mapper.pacingThresholdMs;
+    this.warmupFrames = Math.max(0, opts.warmupFrames ?? 0);
+  }
+  /** Call on every emitted decoded frame (AU or picture). Anchors on first call. */
+  notifyFrame() {
+    if (!__privateGet(this, _anchored)) {
+      __privateSet(this, _anchorWallMs, now());
+      __privateSet(this, _anchored, true);
+    }
+    __privateWrapper(this, _frameCount)._++;
+  }
+  /**
+   * Compute sleep ms before reading the next MoQ object.
+   * Returns 0 when:
+   *   - pacing is disabled (NullBlockMapper → threshold=0)
+   *   - the anchor is not yet set
+   *   - still in warmup
+   *   - decoder is not ahead of wall clock by more than threshold
+   * Otherwise returns aheadMs - thresholdMs/2 (leaves half the threshold as headroom).
+   */
+  computeSleepMs() {
+    if (this.thresholdMs <= 0) return 0;
+    if (!__privateGet(this, _anchored)) return 0;
+    if (__privateGet(this, _frameCount) <= this.warmupFrames) return 0;
+    const mediaMs = __privateGet(this, _frameCount) * this.frameDurationMs;
+    const wallMs = now() - __privateGet(this, _anchorWallMs);
+    const aheadMs = mediaMs - wallMs;
+    if (aheadMs <= this.thresholdMs) return 0;
+    return aheadMs - this.thresholdMs / 2;
+  }
+  /** Reset anchor + counter. Call on source switch (MoQ ↔ multicast). */
+  reset() {
+    __privateSet(this, _frameCount, 0);
+    __privateSet(this, _anchorWallMs, 0);
+    __privateSet(this, _anchored, false);
+  }
+  get snapshot() {
+    const mediaMs = __privateGet(this, _frameCount) * this.frameDurationMs;
+    const wallMs = __privateGet(this, _anchored) ? now() - __privateGet(this, _anchorWallMs) : 0;
+    return {
+      frameCount: __privateGet(this, _frameCount),
+      mediaMs,
+      wallMs,
+      aheadMs: mediaMs - wallMs,
+      thresholdMs: this.thresholdMs,
+      anchored: __privateGet(this, _anchored)
+    };
+  }
+};
+_frameCount = new WeakMap();
+_anchorWallMs = new WeakMap();
+_anchored = new WeakMap();
+function now() {
+  if (typeof performance !== "undefined" && typeof performance.now === "function") {
+    return performance.now();
+  }
+  return Date.now();
+}
+
+// src/pacing/au-duration.ts
+function auDurationMsFromCodec(codec, sampleRate) {
+  if (!(sampleRate > 0)) throw new Error("sampleRate must be > 0");
+  if (codec.startsWith("opus") || codec.startsWith("Opus")) {
+    return 960 / sampleRate * 1e3;
+  }
+  if (codec.startsWith("mp4a") || codec.startsWith("AAC") || codec.startsWith("aac")) {
+    return 1024 / sampleRate * 1e3;
+  }
+  if (codec.startsWith("flac") || codec.startsWith("fLaC")) {
+    return 4096 / sampleRate * 1e3;
+  }
+  throw new Error(`unknown audio codec for AU duration derivation: ${codec}`);
+}
+
+// src/audio/worklet-url.ts
+function workletModuleUrl() {
+  const base = new URL("../worklet/render-worklet.js", import.meta.url);
+  return base.href;
+}
+var WORKLET_PROCESSOR_NAME = "mmt-render";
+
+// src/audio/audio-render-pipeline.ts
+var _context, _pacer, _mapper, _codec, _packetId, _sampleRate, _channels, _description, _latencyMs, _startupFrames, _startupThresholdFrames, _capacityFrames, _diag, _warmupFrames, _onState, _onDiag, _onError, _reassemblyPolicy, _decoder, _pipeline, _worklet, _fecClient, _closed, _readyPromise, _workletGeneration, _startupReleased, _AudioRenderPipeline_instances, initialize_fn, onDecoded_fn;
+var AudioRenderPipeline = class {
+  constructor(opts) {
+    __privateAdd(this, _AudioRenderPipeline_instances);
+    __privateAdd(this, _context);
+    __privateAdd(this, _pacer);
+    __privateAdd(this, _mapper);
+    __privateAdd(this, _codec);
+    __privateAdd(this, _packetId);
+    __privateAdd(this, _sampleRate);
+    __privateAdd(this, _channels);
+    __privateAdd(this, _description);
+    __privateAdd(this, _latencyMs);
+    __privateAdd(this, _startupFrames);
+    __privateAdd(this, _startupThresholdFrames);
+    __privateAdd(this, _capacityFrames);
+    __privateAdd(this, _diag);
+    __privateAdd(this, _warmupFrames);
+    __privateAdd(this, _onState);
+    __privateAdd(this, _onDiag);
+    __privateAdd(this, _onError);
+    __privateAdd(this, _reassemblyPolicy);
+    __privateAdd(this, _decoder, null);
+    __privateAdd(this, _pipeline, null);
+    __privateAdd(this, _worklet, null);
+    __privateAdd(this, _fecClient);
