daniel-c-harvey
5a75da1769
Seek now trims the lead-in so playback lands at the requested time, not the page start; decoder drain polls decodeQueueSize (bounded) instead of a single timeout. Minor cleanups.
316 lines
15 KiB
TypeScript
316 lines
15 KiB
TypeScript
/**
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* OpusStreamDecoder - the WebCodecs streaming Opus decode pipeline.
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*
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* This replaces the fundamentally-broken per-segment `decodeAudioData` Opus model. Instead of cutting
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* the Ogg stream into page-runs and decoding each as a standalone file (which re-applies pre-skip and
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* starts from cold codec state at every boundary), it feeds a single stateful WebCodecs `AudioDecoder`
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* the demuxed Opus packets in order — correct pre-skip-once handling and full inter-frame continuity.
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*
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* Pipeline: OggDemuxer (pages -> Opus packets + granule) -> AudioDecoder (codec 'opus', configured
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* from the OpusHead in the sidecar) -> AudioData (48 kHz PCM) -> AudioBuffer -> PlaybackScheduler.
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*
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* Pre-skip (encoder delay): handled ONCE, by the decoder. WebCodecs decodes Opus with the OpusHead
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* passed as `AudioDecoderConfig.description`; the OpusHead carries `pre_skip`, and the WebCodecs Opus
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* decoder discards those leading samples itself. We do NOT re-trim per packet — doing so on top of the
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* decoder's own trim would double-count. This is the spec-intended path (W3C WebCodecs Opus registration).
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*
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* End-trim: the sidecar's `totalDurationSeconds` is the exact pre-skip-corrected stream length. We cap
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* cumulative emitted audio at that length so the final partial frame's padding does not leak past the
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* true end. (Granule-position end-trim from the EOS page is the alternative; capping on the known total
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* is equivalent and simpler, and the sidecar total is authoritative.)
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*
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* Sample rate: Opus always decodes at 48 kHz (RFC 7845). We force the AudioContext to 48 kHz at init so
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* the decoded AudioData needs no resampling before scheduling — the same `recreateWithSampleRate` seam
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* the WAV path uses for non-44.1 sources.
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*
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* BROWSER-VERIFIED. The actual decode/playback/trim correctness is verified in Daniel's browser
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* (WebCodecs cannot run in Node/headless here). The Ogg demux, packet timing, and end-trim *math* are
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* unit-tested; the WebCodecs glue (configure/decode/flush/AudioData->AudioBuffer) is browser-verified.
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*/
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import { AudioContextManager } from './AudioContextManager.js';
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import { IStreamingDecoder } from './IStreamingDecoder.js';
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import { OggDemuxer, OpusPacket, extractOpusHead, opusHeadChannelCount } from './OggDemuxer.js';
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import { OpusSeekData, OPUS_SAMPLE_RATE } from './OpusSidecar.js';
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/** Opus packet duration ceiling is 120 ms; at 48 kHz that is 5760 frames. Used only for chunk timestamps. */
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const MAX_PACKET_FRAMES = 5760;
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export class OpusStreamDecoder implements IStreamingDecoder {
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private readonly contextManager: AudioContextManager;
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private readonly sidecar: OpusSeekData;
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private demuxer = new OggDemuxer();
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private decoder: AudioDecoder | null = null;
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private channelCount = 2;
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private configured = false;
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// OpusHead bytes used as the AudioDecoder `description`, captured once at first configure and reused
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// verbatim on a range-continuation reconfigure (avoids re-extracting / a non-null assertion).
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private opusHeadDescription: Uint8Array | null = null;
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// Decoded AudioData awaiting conversion, filled by the AudioDecoder output callback.
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private decodedQueue: AudioData[] = [];
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private fatalError = false;
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// Frames to discard from the head of the first post-seek decoded output (AC9 fine re-sync).
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// Set by reinitializeForRangeContinuation to (targetTimeSeconds - landingTimeSeconds) * 48000,
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// consumed frame-by-frame in audioDataToBuffer until exhausted (then zero for the rest of the stream).
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private leadTrimFrames = 0;
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// Monotonic packet timestamp (microseconds) handed to each EncodedAudioChunk. WebCodecs requires
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// strictly increasing timestamps; the true value is irrelevant to us (we schedule by accumulation),
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// so a synthetic 48 kHz-derived counter suffices and stays exact.
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private nextTimestampUs = 0;
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// Cumulative frames already emitted as AudioBuffers, for end-trim against the known total length.
