npm - @catlabtech/webcvt-transcode - Versions diffs - 0.2.0 - Mend

@catlabtech/webcvt-transcode 0.2.0

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package/dist/index.cjs.map ADDED Viewed

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This is the format\n * `AudioData.copyTo('f32-planar')` yields and `new AudioData({format:\n * 'f32-planar'})` accepts, so it bridges the decoder output and the encoder\n * input, and interleaves cleanly to int16 for the WAV sink.\n */\n\nimport type { WavFile } from '@catlabtech/webcvt-container-wav';\n\n/** Planar 32-bit-float PCM. `channels[c][f]` is sample `f` of channel `c`. */\nexport interface DecodedAudio {\n readonly sampleRate: number;\n readonly numberOfChannels: number;\n /** Frames (samples per channel). */\n readonly numberOfFrames: number;\n /** One `Float32Array` of length `numberOfFrames` per channel. */\n readonly channels: readonly Float32Array[];\n}\n\n// ---------------------------------------------------------------------------\n// AudioData → planar float (decode side)\n// ---------------------------------------------------------------------------\n\n/** Copy every channel of an `AudioData` out as planar `Float32Array`s. */\nexport function readAudioDataPlanes(data: AudioData): Float32Array[] {\n const frames = data.numberOfFrames;\n const planes: Float32Array[] = [];\n for (let ch = 0; ch < data.numberOfChannels; ch++) {\n const plane = new Float32Array(frames);\n data.copyTo(plane, { planeIndex: ch, format: 'f32-planar' });\n planes.push(plane);\n }\n return planes;\n}\n\n/**\n * Accumulates the planar `AudioData` chunks a decoder emits into a single\n * {@link DecodedAudio}. Channel count / sample rate are taken from the first\n * chunk seen.\n */\nexport class PlanarAccumulator {\n #perChannel: Float32Array[][] = [];\n #frames = 0;\n #numChannels = 0;\n #sampleRate = 0;\n\n add(planes: Float32Array[], sampleRate: number): void {\n if (this.#numChannels === 0) {\n this.#numChannels = planes.length;\n this.#sampleRate = sampleRate;\n this.#perChannel = planes.map(() => []);\n }\n for (let ch = 0; ch < this.#numChannels; ch++) {\n this.#perChannel[ch]?.push(planes[ch] ?? planes[0] ?? new Float32Array(0));\n }\n this.#frames += planes[0]?.length ?? 0;\n }\n\n finish(fallbackSampleRate: number, fallbackChannels: number): DecodedAudio {\n if (this.#numChannels === 0) {\n return {\n sampleRate: fallbackSampleRate,\n numberOfChannels: fallbackChannels,\n numberOfFrames: 0,\n channels: Array.from({ length: fallbackChannels }, () => new Float32Array(0)),\n };\n }\n const channels = this.#perChannel.map((chunks) => concatFloat(chunks, this.#frames));\n return {\n sampleRate: this.#sampleRate,\n numberOfChannels: this.#numChannels,\n numberOfFrames: this.#frames,\n channels,\n };\n }\n}\n\n// ---------------------------------------------------------------------------\n// WAV raw PCM → planar float (wav is a pcm source; no decoder needed)\n// ---------------------------------------------------------------------------\n\nconst WAVE_FORMAT_IEEE_FLOAT = 3;\n\n/** Decode a parsed WAV's interleaved PCM into planar float. */\nexport function wavToDecoded(file: WavFile): DecodedAudio {\n const { format, audioData } = file;\n const ch = Math.max(1, format.channels);\n const bits = format.bitsPerSample;\n const bytesPerSample = bits / 8;\n const frameSize = bytesPerSample * ch;\n const frames = frameSize > 0 ? Math.floor(audioData.length / frameSize) : 0;\n const isFloat = format.audioFormat === WAVE_FORMAT_IEEE_FLOAT;\n const view = new DataView(audioData.buffer, audioData.byteOffset, audioData.byteLength);\n const channels = Array.from({ length: ch }, () => new Float32Array(frames));\n\n for (let f = 0; f < frames; f++) {\n for (let c = 0; c < ch; c++) {\n const off = f * frameSize + c * bytesPerSample;\n const target = channels[c];\n if (target) target[f] = readSample(view, off, bits, isFloat);\n }\n }\n return { sampleRate: format.sampleRate, numberOfChannels: ch, numberOfFrames: frames, channels };\n}\n\n// ---------------------------------------------------------------------------\n// Planar float → interleaved int16 (WAV sink)\n// ---------------------------------------------------------------------------\n\n/** Interleave planar float to little-endian 16-bit PCM bytes for `serializeWav`. */\nexport function interleaveInt16(decoded: DecodedAudio): Uint8Array {\n const { numberOfChannels: ch, numberOfFrames: n, channels } = decoded;\n const out = new Uint8Array(n * ch * 2);\n const view = new DataView(out.buffer);\n let pos = 0;\n for (let f = 0; f < n; f++) {\n for (let c = 0; c < ch; c++) {\n view.setInt16(pos, floatToInt16(channels[c]?.