+    __privateAdd(this, _closed, false);
+    __privateAdd(this, _readyPromise, null);
+    __privateAdd(this, _workletGeneration, 0);
+    __privateAdd(this, _startupReleased, false);
+    __privateSet(this, _context, opts.context);
+    __privateSet(this, _mapper, opts.mapper);
+    __privateSet(this, _codec, opts.codec);
+    __privateSet(this, _packetId, opts.packetId);
+    __privateSet(this, _sampleRate, opts.sampleRate);
+    __privateSet(this, _channels, opts.channels);
+    __privateSet(this, _description, opts.description);
+    __privateSet(this, _latencyMs, opts.latencyMs);
+    __privateSet(this, _startupFrames, opts.startupFrames);
+    __privateSet(this, _startupThresholdFrames, opts.startupFrames ?? Math.ceil(opts.sampleRate * (opts.latencyMs / 1e3)));
+    __privateSet(this, _capacityFrames, opts.capacityFrames);
+    __privateSet(this, _diag, opts.diag ?? false);
+    __privateSet(this, _warmupFrames, opts.warmupFrames ?? 0);
+    __privateSet(this, _onState, opts.onState);
+    __privateSet(this, _onDiag, opts.onDiag);
+    __privateSet(this, _onError, opts.onError);
+    __privateSet(this, _fecClient, opts.fecClient);
+    __privateSet(this, _reassemblyPolicy, opts.reassemblyPolicy);
+    const auDurationMs = auDurationMsFromCodec(opts.codec, opts.sampleRate);
+    __privateSet(this, _pacer, new FrameCountPacer({
+      frameDurationMs: auDurationMs,
+      mapper: opts.mapper,
+      warmupFrames: __privateGet(this, _warmupFrames)
+    }));
+  }
+  /** Load worklet module + create decoder + pipeline. Call once. */
+  async ready() {
+    if (__privateGet(this, _readyPromise)) return __privateGet(this, _readyPromise);
+    __privateSet(this, _readyPromise, __privateMethod(this, _AudioRenderPipeline_instances, initialize_fn).call(this));
+    return __privateGet(this, _readyPromise);
+  }
+  /** Feed one MoQ object (raw MMTP packet). */
+  feedMoqObject(packet) {
+    __privateGet(this, _pipeline)?.feedMoqObject(packet);
+  }
+  /** ms to sleep before the next object read. 0 when not anchored / not ahead. */
+  paceSleepMs() {
+    return __privateGet(this, _pacer).computeSleepMs();
+  }
+  /** Pacer snapshot (for diagnostics / UI). */
+  get pacerSnapshot() {
+    return __privateGet(this, _pacer).snapshot;
+  }
+  /** Reset pacer on source switch (MoQ ↔ multicast). */
+  resetPacer() {
+    __privateGet(this, _pacer).reset();
+  }
+  /** Drop queued PCM and its timestamp epoch at an explicit source boundary. */
+  resetPlayout(startUnstalled = false) {
+    __privateGet(this, _pacer).reset();
+    __privateSet(this, _startupReleased, true);
+    const message = {
+      type: "reset",
+      generation: ++__privateWrapper(this, _workletGeneration)._,
+      startUnstalled
+    };
+    __privateGet(this, _worklet)?.port.postMessage(message);
+  }
+  /** Node to connect to destination / gain. */
+  get rootNode() {
+    if (!__privateGet(this, _worklet)) throw new Error("AudioRenderPipeline.ready() must resolve first");
+    return __privateGet(this, _worklet);
+  }
+  close() {
+    if (__privateGet(this, _closed)) return;
+    __privateSet(this, _closed, true);
+    try {
+      __privateGet(this, _decoder)?.close();
+    } catch {
+    }
+    try {
+      __privateGet(this, _pipeline)?.dispose();
+    } catch {
+    }
+    try {
+      __privateGet(this, _worklet)?.disconnect();
+    } catch {
+    }
+    __privateSet(this, _decoder, null);
+    __privateSet(this, _pipeline, null);
+    __privateSet(this, _worklet, null);
+  }
+};
+_context = new WeakMap();
+_pacer = new WeakMap();
+_mapper = new WeakMap();
+_codec = new WeakMap();
+_packetId = new WeakMap();
+_sampleRate = new WeakMap();
+_channels = new WeakMap();
+_description = new WeakMap();
+_latencyMs = new WeakMap();
+_startupFrames = new WeakMap();
+_startupThresholdFrames = new WeakMap();
+_capacityFrames = new WeakMap();
+_diag = new WeakMap();
+_warmupFrames = new WeakMap();
+_onState = new WeakMap();
+_onDiag = new WeakMap();
+_onError = new WeakMap();
+_reassemblyPolicy = new WeakMap();
+_decoder = new WeakMap();
+_pipeline = new WeakMap();
+_worklet = new WeakMap();
+_fecClient = new WeakMap();
+_closed = new WeakMap();
+_readyPromise = new WeakMap();
+_workletGeneration = new WeakMap();
+_startupReleased = new WeakMap();