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private emittedFrames = 0;
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private readonly totalFrames: number;
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constructor(contextManager: AudioContextManager, sidecar: OpusSeekData) {
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this.contextManager = contextManager;
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this.sidecar = sidecar;
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this.totalFrames = sidecar.totalDurationSeconds > 0
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? Math.round(sidecar.totalDurationSeconds * OPUS_SAMPLE_RATE)
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: Number.POSITIVE_INFINITY;
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}
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get hasFatalError(): boolean {
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return this.fatalError;
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}
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get ready(): boolean {
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return this.configured;
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}
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get totalDuration(): number | null {
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return this.sidecar.totalDurationSeconds > 0 ? this.sidecar.totalDurationSeconds : null;
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}
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/**
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* Lazily build + configure the WebCodecs decoder from the sidecar's OpusHead. Idempotent. Forces the
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* AudioContext to 48 kHz so decoded AudioData schedules without resampling. Returns false on a config
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* the browser cannot support (caller should never reach here — the capability gate runs first — but
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* we fail safe rather than throw into the stream loop).
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*/
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private async ensureConfigured(): Promise<boolean> {
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if (this.configured) return true;
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if (typeof AudioDecoder === 'undefined') {
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this.fatalError = true;
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return false;
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}
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const opusHead = extractOpusHead(this.sidecar.setupHeaderBytes);
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if (!opusHead) {
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this.fatalError = true;
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return false;
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}
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this.channelCount = opusHeadChannelCount(opusHead);
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// Copy the OpusHead into a standalone buffer — the sidecar subarray is a view we keep.
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this.opusHeadDescription = opusHead.slice();
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// Opus decodes at 48 kHz; align the context so no resample is needed.
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if (this.contextManager.sampleRate !== OPUS_SAMPLE_RATE) {
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await this.contextManager.recreateWithSampleRate(OPUS_SAMPLE_RATE);
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}
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this.decoder = new AudioDecoder({
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output: (data) => this.decodedQueue.push(data),
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error: (err) => {
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console.error('Opus AudioDecoder error:', err.message);
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this.fatalError = true;
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}
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});
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this.decoder.configure(this.buildConfig());
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this.configured = true;
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return true;
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}
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private buildConfig(): AudioDecoderConfig {
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return {
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codec: 'opus',
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sampleRate: OPUS_SAMPLE_RATE,
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numberOfChannels: this.channelCount,
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description: this.opusHeadDescription ?? undefined
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};
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}
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async push(chunk: Uint8Array): Promise<AudioBuffer[]> {
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if (this.fatalError) return [];
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if (!(await this.ensureConfigured())) return [];
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const packets = this.demuxer.push(chunk);
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this.decodePackets(packets);
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// Wait until the WebCodecs decoder has processed the queued packets before draining.
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await this.yieldToDecoder();
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return this.drainDecoded();
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}
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async complete(): Promise<AudioBuffer[]> {
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if (this.fatalError || !this.decoder || this.decoder.state !== 'configured') {
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return this.drainDecoded();
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}
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try {
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await this.decoder.flush();
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} catch (err) {
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// A flush can reject if the decoder was reset/closed concurrently (track switch); the loop's
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// own cancellation handles that — surface nothing, just drain what we have.
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console.warn('Opus decoder flush interrupted:', (err as Error).message);
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}
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return this.drainDecoded();
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}
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/**
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* Reinitialize for a Range-continuation stream after seek-beyond-buffer.
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*
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* @param landingTimeSeconds The actual page-start presentation time resolved from the seek index
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* (t_page ≤ targetTimeSeconds). This is the time at which the decoder
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* will begin emitting audio after reconfigure.
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* @param targetTimeSeconds The user-requested seek position. The difference
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* `(target - landing) * OPUS_SAMPLE_RATE` frames are trimmed from the
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* head of the decoded output so playback lands precisely at the target
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* (AC9 fine re-sync, §3.4a step 4).
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*
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* Pre-skip note: the reconfigure re-applies the WebCodecs Opus decoder's own pre-skip trim. The
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* W3C spec is non-normative on the exact sample count and browsers vary (~312 samples at 48 kHz in
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* practice). `leadTrimFrames` is computed from the sidecar's pre-skip-corrected presentation times
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* (via `presentationTimeSeconds`), so it does NOT double-count the per-reconfigure pre-skip; the
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* decoder handles that internally. If browser testing reveals a residual offset, adjust the
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* `leadTrimFrames` calculation here — this is the single point of control.
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*/
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reinitializeForRangeContinuation(landingTimeSeconds: number, targetTimeSeconds: number): void {
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// New 206 body starts on a page boundary with no setup pages; the codec config is unchanged but
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// inter-frame state must restart cleanly. AudioDecoder.reset() drops queued work and returns the
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// decoder to 'unconfigured', so we reconfigure with the cached config. The demuxer goes into
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// continuation mode (treat the first page's packets as audio — no setup pages in a 206 body).
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this.demuxer.reset(true);
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this.decodedQueue = [];
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this.emittedFrames = 0; // post-seek buffers are positioned by the scheduler's playbackOffset
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// Arm the lead trim: skip enough decoded frames to land at targetTimeSeconds, not at
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// landingTimeSeconds (the page start). Clamp to ≥0 to guard against floating-point rounding.
