@blockcast/mmt-render 0.1.0 → 0.3.0-main.d9354616fdf2

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

@@ -1,7 +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"],
-  "mappings": 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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   *   - the caller reports a standing backlog (see `observedDelayMs`)\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   * `observedDelayMs` is how far this frame is behind the producer's own clock,\n   * measured by the caller against a producer-supplied sending instant (for\n   * MMTP, the header send-wall timestamp). It is optional because the anchored\n   * rate below is blind to it: `#anchorWallMs` is the wall time of the FIRST\n   * frame read, so `aheadMs` measures consumption against the join instant and\n   * not against the producer. Once `aheadMs` reaches the threshold the loop\n   * consumes at exactly the source rate, and recovering latency present at join\n   * requires consuming FASTER than the source rate. So without this argument a\n   * consumer that joined a live stream mid-block keeps that join latency for the\n   * life of the subscription.\n   *\n   * A caller that can measure the real distance from the producer passes it and\n   * gets a drain: while the backlog exceeds the threshold there is nothing to\n   * pace against, so return 0 and let the transport set the rate. At the live\n   * edge the anchored-rate behavior below applies unchanged.\n   *\n   * Callers with no producer clock omit it and keep the pure ATSC3-style\n   * anchored pacing (see the class doc), where a first-frame anchor is correct\n   * because a broadcast push has no live edge to catch up to.\n   */\n  computeSleepMs(observedDelayMs?: number): number {\n    if (this.thresholdMs <= 0) return 0\n    if (!this.#anchored) return 0\n    if (this.#frameCount <= this.warmupFrames) return 0\n    // Only a finite measurement can license the drain. NaN/Infinity mean the\n    // caller could not measure the distance, which is not the same claim as\n    // \"the consumer is at the edge\", so fall through to anchored pacing.\n    if (observedDelayMs !== undefined && Number.isFinite(observedDelayMs) && observedDelayMs > this.thresholdMs) {\n      return 0\n    }\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.js

@@ -1,39 +1,20 @@
-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-ring-buffer.ts
-var _buffer, _writeIndex, _readIndex, _stalled, _startupFrames, _resumeFrames, _resumeFollowsStartup, _hasStarted, _capacityHeadroomFrames, _timestampAnchors, _diagEnabled, _diagReads, _diagFillSum, _diagOverflowSamples, _AudioRingBuffer_instances, pruneTimestampAnchors_fn, refillFrames_fn;
 var AudioRingBuffer = class {
+  #buffer;
+  #writeIndex = 0;
+  #readIndex = 0;
+  #stalled;
+  #startupFrames;
+  #resumeFrames;
+  #resumeFollowsStartup;
+  #hasStarted;
+  #capacityHeadroomFrames;
+  #timestampAnchors = [];
+  #diagEnabled;
+  #diagReads = 0;
+  #diagFillSum = 0;
+  #diagOverflowSamples = 0;
   constructor(config) {
-    __privateAdd(this, _AudioRingBuffer_instances);
-    __privateAdd(this, _buffer);
-    __privateAdd(this, _writeIndex, 0);
-    __privateAdd(this, _readIndex, 0);
-    __privateAdd(this, _stalled);
-    __privateAdd(this, _startupFrames);
-    __privateAdd(this, _resumeFrames);
-    __privateAdd(this, _resumeFollowsStartup);
-    __privateAdd(this, _hasStarted);
-    __privateAdd(this, _capacityHeadroomFrames);
-    __privateAdd(this, _timestampAnchors, []);
-    __privateAdd(this, _diagEnabled);
-    __privateAdd(this, _diagReads, 0);
-    __privateAdd(this, _diagFillSum, 0);
-    __privateAdd(this, _diagOverflowSamples, 0);