+_AudioRenderPipeline_instances = new WeakSet();
+initialize_fn = async function() {
+  await __privateGet(this, _context).audioWorklet.addModule(workletModuleUrl());
+  const worklet = new AudioWorkletNode(__privateGet(this, _context), WORKLET_PROCESSOR_NAME, {
+    numberOfInputs: 0,
+    numberOfOutputs: 1,
+    outputChannelCount: [__privateGet(this, _channels)]
+  });
+  const initMsg = {
+    type: "init",
+    generation: ++__privateWrapper(this, _workletGeneration)._,
+    channels: __privateGet(this, _channels),
+    rate: __privateGet(this, _sampleRate),
+    latencyMs: __privateGet(this, _latencyMs),
+    startupFrames: __privateGet(this, _startupFrames),
+    capacityFrames: __privateGet(this, _capacityFrames),
+    holdUntilStart: true,
+    diag: __privateGet(this, _diag)
+  };
+  worklet.port.postMessage(initMsg);
+  worklet.port.onmessage = (ev) => {
+    var _a, _b;
+    const m = ev.data;
+    if (m.type === "state") {
+      if (m.generation !== __privateGet(this, _workletGeneration)) return;
+      (_a = __privateGet(this, _onState)) == null ? void 0 : _a.call(this, {
+        timestampUs: m.timestamp,
+        stalled: m.stalled,
+        bufferedFrames: m.bufferedFrames,
+        writtenFrames: m.writtenFrames
+      });
+    } else if (m.type === "ringBufDiag") {
+      if (m.generation !== __privateGet(this, _workletGeneration)) return;
+      (_b = __privateGet(this, _onDiag)) == null ? void 0 : _b.call(this, {
+        reads: m.reads,
+        avgFill: m.avgFill,
+        capacity: m.capacity,
+        skipsPerSec: m.skipsPerSec,
+        dupsPerSec: m.dupsPerSec,
+        overflowSamplesPerSec: m.overflowSamplesPerSec,
+        underflowSamplesPerSec: m.underflowSamplesPerSec
+      });
+    } else if (m.type === "writeAck") {
+      if (m.generation !== __privateGet(this, _workletGeneration) || __privateGet(this, _startupReleased) || m.bufferedFrames < __privateGet(this, _startupThresholdFrames)) return;
+      __privateSet(this, _startupReleased, true);
+      const start = {
+        type: "start",
+        generation: __privateGet(this, _workletGeneration)
+      };
+      worklet.port.postMessage(start);
+    }
+  };
+  __privateSet(this, _worklet, worklet);
+  const decoder = new AudioDecoder({
+    output: (data) => __privateMethod(this, _AudioRenderPipeline_instances, onDecoded_fn).call(this, data),
+    error: (err) => {
+      var _a;
+      return (_a = __privateGet(this, _onError)) == null ? void 0 : _a.call(this, err);
+    }
+  });
+  const config = {
+    codec: __privateGet(this, _codec),
+    sampleRate: __privateGet(this, _sampleRate),
+    numberOfChannels: __privateGet(this, _channels),
+    ...__privateGet(this, _description) ? { description: __privateGet(this, _description) } : {}
+  };
+  decoder.configure(config);
+  __privateSet(this, _decoder, decoder);
+  const pipeline = new MmtpDecodePipeline();
+  pipeline.configure({
+    codec: __privateGet(this, _codec),
+    packetId: __privateGet(this, _packetId),
+    description: __privateGet(this, _description),
+    reorderWindow: 0,
+    // audio doesn't need reorder — AAC/Opus frames are self-contained
+    reassemblyPolicy: __privateGet(this, _reassemblyPolicy)
+  });
+  if (__privateGet(this, _fecClient)) pipeline.setFecClient(__privateGet(this, _fecClient));
+  pipeline.onFrame((codecData, meta) => {
+    var _a;
+    if (!__privateGet(this, _decoder) || __privateGet(this, _decoder).state !== "configured") return;
+    if (codecData.byteLength === 0) return;
+    __privateGet(this, _pacer).notifyFrame();
+    try {
+      __privateGet(this, _decoder).decode(new EncodedAudioChunk({
+        type: "key",
+        data: codecData,
+        timestamp: meta.timestamp
+      }));
+    } catch (err) {
+      (_a = __privateGet(this, _onError)) == null ? void 0 : _a.call(this, err);
+    }
+  });
+  __privateSet(this, _pipeline, pipeline);
+};
+// ─── Private ─────────────────────────────────────────────────────
+onDecoded_fn = function(sample) {
+  if (__privateGet(this, _closed) || !__privateGet(this, _worklet)) {
+    sample.close();
+    return;
+  }
+  const timestamp = sample.timestamp;
+  const channelData = [];
+  for (let ch = 0; ch < sample.numberOfChannels; ch++) {
+    const data = new Float32Array(sample.numberOfFrames);