[f] ?? 0), true);\n pos += 2;\n }\n }\n return out;\n}\n\n// ---------------------------------------------------------------------------\n// Planar float → AudioData (encode side)\n// ---------------------------------------------------------------------------\n\n/**\n * Build an `AudioData` from a frame window of a {@link DecodedAudio}, in\n * `f32-planar` layout, for feeding an `AudioEncoder`. The encoder takes\n * ownership and closes it.\n */\nexport function buildAudioData(\n decoded: DecodedAudio,\n startFrame: number,\n frameCount: number,\n timestampUs: number,\n): AudioData {\n const ch = decoded.numberOfChannels;\n const data = new Float32Array(ch * frameCount);\n for (let c = 0; c < ch; c++) {\n const plane = decoded.channels[c];\n if (plane) data.set(plane.subarray(startFrame, startFrame + frameCount), c * frameCount);\n }\n return new AudioData({\n format: 'f32-planar',\n sampleRate: decoded.sampleRate,\n numberOfFrames: frameCount,\n numberOfChannels: ch,\n timestamp: timestampUs,\n data,\n });\n}\n\n// ---------------------------------------------------------------------------\n// Internal helpers\n// ---------------------------------------------------------------------------\n\nfunction concatFloat(chunks: Float32Array[], total: number): Float32Array {\n const out = new Float32Array(total);\n let off = 0;\n for (const c of chunks) {\n out.set(c, off);\n off += c.length;\n }\n return out;\n}\n\nfunction floatToInt16(v: number): number {\n const s = Math.round(v * 32767);\n return s < -32768 ? -32768 : s > 32767 ? 32767 : s;\n}\n\nfunction readSample(view: DataView, off: number, bits: number, isFloat: boolean): number {\n if (isFloat) {\n return bits === 64 ? view.getFloat64(off, true) : view.getFloat32(off, true);\n }\n switch (bits) {\n case 8:\n return ((view.getUint8(off) - 128) / 128) as number;\n case 16:\n return view.getInt16(off, true) / 32768;\n case 24: {\n const b0 = view.getUint8(off);\n const b1 = view.getUint8(off + 1);\n const b2 = view.getUint8(off + 2);\n let val = b0 | (b1 << 8) | (b2 << 16);\n if (val & 0x800000) val -= 0x1000000; // sign-extend 24-bit\n return val / 8388608;\n }\n case 32:\n return view.getInt32(off, true) / 2147483648;\n default:\n return 0;\n }\n}\n","/**\n * Decode driver: turn a {@link DemuxResult} into planar-float {@link DecodedAudio}.\n *\n * PCM inputs (wav) short-circuit — no WebCodecs needed. Encoded inputs feed a\n * `WebCodecsAudioDecoder` chunk-by-chunk, accumulating each emitted `AudioData`\n * and `close()`-ing it promptly. The decoder is always `close()`-d in `finally`\n * (and via an abort listener that unblocks a pending `flush()`), so no codec\n * surface leaks on error or cancellation.\n */\n\nimport { WebCodecsAudioDecoder } from '@catlabtech/webcvt-codec-webcodecs';\nimport { asCodecError, throwIfAborted } from './abort.ts';\nimport type { DemuxResult } from './demux.ts';\nimport { type DecodedAudio, PlanarAccumulator, readAudioDataPlanes } from './pcm.ts';\n\nexport interface DecodeContext {\n readonly signal?: AbortSignal;\n /** Progress callback: (framesDecoded fraction 0–1). */\n readonly onProgress?: (fraction: number) => void;\n}\n\n/** Decode a demuxed source to planar float PCM. */\nexport async function decodeToPcm(demux: DemuxResult, ctx: DecodeContext): Promise<DecodedAudio> {\n throwIfAborted(ctx.signal);\n\n if (demux.kind === 'pcm') {\n ctx.onProgress?.