     if (!Number.isInteger(config.channels) || config.channels <= 0) {
       throw new Error("invalid channels");
     }
@@ -59,35 +40,35 @@ var AudioRingBuffer = class {
     }
     this.channels = config.channels;
     this.rate = config.rate;
-    __privateSet(this, _startupFrames, startupFrames);
-    __privateSet(this, _resumeFrames, resumeFrames);
-    __privateSet(this, _resumeFollowsStartup, config.resumeFrames === void 0);
-    __privateSet(this, _hasStarted, config.startUnstalled ?? false);
-    __privateSet(this, _capacityHeadroomFrames, capacityFrames - thresholdFrames);
-    __privateSet(this, _stalled, !(config.startUnstalled ?? false));
-    __privateSet(this, _diagEnabled, config.diag ?? false);
-    __privateSet(this, _buffer, Array.from(
+    this.#startupFrames = startupFrames;
+    this.#resumeFrames = resumeFrames;
+    this.#resumeFollowsStartup = config.resumeFrames === void 0;
+    this.#hasStarted = config.startUnstalled ?? false;
+    this.#capacityHeadroomFrames = capacityFrames - thresholdFrames;
+    this.#stalled = !(config.startUnstalled ?? false);
+    this.#diagEnabled = config.diag ?? false;
+    this.#buffer = Array.from(
       { length: config.channels },
       () => new Float32Array(capacityFrames)
-    ));
+    );
   }
   get stalled() {
-    return __privateGet(this, _stalled);
+    return this.#stalled;
   }
   get startupFrames() {
-    return __privateGet(this, _startupFrames);
+    return this.#startupFrames;
   }
   get capacity() {
-    return __privateGet(this, _buffer)[0].length;
+    return this.#buffer[0].length;
   }
   get length() {
-    return __privateGet(this, _writeIndex) - __privateGet(this, _readIndex);
+    return this.#writeIndex - this.#readIndex;
   }
   /** Timestamp of the next media sample after the PCM already emitted. */
   get timestampUs() {
-    const anchor = __privateGet(this, _timestampAnchors)[0];
-    if (!anchor || anchor.sampleIndex > __privateGet(this, _readIndex)) return void 0;
-    return anchor.timestampUs + (__privateGet(this, _readIndex) - anchor.sampleIndex) / this.rate * 1e6;
+    const anchor = this.#timestampAnchors[0];
+    if (!anchor || anchor.sampleIndex > this.#readIndex) return void 0;
+    return anchor.timestampUs + (this.#readIndex - anchor.sampleIndex) / this.rate * 1e6;
   }
   write(timestampUs, data) {
     if (data.length !== this.channels) throw new Error("wrong channel count");
@@ -97,68 +78,68 @@ var AudioRingBuffer = class {
     if (data.some((channel) => channel.length !== samples)) {
       throw new Error("mismatched channel lengths");
     }
-    const start = __privateGet(this, _writeIndex);
+    const start = this.#writeIndex;
     const end = start + samples;
-    __privateGet(this, _timestampAnchors).push({ sampleIndex: start, timestampUs });
-    const overflow = end - __privateGet(this, _readIndex) - this.capacity;
+    this.#timestampAnchors.push({ sampleIndex: start, timestampUs });
+    const overflow = end - this.#readIndex - this.capacity;
     if (overflow > 0) {
-      __privateSet(this, _stalled, false);
-      __privateSet(this, _hasStarted, true);
-      __privateSet(this, _readIndex, __privateGet(this, _readIndex) + overflow);
-      if (__privateGet(this, _diagEnabled)) __privateSet(this, _diagOverflowSamples, __privateGet(this, _diagOverflowSamples) + overflow);
-      __privateMethod(this, _AudioRingBuffer_instances, pruneTimestampAnchors_fn).call(this);
+      this.#stalled = false;
+      this.#hasStarted = true;
+      this.#readIndex += overflow;
+      if (this.#diagEnabled) this.#diagOverflowSamples += overflow;
+      this.#pruneTimestampAnchors();
     }