+    sample.copyTo(data, { format: "f32-planar", planeIndex: ch });
+    channelData.push(data);
+  }
+  sample.close();
+  const msg = {
+    type: "data",
+    data: channelData,
+    timestamp
+  };
+  __privateGet(this, _worklet).port.postMessage(msg, channelData.map((d) => d.buffer));
+};
+export {
+  AudioRenderPipeline
+};
+//# sourceMappingURL=audio-render-pipeline.js.map

package/dist/audio/audio-render-pipeline.js.map

@@ -0,0 +1,7 @@
+{
+  "version": 3,
+  "sources": ["../../src/audio/audio-render-pipeline.ts", "../../src/pacing/frame-pacer.ts", "../../src/pacing/au-duration.ts", "../../src/audio/worklet-url.ts"],
+  "sourcesContent": ["/**\n * AudioRenderPipeline \u2014 end-to-end audio render for MoQ/MMT players.\n *\n * Pipeline: MoQ object \u2192 MmtpDecodePipeline \u2192 WebCodecs AudioDecoder\n *           \u2192 AudioWorkletNode (exact-timestamp FIFO)\n *\n * The consumer:\n *   1. constructs with codec/sampleRate/channels/mapper from catalog\n *   2. awaits ready()\n *   3. calls feedMoqObject(packet) for each arriving MoQ object\n *   4. between objects, awaits paceSleepMs() to throttle burst delivery\n *   5. connects rootNode to an AudioContext destination (or a GainNode)\n *   6. close() on teardown\n *\n * Pacing uses FrameCountPacer (anchor on first decoded AU, frame count \u00D7 AU\n * duration) with threshold = mapper.pacingThresholdMs \u2014 NOT MPU timestamps\n * (ambiguous under FEC interleave).\n */\n\nimport { MmtpDecodePipeline, type PipelineFecClient } from '@blockcast/mmt-container/mmtp-decode-pipeline'\nimport type { MfuReassemblyPolicy } from '@blockcast/mmt-container/mfu-reassembly-policy'\nimport type { FecBlockMapper } from '@blockcast/mmt-fec'\nimport { FrameCountPacer } from '../pacing/frame-pacer.js'\nimport { auDurationMsFromCodec } from '../pacing/au-duration.js'\nimport type {\n  ToMain,\n  DataMessage,\n  InitMessage,\n  ResetMessage,\n  StartMessage,\n} from './render-messages.js'\nimport { WORKLET_PROCESSOR_NAME, workletModuleUrl } from './worklet-url.js'\n\nexport interface AudioRenderPipelineOpts {\n  context: AudioContext\n  codec: string\n  /** MMTP packet_id for this audio track \u2014 from catalog. */\n  packetId?: number\n  sampleRate: number\n  channels: number\n  description?: Uint8Array\n  mapper: FecBlockMapper\n  fecClient?: PipelineFecClient\n  /** Reliable MoQ is clock-free; datagram deadlines come from catalog/in-band timing. */\n  reassemblyPolicy: MfuReassemblyPolicy\n  /** Catalog-derived PCM startup buffer in milliseconds. */\n  latencyMs: number\n  /** Exact PCM startup threshold when catalog grouping is codec-aligned. */\n  startupFrames?: number\n  /** Exact PCM storage ceiling, including one successor delivery burst. */\n  capacityFrames?: number\n  /** Enable ring-buffer telemetry messages. Default false. */\n  diag?: boolean\n  /** Called when worklet reports state (playback timestamp + exact buffer accounting). Optional. */\n  onState?: (state: {\n    timestampUs: number | undefined\n    stalled: boolean\n    bufferedFrames: number\n    writtenFrames: number\n  }) => void\n  /** Called when worklet reports ring-buffer diag (telemetry). Optional. */\n  onDiag?: (diag: {\n    reads: number; avgFill: number; capacity: number\n    skipsPerSec: number; dupsPerSec: number; overflowSamplesPerSec: number\n    underflowSamplesPerSec: number\n  }) => void\n  /** Warmup frames before pacing engages. Default 0 (audio tolerates instant pacing). */\n  warmupFrames?: number\n  /** Audio underflow / config event logger. Optional. */\n  onError?: (err: unknown) => void\n}\n\nexport class AudioRenderPipeline {\n  readonly #context: AudioContext\n  readonly #pacer: FrameCountPacer\n  readonly #mapper: FecBlockMapper\n  readonly #codec: string\n  readonly #packetId: number | undefined\n  readonly #sampleRate: number\n  readonly #channels: number\n  readonly #description?: Uint8Array\n  readonly #latencyMs: number\n  readonly #startupFrames?: number\n  readonly #startupThresholdFrames: number\n  readonly #capacityFrames?: number\n  readonly #diag: boolean\n  readonly #warmupFrames: number\n  readonly #onState?: AudioRenderPipelineOpts['onState']\n  readonly #onDiag?: AudioRenderPipelineOpts['onDiag']\n  readonly #onError?: AudioRenderPipelineOpts['onError']\n  readonly #reassemblyPolicy: MfuReassemblyPolicy\n\n  #decoder: AudioDecoder | null = null\n  #pipeline: MmtpDecodePipeline | null = null\n  #worklet: AudioWorkletNode | null = null\n  #fecClient?: PipelineFecClient\n  #closed = false\n  #readyPromise: Promise<void> | null = null\n  #workletGeneration = 0\n  #startupReleased = false\n\n  constructor(opts: AudioRenderPipelineOpts) {\n    this.