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this.leadTrimFrames = Math.max(0, Math.round((targetTimeSeconds - landingTimeSeconds) * OPUS_SAMPLE_RATE));
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if (this.decoder && this.decoder.state === 'configured') {
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this.decoder.reset();
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this.decoder.configure(this.buildConfig());
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}
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}
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dispose(): void {
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for (const d of this.decodedQueue) {
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try { d.close(); } catch { /* already closed */ }
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}
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this.decodedQueue = [];
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if (this.decoder && this.decoder.state !== 'closed') {
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try { this.decoder.close(); } catch { /* already closed */ }
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}
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this.decoder = null;
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this.configured = false;
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}
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private decodePackets(packets: OpusPacket[]): void {
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if (!this.decoder || this.decoder.state !== 'configured') return;
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for (const pkt of packets) {
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if (pkt.data.length === 0) continue;
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// Every Opus packet is independently a "key" frame at the container level for WebCodecs's
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// purposes — Opus has no key/delta distinction; 'key' is the correct type for all packets.
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const chunk = new EncodedAudioChunk({
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type: 'key',
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timestamp: this.nextTimestampUs,
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data: pkt.data
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});
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// Advance the synthetic clock by a packet's max duration; exact value is immaterial to us.
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this.nextTimestampUs += Math.round((MAX_PACKET_FRAMES / OPUS_SAMPLE_RATE) * 1_000_000);
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try {
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this.decoder.decode(chunk);
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} catch (err) {
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console.error('Opus decode() threw:', (err as Error).message);
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this.fatalError = true;
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return;
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}
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}
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}
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/**
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* Convert every queued AudioData into an AudioBuffer at the context sample rate, applying
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* end-trim against the known total frame count and lead-trim for post-seek fine re-sync.
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*/
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private drainDecoded(): AudioBuffer[] {
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const out: AudioBuffer[] = [];
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const ctx = this.contextManager.getContext();
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while (this.decodedQueue.length > 0) {
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const data = this.decodedQueue.shift()!;
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try {
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const buffer = this.audioDataToBuffer(ctx, data);
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if (buffer) out.push(buffer);
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} finally {
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try { data.close(); } catch { /* already closed */ }
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}
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}
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return out;
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}
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/**
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* Copy an AudioData's PCM into a new AudioBuffer, applying:
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* 1. Lead-trim (post-seek fine re-sync): skip `leadTrimFrames` from the front so the audible
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* start lands at the requested seek position, not at the preceding page boundary (AC9).
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* 2. End-trim: cap cumulative output at `totalFrames` so the final partial frame's padding
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* does not leak past the true stream end.
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* Returns null when either trim leaves zero usable frames.
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*/
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private audioDataToBuffer(ctx: BaseAudioContext, data: AudioData): AudioBuffer | null {
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const frames = data.numberOfFrames;
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const channels = data.numberOfChannels;
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// Lead-trim: consume frames from the front for post-seek fine re-sync (AC9).
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let skip = 0;
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if (this.leadTrimFrames > 0) {
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skip = Math.min(this.leadTrimFrames, frames);
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this.leadTrimFrames -= skip;
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}
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const available = frames - skip;
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if (available <= 0) return null;
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// End-trim: cap cumulative output at totalFrames.
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let keep = available;
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if (Number.isFinite(this.totalFrames)) {
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const room = this.totalFrames - this.emittedFrames;
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if (room <= 0) return null;
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if (room < available) keep = room;
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}
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if (keep <= 0) return null;
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const buffer = ctx.createBuffer(channels, keep, data.sampleRate);
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// Allocate only for the frames we actually copy; frameOffset skips the lead-trim region.
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const plane = new Float32Array(keep);
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for (let ch = 0; ch < channels; ch++) {
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data.copyTo(plane, { planeIndex: ch, frameOffset: skip, frameCount: keep, format: 'f32-planar' });
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buffer.copyToChannel(plane, ch);
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}
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this.emittedFrames += keep;
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return buffer;
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}
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/**
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* Wait until the AudioDecoder's internal work queue drains (decodeQueueSize → 0), so output
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* callbacks have fired before we drain decodedQueue. Bounded to MAX_YIELD_ITERS × 4 ms to guard
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* against a stuck decoder; any outputs collected before the cap are still returned. `complete()`
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* uses decoder.flush() as its final barrier instead (flush() is the authoritative end-of-stream
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* drain).
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*/
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private yieldToDecoder(): Promise<void> {
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const MAX_YIELD_ITERS = 50; // 50 × 4 ms = 200 ms ceiling
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return new Promise<void>((resolve) => {
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let iters = 0;
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const poll = () => {
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if (!this.decoder || this.decoder.decodeQueueSize === 0 || iters >= MAX_YIELD_ITERS) {
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resolve();
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return;
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}
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iters++;
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setTimeout(poll, 4);
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};
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poll();
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});
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}
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}
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