-    const refillFrames = __privateGet(this, _hasStarted) ? __privateGet(this, _resumeFrames) : __privateGet(this, _startupFrames);
-    if (__privateGet(this, _stalled) && end - __privateGet(this, _readIndex) >= refillFrames) {
-      __privateSet(this, _stalled, false);
-      __privateSet(this, _hasStarted, true);
+    const refillFrames = this.#hasStarted ? this.#resumeFrames : this.#startupFrames;
+    if (this.#stalled && end - this.#readIndex >= refillFrames) {
+      this.#stalled = false;
+      this.#hasStarted = true;
     }
     for (let channel = 0; channel < this.channels; channel++) {
       const src = data[channel];
-      const dst = __privateGet(this, _buffer)[channel];
+      const dst = this.#buffer[channel];
       for (let i = 0; i < samples; i++) {
         dst[(start + i) % dst.length] = src[i];
       }
     }
-    __privateSet(this, _writeIndex, end);
+    this.#writeIndex = end;
   }
   read(output) {
     if (output.length !== this.channels) throw new Error("wrong channel count");
-    if (__privateGet(this, _stalled)) return 0;
+    if (this.#stalled) return 0;
     const available = this.length;
     if (available === 0) {
-      __privateSet(this, _stalled, true);
+      this.#stalled = true;
       return 0;
     }
     const requested = output[0].length;
     const readSamples = Math.min(available, requested);
     for (let channel = 0; channel < this.channels; channel++) {
       const dst = output[channel];
-      const src = __privateGet(this, _buffer)[channel];
+      const src = this.#buffer[channel];
       for (let i = 0; i < readSamples; i++) {
-        dst[i] = src[(__privateGet(this, _readIndex) + i) % src.length];
+        dst[i] = src[(this.#readIndex + i) % src.length];
       }
       for (let i = readSamples; i < requested; i++) dst[i] = 0;
     }
-    __privateSet(this, _readIndex, __privateGet(this, _readIndex) + readSamples);
-    __privateMethod(this, _AudioRingBuffer_instances, pruneTimestampAnchors_fn).call(this);
-    if (readSamples < requested) __privateSet(this, _stalled, true);
-    if (__privateGet(this, _diagEnabled)) {
-      __privateWrapper(this, _diagReads)._++;
-      __privateSet(this, _diagFillSum, __privateGet(this, _diagFillSum) + available);
+    this.#readIndex += readSamples;
+    this.#pruneTimestampAnchors();
+    if (readSamples < requested) this.#stalled = true;
+    if (this.#diagEnabled) {
+      this.#diagReads++;
+      this.#diagFillSum += available;
     }
     return readSamples;
   }
   /** Clear queued PCM; optionally treat the next fill as post-start recovery. */
   reset(startUnstalled = false, resumeAfterReset = false) {
-    __privateSet(this, _writeIndex, 0);
-    __privateSet(this, _readIndex, 0);
-    __privateSet(this, _stalled, !startUnstalled);
-    __privateSet(this, _hasStarted, startUnstalled || resumeAfterReset);
-    __privateSet(this, _timestampAnchors, []);
-    __privateSet(this, _diagReads, 0);
-    __privateSet(this, _diagFillSum, 0);
-    __privateSet(this, _diagOverflowSamples, 0);
+    this.#writeIndex = 0;
+    this.#readIndex = 0;
+    this.#stalled = !startUnstalled;
+    this.#hasStarted = startUnstalled || resumeAfterReset;
+    this.#timestampAnchors = [];
+    this.#diagReads = 0;
+    this.#diagFillSum = 0;
+    this.#diagOverflowSamples = 0;
   }
   /**
    * Change the startup latency while preserving the configured storage
@@ -173,18 +154,18 @@ var AudioRingBuffer = class {
     if (!Number.isSafeInteger(newStartupFrames) || newStartupFrames <= 0) {
       throw new Error("invalid latency");
     }
-    const newResumeFrames = __privateGet(this, _resumeFollowsStartup) ? newStartupFrames : __privateGet(this, _resumeFrames);