#context = opts.context\n    this.#mapper = opts.mapper\n    this.#codec = opts.codec\n    this.#packetId = opts.packetId\n    this.#sampleRate = opts.sampleRate\n    this.#channels = opts.channels\n    this.#description = opts.description\n    this.#latencyMs = opts.latencyMs\n    this.#startupFrames = opts.startupFrames\n    this.#startupThresholdFrames = opts.startupFrames\n      ?? Math.ceil(opts.sampleRate * (opts.latencyMs / 1000))\n    this.#capacityFrames = opts.capacityFrames\n    this.#diag = opts.diag ?? false\n    this.#warmupFrames = opts.warmupFrames ?? 0\n    this.#onState = opts.onState\n    this.#onDiag = opts.onDiag\n    this.#onError = opts.onError\n    this.#fecClient = opts.fecClient\n    this.#reassemblyPolicy = opts.reassemblyPolicy\n\n    // AU duration from codec + sampleRate. Keep inside the lib so consumers\n    // don't re-derive it.\n    const auDurationMs = auDurationMsFromCodec(opts.codec, opts.sampleRate)\n    this.#pacer = new FrameCountPacer({\n      frameDurationMs: auDurationMs,\n      mapper: opts.mapper,\n      warmupFrames: this.#warmupFrames,\n    })\n  }\n\n  /** Load worklet module + create decoder + pipeline. Call once. */\n  async ready(): Promise<void> {\n    if (this.#readyPromise) return this.#readyPromise\n    this.#readyPromise = this.#initialize()\n    return this.#readyPromise\n  }\n\n  async #initialize(): Promise<void> {\n    // 1. Load shared worklet module.\n    await this.#context.audioWorklet.addModule(workletModuleUrl())\n\n    // 2. Create worklet node.\n    const worklet = new AudioWorkletNode(this.#context, WORKLET_PROCESSOR_NAME, {\n      numberOfInputs: 0,\n      numberOfOutputs: 1,\n      outputChannelCount: [this.#channels],\n    })\n    const initMsg: InitMessage = {\n      type: 'init',\n      generation: ++this.#workletGeneration,\n      channels: this.#channels,\n      rate: this.#sampleRate,\n      latencyMs: this.#latencyMs,\n      startupFrames: this.#startupFrames,\n      capacityFrames: this.#capacityFrames,\n      holdUntilStart: true,\n      diag: this.#diag,\n    }\n    worklet.port.postMessage(initMsg)\n    worklet.port.onmessage = (ev: MessageEvent<ToMain>) => {\n      const m = ev.data\n      if (m.type === 'state') {\n        if (m.generation !== this.#workletGeneration) return\n        this.#onState?.({\n          timestampUs: m.timestamp,\n          stalled: m.stalled,\n          bufferedFrames: m.bufferedFrames,\n          writtenFrames: m.writtenFrames,\n        })\n      } else if (m.type === 'ringBufDiag') {\n        if (m.generation !== this.#workletGeneration) return\n        this.#onDiag?.({\n          reads: m.reads, avgFill: m.avgFill, capacity: m.capacity,\n          skipsPerSec: m.skipsPerSec, dupsPerSec: m.dupsPerSec,\n          overflowSamplesPerSec: m.overflowSamplesPerSec,\n          underflowSamplesPerSec: m.underflowSamplesPerSec,\n        })\n      } else if (m.type === 'writeAck') {\n        if (\n          m.generation !== this.#workletGeneration ||\n          this.#startupReleased ||\n          m.bufferedFrames < this.#startupThresholdFrames\n        ) return\n        this.#startupReleased = true\n        const start: StartMessage = {\n          type: 'start',\n          generation: this.#workletGeneration,\n        }\n        worklet.port.postMessage(start)\n      }\n    }\n    this.#worklet = worklet\n\n    // 3. Create decoder.\n    const decoder = new AudioDecoder({\n      output: (data) => this.#onDecoded(data),\n      error: (err) => this.#onError?.(err),\n    })\n    const config: AudioDecoderConfig = {\n      codec: this.#codec,\n      sampleRate: this.#sampleRate,\n      numberOfChannels: this.#channels,\n      ...(this.#description ? { description: this.#description } : {}),\n    }\n    decoder.configure(config)\n    this.#decoder = decoder\n\n    // 4. Create decode pipeline.\n    const pipeline = new MmtpDecodePipeline()\n    pipeline.configure({\n      codec: this.#codec,\n      packetId: this.#packetId,\n      description: this.#description,\n      reorderWindow: 0, // audio doesn't need reorder \u2014 AAC/Opus frames are self-contained\n      reassemblyPolicy: this.#reassemblyPolicy,\n    })\n    if (this.#fecClient) pipeline.setFecClient(this.#fecClient)\n    pipeline.onFrame((codecData, meta) => {\n      if (!this.#decoder || this.#decoder.state !== 'configured') return\n      if (codecData.byteLength === 0) return\n      this.#pacer.notifyFrame()\n      try {\n        this.#decoder.decode(new EncodedAudioChunk({\n          type: 'key',\n          data: codecData,\n          timestamp: meta.timestamp,\n        }))\n      } catch (err) {\n        this.#onError?.(err)\n      }\n    })\n    this.#pipeline = pipeline\n  }\n\n  /** Feed one MoQ object (raw MMTP packet). */\n  feedMoqObject(packet: Uint8Array): void {\n    this.#pipeline?.feedMoqObject(packet)\n  }\n\n  /** ms to sleep before the next object read. 0 when not anchored / not ahead. */\n  paceSleepMs(): number {\n    return this.#pacer.computeSleepMs()\n  }\n\n  /** Pacer snapshot (for diagnostics / UI). */\n  get pacerSnapshot() {\n    return this.#pacer.snapshot\n  }\n\n  /** Reset pacer on source switch (MoQ \u2194 multicast). */\n  resetPacer(): void {\n    this.#pacer.reset()\n  }\n\n  /** Drop queued PCM and its timestamp epoch at an explicit source boundary. */\n  resetPlayout(startUnstalled = false): void {\n    this.#pacer.reset()\n    // The explicit gate owns only the initial prefill. Source-boundary resets\n    // use the worklet's one-frame resume threshold.\n    this.#startupReleased = true\n    const message: ResetMessage = {\n      type: 'reset',\n      generation: ++this.#workletGeneration,\n      startUnstalled,\n    }\n    this.#worklet?.port.postMessage(message)\n  }\n\n  /** Node to connect to destination / gain. */\n  get rootNode(): AudioNode {\n    if (!this.#worklet) throw new Error('AudioRenderPipeline.ready() must resolve first')\n    return this.#worklet\n  }\n\n  close(): void {\n    if (this.#closed) return\n    this.#closed = true\n    try { this.#decoder?.close() } catch { /* already closed */ }\n    try { this.#pipeline?.dispose() } catch { /* noop */ }\n    try { this.#worklet?.disconnect() } catch { /* noop */ }\n    this.#decoder = null\n    this.#pipeline = null\n    this.#worklet = null\n  }\n\n  // \u2500\u2500\u2500 Private \u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\u2500\n\n  #onDecoded(sample: AudioData): void {\n    if (this.#closed || !this.#worklet) { sample.close(); return }\n\n    const timestamp = sample.timestamp\n    const channelData: Float32Array[] = []\n    for (let ch = 0; ch < sample.numberOfChannels; ch++) {\n      const data = new Float32Array(sample.numberOfFrames)\n      sample.copyTo(data, { format: 'f32-planar', planeIndex: ch })\n      channelData.push(data)\n    }\n    sample.close()\n\n    const msg: DataMessage = {\n      type: 'data',\n      data: channelData,\n      timestamp,\n    }\n    this.#worklet.port.postMessage(msg, channelData.map((d) => d.buffer))\n  }\n}\n\n// auDurationMsFromCodec lives in ../pacing/au-duration.ts (canonical, no\n// mmt-container dep). Import it from '@blockcast/mmt-render/pacing' directly\n// or via the top-level '@blockcast/mmt-render' export.\n", "/**\n * FrameCountPacer \u2014 media-agnostic wall-clock pacer for MMT A/V decode loops.\n *\n * Why not use MPU timestamps? Under FEC interleaving, encoders pack multiple\n * Access Units (AUs) per MMTP MPU with a single mpu_timestamp. The observed\n * timestamp delta between MPUs is a fraction of the real per-AU duration\n * (e.g. 10.22ms between MPUs for 21.33ms AAC frames at D=2 interleave).\n * Pacing on that delta sees \"media \u2248 wall\" and never sleeps, letting burst\n * relay delivery saturate the decoder / ring buffer.\n *\n * The Sony libatsc3 pattern (atsc3_mmt_context_mfu_depacketizer) solves this\n * by anchoring wall-clock time to the first received AU and counting AUs:\n *   media_elapsed_ms = au_count \u00D7 nominal_au_duration_ms\n * The MPU timestamp is used elsewhere as an absolute-time anchor, never as\n * an incremental clock.\n *\n * The sleep threshold comes from the catalog's FEC interleaveDepthMs (via\n * FecBlockMapper.pacingThresholdMs). When no FEC is configured, a\n * NullBlockMapper yields threshold=0 and this pacer never sleeps.