-    const newCapacity = Math.max(newStartupFrames, newResumeFrames) + __privateGet(this, _capacityHeadroomFrames);
+    const newResumeFrames = this.#resumeFollowsStartup ? newStartupFrames : this.#resumeFrames;
+    const newCapacity = Math.max(newStartupFrames, newResumeFrames) + this.#capacityHeadroomFrames;
     if (!Number.isSafeInteger(newCapacity) || newCapacity < Math.max(newStartupFrames, newResumeFrames)) {
       throw new Error("invalid capacity frames");
     }
-    const wasStalled = __privateGet(this, _stalled);
-    __privateSet(this, _startupFrames, newStartupFrames);
-    __privateSet(this, _resumeFrames, newResumeFrames);
+    const wasStalled = this.#stalled;
+    this.#startupFrames = newStartupFrames;
+    this.#resumeFrames = newResumeFrames;
     if (newCapacity === this.capacity) {
-      if (wasStalled && this.length >= __privateMethod(this, _AudioRingBuffer_instances, refillFrames_fn).call(this)) {
-        __privateSet(this, _stalled, false);
-        __privateSet(this, _hasStarted, true);
+      if (wasStalled && this.length >= this.#refillFrames()) {
+        this.#stalled = false;
+        this.#hasStarted = true;
       }
       return;
     }
@@ -193,61 +174,46 @@ var AudioRingBuffer = class {
       () => new Float32Array(newCapacity)
     );
     const samplesToKeep = Math.min(this.length, newCapacity);
-    const copyStart = __privateGet(this, _writeIndex) - samplesToKeep;
+    const copyStart = this.#writeIndex - samplesToKeep;
     for (let channel = 0; channel < this.channels; channel++) {
-      const src = __privateGet(this, _buffer)[channel];
+      const src = this.#buffer[channel];
       const dst = next[channel];
       for (let i = 0; i < samplesToKeep; i++) {
         const sampleIndex = copyStart + i;
         dst[sampleIndex % dst.length] = src[sampleIndex % src.length];
       }
     }
-    __privateSet(this, _buffer, next);
-    __privateSet(this, _readIndex, copyStart);
-    __privateMethod(this, _AudioRingBuffer_instances, pruneTimestampAnchors_fn).call(this);
-    __privateSet(this, _stalled, samplesToKeep === 0 || wasStalled && samplesToKeep < __privateMethod(this, _AudioRingBuffer_instances, refillFrames_fn).call(this));
-    if (!__privateGet(this, _stalled)) __privateSet(this, _hasStarted, true);
+    this.#buffer = next;
+    this.#readIndex = copyStart;
+    this.#pruneTimestampAnchors();
+    this.#stalled = samplesToKeep === 0 || wasStalled && samplesToKeep < this.#refillFrames();
+    if (!this.#stalled) this.#hasStarted = true;
   }
   getDiagAndReset() {
-    if (!__privateGet(this, _diagEnabled) || __privateGet(this, _diagReads) === 0) return null;
+    if (!this.#diagEnabled || this.#diagReads === 0) return null;
     const result = {
-      reads: __privateGet(this, _diagReads),
-      avgFill: Math.round(__privateGet(this, _diagFillSum) / __privateGet(this, _diagReads)),
+      reads: this.#diagReads,
+      avgFill: Math.round(this.#diagFillSum / this.#diagReads),
       capacity: this.capacity,
       skips: 0,
       dups: 0,
-      overflowSamples: __privateGet(this, _diagOverflowSamples)
+      overflowSamples: this.#diagOverflowSamples
     };
-    __privateSet(this, _diagReads, 0);
-    __privateSet(this, _diagFillSum, 0);
-    __privateSet(this, _diagOverflowSamples, 0);
+    this.#diagReads = 0;
+    this.#diagFillSum = 0;
+    this.#diagOverflowSamples = 0;
     return result;
   }
-};
-_buffer = new WeakMap();
-_writeIndex = new WeakMap();
-_readIndex = new WeakMap();
-_stalled = new WeakMap();
-_startupFrames = new WeakMap();
-_resumeFrames = new WeakMap();
-_resumeFollowsStartup = new WeakMap();