\n *\n * One pacer instance = one track (e.g. audio, video) on one player.\n */\n\nimport type { FecBlockMapper } from '@blockcast/mmt-fec'\n\nexport interface FrameCountPacerOpts {\n  /** Required: AU duration (audio) or picture interval (video), in ms. */\n  frameDurationMs: number\n  /** FEC mapper from @blockcast/mmt-fec. NullBlockMapper disables pacing. */\n  mapper: FecBlockMapper\n  /**\n   * Number of frames to let through before pacing starts. Lets startup\n   * burst drain populate the ring buffer / decoder queue. Default 0.\n   */\n  warmupFrames?: number\n}\n\nexport interface PacerSnapshot {\n  /** Frames observed (from notifyFrame). */\n  frameCount: number\n  /** Computed media time: frameCount \u00D7 frameDurationMs. */\n  mediaMs: number\n  /** Wall-clock time since anchor. */\n  wallMs: number\n  /** mediaMs - wallMs. Positive \u2192 decoder ahead of realtime. */\n  aheadMs: number\n  /** Pacing threshold (ms) from the FEC mapper. */\n  thresholdMs: number\n  /** Has the first frame been observed (wall anchor set)? */\n  anchored: boolean\n}\n\nexport class FrameCountPacer {\n  readonly frameDurationMs: number\n  readonly thresholdMs: number\n  readonly warmupFrames: number\n\n  #frameCount = 0\n  #anchorWallMs = 0\n  #anchored = false\n\n  constructor(opts: FrameCountPacerOpts) {\n    if (!(opts.frameDurationMs > 0)) {\n      throw new Error('frameDurationMs must be > 0')\n    }\n    this.frameDurationMs = opts.frameDurationMs\n    this.thresholdMs = opts.mapper.pacingThresholdMs\n    this.warmupFrames = Math.max(0, opts.warmupFrames ?? 0)\n  }\n\n  /** Call on every emitted decoded frame (AU or picture). Anchors on first call. */\n  notifyFrame(): void {\n    if (!this.#anchored) {\n      this.#anchorWallMs = now()\n      this.#anchored = true\n    }\n    this.#frameCount++\n  }\n\n  /**\n   * Compute sleep ms before reading the next MoQ object.\n   * Returns 0 when:\n   *   - pacing is disabled (NullBlockMapper \u2192 threshold=0)\n   *   - the anchor is not yet set\n   *   - still in warmup\n   *   - decoder is not ahead of wall clock by more than threshold\n   * Otherwise returns aheadMs - thresholdMs/2 (leaves half the threshold as headroom).\n   */\n  computeSleepMs(): number {\n    if (this.thresholdMs <= 0) return 0\n    if (!this.#anchored) return 0\n    if (this.#frameCount <= this.warmupFrames) return 0\n\n    const mediaMs = this.#frameCount * this.frameDurationMs\n    const wallMs = now() - this.#anchorWallMs\n    const aheadMs = mediaMs - wallMs\n    if (aheadMs <= this.thresholdMs) return 0\n    return aheadMs - this.thresholdMs / 2\n  }\n\n  /** Reset anchor + counter. Call on source switch (MoQ \u2194 multicast). */\n  reset(): void {\n    this.#frameCount = 0\n    this.#anchorWallMs = 0\n    this.#anchored = false\n  }\n\n  get snapshot(): PacerSnapshot {\n    const mediaMs = this.#frameCount * this.frameDurationMs\n    const wallMs = this.#anchored ? now() - this.#anchorWallMs : 0\n    return {\n      frameCount: this.#frameCount,\n      mediaMs,\n      wallMs,\n      aheadMs: mediaMs - wallMs,\n      thresholdMs: this.thresholdMs,\n      anchored: this.#anchored,\n    }\n  }\n}\n\nfunction now(): number {\n  // performance.now() is available in main thread, workers, and AudioWorklet.\n  // Fall back to Date.now() for exotic runtimes (shouldn't happen in browsers).\n  if (typeof performance !== 'undefined' && typeof performance.now === 'function') {\n    return performance.now()\n  }\n  return Date.now()\n}\n", "/**\n * Nominal Access Unit duration helpers for common audio + video codecs.\n *\n * Kept in the pacing subpath (no mmt-container dep) so lean consumers like\n * moqtail's player.ts can import the pacer + AU duration without pulling\n * in the full audio-render-pipeline (which depends on @blockcast/mmt-container).\n */\n\n/**\n * Nominal AU duration in ms for common audio codecs at the given sample rate.\n * AAC-LC: 1024 samples. Opus (default 20ms): 960 samples @ 48kHz.\n * Throws for unknown codecs so callers supply correct timing.\n */\nexport function auDurationMsFromCodec(codec: string, sampleRate: number): number {\n  if (!(sampleRate > 0)) throw new Error('sampleRate must be > 0')\n  if (codec.startsWith('opus') || codec.startsWith('Opus')) {\n    return (960 / sampleRate) * 1000\n  }\n  if (codec.startsWith('mp4a') || codec.startsWith('AAC') || codec.startsWith('aac')) {\n    return (1024 / sampleRate) * 1000\n  }\n  if (codec.startsWith('flac') || codec.startsWith('fLaC')) {\n    return (4096 / sampleRate) * 1000\n  }\n  throw new Error(`unknown audio codec for AU duration derivation: ${codec}`)\n}\n\n/**\n * Nominal picture duration in ms from framerate.\n */\nexport function pictureDurationMs(framerate: number): number {\n  if (!(framerate > 0)) throw new Error('framerate must be > 0')\n  return 1000 / framerate\n}\n", "/**\n * Helper for loading the prebuilt AudioWorklet bundle.\n *\n * Two equivalent loading patterns supported:\n *\n *   1. Vite-style static URL (recommended for bundled apps):\n *        import workletUrl from '@blockcast/mmt-render/audio/worklet.js?url'\n *        await ctx.audioWorklet.addModule(workletUrl)\n *      Vite resolves the `?url` suffix + the package's \"./audio/worklet.js\"\n *      export to the prebuilt IIFE file.\n *\n *   2. Runtime import.meta.url resolution (no bundler):\n *        import { workletModuleUrl } from '@blockcast/mmt-render/audio/worklet-url'\n *        await ctx.audioWorklet.addModule(workletModuleUrl())\n *\n * Both resolve to dist/worklet/render-worklet.js.\n */\n\n/**\n * Returns a URL (or URL string) pointing to the prebuilt worklet bundle.\n * Resolves relative to this module's location via import.meta.url.\n */\nexport function workletModuleUrl(): string {\n  // dist layout: dist/audio/worklet-url.js \u2192 dist/worklet/render-worklet.js\n  const base = new URL('../worklet/render-worklet.js', import.meta.url)\n  return base.href\n}\n\n/** Processor name registered in the worklet bundle. */\nexport const WORKLET_PROCESSOR_NAME = 'mmt-render'\n"],
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+}

package/dist/audio/audio-ring-buffer.d.ts

@@ -0,0 +1,55 @@
+/**
+ * Timestamp-anchored FIFO shared by MMT AudioWorklet consumers.
+ *
+ * PCM remains in decoder emission order. Media timestamps are exact anchors
+ * for playout reporting only; they never reposition samples or trigger a
+ * guessed discontinuity. The FIFO starts after its full catalog-derived
+ * latency is buffered and performs no sample skip/dup clock recovery.
+ */
+export interface AudioRingBufferConfig {
+    channels: number;
+    rate: number;
+    latencyMs: number;
+    /** Exact PCM startup threshold. Overrides latencyMs when provided. */
+    startupFrames?: number;
+    /** PCM refill threshold after the first underrun. Defaults to startupFrames. */
+    resumeFrames?: number;
+    /** Exact PCM storage ceiling. Must cover the startup and resume thresholds. */
+    capacityFrames?: number;
+    startUnstalled?: boolean;
+    /** Deprecated compatibility input. Clock recovery is not performed. */
+    warmupReads?: number;
+    diag?: boolean;
+}
+export interface AudioRingBufferDiag {
+    reads: number;
+    avgFill: number;
+    capacity: number;
+    skips: number;
+    dups: number;
+    overflowSamples: number;
+}
+export declare class AudioRingBuffer {
+    #private;
+    readonly channels: number;
+    readonly rate: number;
+    constructor(config: AudioRingBufferConfig);
+    get stalled(): boolean;
+    get startupFrames(): number;
+    get capacity(): number;
+    get length(): number;
+    /** Timestamp of the next media sample after the PCM already emitted. */
+    get timestampUs(): number | undefined;
+    write(timestampUs: number, data: Float32Array[]): void;
+    read(output: Float32Array[]): number;
+    /** Clear queued PCM; optionally treat the next fill as post-start recovery. */
+    reset(startUnstalled?: boolean, resumeAfterReset?: boolean): void;
+    /**
+     * Change the startup latency while preserving the configured storage
+     * headroom above its buffering thresholds. Active playout keeps draining
+     * retained PCM; a stalled buffer waits for its startup or resume threshold.
+     */
+    resize(latencyMs: number): void;
+    getDiagAndReset(): AudioRingBufferDiag | null;
+}
+//# sourceMappingURL=audio-ring-buffer.d.ts.map

package/dist/audio/audio-ring-buffer.d.ts.map

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