-_hasStarted = new WeakMap();
-_capacityHeadroomFrames = new WeakMap();
-_timestampAnchors = new WeakMap();
-_diagEnabled = new WeakMap();
-_diagReads = new WeakMap();
-_diagFillSum = new WeakMap();
-_diagOverflowSamples = new WeakMap();
-_AudioRingBuffer_instances = new WeakSet();
-pruneTimestampAnchors_fn = function() {
-  let keep = 0;
-  while (keep + 1 < __privateGet(this, _timestampAnchors).length && __privateGet(this, _timestampAnchors)[keep + 1].sampleIndex <= __privateGet(this, _readIndex)) {
-    keep++;
+  #pruneTimestampAnchors() {
+    let keep = 0;
+    while (keep + 1 < this.#timestampAnchors.length && this.#timestampAnchors[keep + 1].sampleIndex <= this.#readIndex) {
+      keep++;
+    }
+    if (keep > 0) this.#timestampAnchors.splice(0, keep);
+  }
+  #refillFrames() {
+    return this.#hasStarted ? this.#resumeFrames : this.#startupFrames;
   }
-  if (keep > 0) __privateGet(this, _timestampAnchors).splice(0, keep);
-};
-refillFrames_fn = function() {
-  return __privateGet(this, _hasStarted) ? __privateGet(this, _resumeFrames) : __privateGet(this, _startupFrames);
 };
 export {
   AudioRingBuffer

package/dist/audio/audio-ring-buffer.js.map

@@ -2,6 +2,6 @@
   "version": 3,
   "sources": ["../../src/audio/audio-ring-buffer.ts"],
   "sourcesContent": ["/**\n * Timestamp-anchored FIFO shared by MMT AudioWorklet consumers.\n *\n * PCM remains in decoder emission order. Media timestamps are exact anchors\n * for playout reporting only; they never reposition samples or trigger a\n * guessed discontinuity. The FIFO starts after its full catalog-derived\n * latency is buffered and performs no sample skip/dup clock recovery.\n */\n\nexport interface AudioRingBufferConfig {\n  channels: number\n  rate: number\n  latencyMs: number\n  /** Exact PCM startup threshold. Overrides latencyMs when provided. */\n  startupFrames?: number\n  /** PCM refill threshold after the first underrun. Defaults to startupFrames. */\n  resumeFrames?: number\n  /** Exact PCM storage ceiling. Must cover the startup and resume thresholds. */\n  capacityFrames?: number\n  startUnstalled?: boolean\n  /** Deprecated compatibility input. Clock recovery is not performed. */\n  warmupReads?: number\n  diag?: boolean\n}\n\nexport interface AudioRingBufferDiag {\n  reads: number\n  avgFill: number\n  capacity: number\n  skips: number\n  dups: number\n  overflowSamples: number\n}\n\ntype TimestampAnchor = {\n  sampleIndex: number\n  timestampUs: number\n}\n\nexport class AudioRingBuffer {\n  readonly channels: number\n  readonly rate: number\n\n  #buffer: Float32Array[]\n  #writeIndex = 0\n  #readIndex = 0\n  #stalled: boolean\n  #startupFrames: number\n  #resumeFrames: number\n  #resumeFollowsStartup: boolean\n  #hasStarted: boolean\n  #capacityHeadroomFrames: number\n  #timestampAnchors: TimestampAnchor[] = []\n\n  #diagEnabled: boolean\n  #diagReads = 0\n  #diagFillSum = 0\n  #diagOverflowSamples = 0\n\n  constructor(config: AudioRingBufferConfig) {\n    if (!Number.isInteger(config.channels) || config.channels <= 0) {\n      throw new Error('invalid channels')\n    }\n    if (!Number.isFinite(config.rate) || config.rate <= 0) {\n      throw new Error('invalid sample rate')\n    }\n    if (!Number.isFinite(config.latencyMs) || config.latencyMs <= 0) {\n      throw new Error('invalid latency')\n    }\n\n    const latencyFrames = Math.ceil(config.rate * (config.latencyMs / 1000))\n    const startupFrames = config.startupFrames ?? latencyFrames\n    const resumeFrames = config.resumeFrames ?? startupFrames\n    const thresholdFrames = Math.max(startupFrames, resumeFrames)\n    const capacityFrames = config.capacityFrames ?? Math.max(latencyFrames, thresholdFrames)\n    if (!Number.isSafeInteger(startupFrames) || startupFrames <= 0) {\n      throw new Error('invalid startup frames')\n    }\n    if (!Number.isSafeInteger(resumeFrames) || resumeFrames <= 0) {\n      throw new Error('invalid resume frames')\n    }\n    if (!Number.isSafeInteger(capacityFrames) || capacityFrames < thresholdFrames) {\n      throw new Error('invalid capacity frames')\n    }\n\n    this.channels = config.channels\n    this.rate = config.rate\n    this.#startupFrames = startupFrames\n    this.#resumeFrames = resumeFrames\n    this.#resumeFollowsStartup = config.resumeFrames === undefined\n    this.#hasStarted = config.startUnstalled ?? false\n    this.#capacityHeadroomFrames = capacityFrames - thresholdFrames\n    this.#stalled = !(config.startUnstalled ?? false)\n    this.#diagEnabled = config.diag ?? false\n    this.#buffer = Array.from(\n      { length: config.channels },\n      () => new Float32Array(capacityFrames),\n    )\n  }\n\n  get stalled(): boolean { return this.#stalled }\n  get startupFrames(): number { return this.#startupFrames }\n  get capacity(): number { return this.#buffer[0].length }\n  get length(): number { return this.#writeIndex - this.#readIndex }\n\n  /** Timestamp of the next media sample after the PCM already emitted. */\n  get timestampUs(): number | undefined {\n    const anchor = this.#timestampAnchors[0]\n    if (!anchor || anchor.sampleIndex > this.#readIndex) return undefined\n    return anchor.timestampUs + ((this.#readIndex - anchor.sampleIndex) / this.rate) * 1_000_000\n  }\n\n  write(timestampUs: number, data: Float32Array[]): void {\n    if (data.length !== this.channels) throw new Error('wrong channel count')\n    if (!Number.isFinite(timestampUs)) throw new Error('invalid timestamp')\n    const samples = data[0].length\n    if (samples === 0) return\n    if (data.some((channel) => channel.length !== samples)) {\n      throw new Error('mismatched channel lengths')\n    }\n\n    const start = this.#writeIndex\n    const end = start + samples\n    this.#timestampAnchors.push({ sampleIndex: start, timestampUs })\n\n    const overflow = end - this.#readIndex - this.capacity\n    if (overflow > 0) {\n      this.#stalled = false\n      this.#hasStarted = true\n      this.#readIndex += overflow\n      if (this.#diagEnabled) this.#diagOverflowSamples += overflow\n      this.#pruneTimestampAnchors()\n    }\n\n    const refillFrames = this.#hasStarted ? this.#resumeFrames : this.#startupFrames\n    if (this.#stalled && end - this.#readIndex >= refillFrames) {\n      this.#stalled = false\n      this.#hasStarted = true\n    }\n\n    for (let channel = 0; channel < this.channels; channel++) {\n      const src = data[channel]\n      const dst = this.#buffer[channel]\n      for (let i = 0; i < samples; i++) {\n        dst[(start + i) % dst.length] = src[i]\n      }\n    }\n    this.#writeIndex = end\n  }\n\n  read(output: Float32Array[]): number {\n    if (output.length !== this.channels) throw new Error('wrong channel count')\n    if (this.#stalled) return 0\n\n    const available = this.length\n    if (available === 0) {\n      this.#stalled = true\n      return 0\n    }\n\n    const requested = output[0].length\n    const readSamples = Math.min(available, requested)\n\n    for (let channel = 0; channel < this.channels; channel++) {\n      const dst = output[channel]\n      const src = this.#buffer[channel]\n      for (let i = 0; i < readSamples; i++) {\n        dst[i] = src[(this.#readIndex + i) % src.length]\n      }\n      for (let i = readSamples; i < requested; i++) dst[i] = 0\n    }\n\n    this.#readIndex += readSamples\n    this.#pruneTimestampAnchors()\n    if (readSamples < requested) this.#stalled = true\n\n    if (this.#diagEnabled) {\n      this.#diagReads++\n      this.#diagFillSum += available\n    }\n    return readSamples\n  }\n\n  /** Clear queued PCM; optionally treat the next fill as post-start recovery. */\n  reset(startUnstalled = false, resumeAfterReset = false): void {\n    this.#writeIndex = 0\n    this.#readIndex = 0\n    this.#stalled = !startUnstalled\n    this.#hasStarted = startUnstalled || resumeAfterReset\n    this.#timestampAnchors = []\n    this.#diagReads = 0\n    this.#diagFillSum = 0\n    this.#diagOverflowSamples = 0\n  }\n\n  /**\n   * Change the startup latency while preserving the configured storage\n   * headroom above its buffering thresholds. Active playout keeps draining\n   * retained PCM; a stalled buffer waits for its startup or resume threshold.\n   */\n  resize(latencyMs: number): void {\n    if (!Number.isFinite(latencyMs) || latencyMs <= 0) {\n      throw new Error('invalid latency')\n    }\n    const newStartupFrames = Math.ceil(this.rate * (latencyMs / 1000))\n    if (!Number.isSafeInteger(newStartupFrames) || newStartupFrames <= 0) {\n      throw new Error('invalid latency')\n    }\n    const newResumeFrames = this.#resumeFollowsStartup ? newStartupFrames : this.#resumeFrames\n    const newCapacity =\n      Math.max(newStartupFrames, newResumeFrames) + this.#capacityHeadroomFrames\n    if (\n      !Number.isSafeInteger(newCapacity) ||\n      newCapacity < Math.max(newStartupFrames, newResumeFrames)\n    ) {\n      throw new Error('invalid capacity frames')\n    }\n    const wasStalled = this.#stalled\n    this.#startupFrames = newStartupFrames\n    this.#resumeFrames = newResumeFrames\n    if (newCapacity === this.capacity) {\n      if (wasStalled && this.length >= this.#refillFrames()) {\n        this.#stalled = false\n        this.#hasStarted = true\n      }\n      return\n    }\n\n    const next = Array.from(\n      { length: this.channels },\n      () => new Float32Array(newCapacity),\n    )\n    const samplesToKeep = Math.min(this.length, newCapacity)\n    const copyStart = this.#writeIndex - samplesToKeep\n    for (let channel = 0; channel < this.channels; channel++) {\n      const src = this.#buffer[channel]\n      const dst = next[channel]\n      for (let i = 0; i < samplesToKeep; i++) {\n        const sampleIndex = copyStart + i\n        dst[sampleIndex % dst.length] = src[sampleIndex % src.length]\n      }\n    }\n\n    this.#buffer = next\n    this.#readIndex = copyStart\n    this.#pruneTimestampAnchors()\n    this.#stalled = samplesToKeep === 0 || (wasStalled && samplesToKeep < this.#refillFrames())\n    if (!this.#stalled) this.#hasStarted = true\n  }\n\n  getDiagAndReset(): AudioRingBufferDiag | null {\n    if (!this.#diagEnabled || this.#diagReads === 0) return null\n    const result: AudioRingBufferDiag = {\n      reads: this.#diagReads,\n      avgFill: Math.round(this.#diagFillSum / this.#diagReads),\n      capacity: this.capacity,\n      skips: 0,\n      dups: 0,\n      overflowSamples: this.#diagOverflowSamples,\n    }\n    this.#diagReads = 0\n    this.#diagFillSum = 0\n    this.#diagOverflowSamples = 0\n    return result\n  }\n\n  #pruneTimestampAnchors(): void {\n    let keep = 0\n    while (\n      keep + 1 < this.#timestampAnchors.length &&\n      this.#timestampAnchors[keep + 1].sampleIndex <= this.#readIndex\n    ) {\n      keep++\n    }\n    if (keep > 0) this.#timestampAnchors.splice(0, keep)\n  }\n\n  #refillFrames(): number {\n    return this.#hasStarted ? this.#resumeFrames : this.#startupFrames\n  }\n}\n"],
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