(1);\n return demux.decoded;\n }\n\n const { config, chunks, sampleRate, numberOfChannels } = demux;\n const acc = new PlanarAccumulator();\n\n const decoder = new WebCodecsAudioDecoder({ config }, (data) => {\n acc.add(readAudioDataPlanes(data), data.sampleRate);\n data.close();\n });\n\n const onAbort = (): void => decoder.close();\n ctx.signal?.addEventListener('abort', onAbort, { once: true });\n\n try {\n const total = chunks.length || 1;\n for (let i = 0; i < chunks.length; i++) {\n throwIfAborted(ctx.signal);\n const spec = chunks[i];\n if (!spec) continue;\n decoder.decode(\n new EncodedAudioChunk({\n type: spec.type,\n timestamp: spec.timestampUs,\n duration: spec.durationUs,\n data: spec.data,\n }),\n );\n ctx.onProgress?.((i + 1) / total);\n }\n await decoder.flush();\n throwIfAborted(ctx.signal);\n } catch (err) {\n throw asCodecError(err, 'decode');\n } finally {\n ctx.signal?.removeEventListener('abort', onAbort);\n decoder.close();\n }\n\n return acc.finish(sampleRate, numberOfChannels);\n}\n","/**\n * Container demux for the video stage: pull encoded video/audio tracks out of\n * mp4 / webm / mkv so they can be fed to WebCodecs decoders.\n *\n * - `demuxContainerVideo` extracts the video track (required) plus its audio\n * track (optional — `null` when the container has no audio or the audio codec\n * is not decodable here, e.g. Vorbis-in-webm → video-only output).\n * - `demuxContainerAudio` extracts JUST the audio track (for routing a\n * container's audio into the existing audio matrix, e.g. mp4/m4a AAC → wav).\n *\n * Descriptions come verbatim from the container per the design note §B:\n * mp4 avcC via `codecConfig.bytes`, mp4 AAC ASC via `decoderSpecificInfo`,\n * webm/mkv via `codecPrivate` (OpusHead / avcC / ASC). All sample/block access\n * is the containers' own zero-copy iterators.\n *\n * See docs/design-notes/transcode.md §B (demux capability audit) and §D (video).\n */\n\nimport {\n type AudioChunk as MkvAudioChunk,\n type MkvAudioTrack,\n type MkvFile,\n type VideoChunk as MkvVideoChunk,\n type MkvVideoTrack,\n iterateAudioChunks as iterateMkvAudioChunks,\n iterateVideoChunks as iterateMkvVideoChunks,\n parseMkv,\n} from '@catlabtech/webcvt-container-mkv';\nimport {\n type Mp4File,\n type Mp4Track,\n deriveCodecString,\n findAudioTrack,\n findVideoTrack,\n iterateAudioSamplesAuto,\n iterateVideoSamples,\n parseMp4,\n} from '@catlabtech/webcvt-container-mp4';\nimport {\n type AudioChunk as WebmAudioChunk,\n type WebmAudioTrack,\n type WebmFile,\n type VideoChunk as WebmVideoChunk,\n type WebmVideoTrack,\n iterateAudioChunks as iterateWebmAudioChunks,\n iterateVideoChunks as iterateWebmVideoChunks,\n parseWebm,\n} from '@catlabtech/webcvt-container-webm';\nimport type { EncodedChunkSpec } from './demux.ts';\nimport { TranscodeDemuxError } from './errors.ts';\nimport type { ContainerFamily, VideoSideCodec } from './matrix.ts';\n\nconst US_PER_SEC = 1_000_000;\n/** WebCodecs decodes/encodes Opus at 48 kHz internally regardless of container. */\nconst OPUS_RATE = 48_000;\n/** Fallback per-chunk duration when a single-chunk stream gives no delta. */\nconst NOMINAL_AUDIO_FRAME_US = 20_000; // 20 ms\n\n// ---------------------------------------------------------------------------\n// Public types\n// ---------------------------------------------------------------------------\n\n/** One encoded video access unit to hand to the VideoDecoder. */\nexport interface EncodedVideoChunkSpec {\n readonly type: 'key' | 'delta';\n readonly timestampUs: number;\n readonly durationUs: number;\n readonly data: Uint8Array;\n}\n\n/** Encoded audio track ready to feed an AudioDecoder (shape of DemuxResult's `encoded`). */\nexport interface EncodedAudioTrack {\n readonly config: AudioDecoderConfig;\n readonly chunks: EncodedChunkSpec[];\n readonly sampleRate: number;\n readonly numberOfChannels: number;\n}\n\n/** Encoded video track ready to feed a VideoDecoder. */\nexport interface EncodedVideoTrack {\n readonly config: VideoDecoderConfig;\n readonly chunks: EncodedVideoChunkSpec[];\n readonly width: number;\n readonly height: number;\n /** Estimated frame rate (fps) for the encoder config; 0 when underivable. */\n readonly frameRate: number;\n /** Coarse codec token for decoder probing. */\n readonly codec: VideoSideCodec;\n}\n\nexport interface ContainerVideoDemux {\n readonly video: EncodedVideoTrack;\n /** Null → no audio track, or the audio codec is not decodable here. */\n readonly audio: EncodedAudioTrack | null;\n}\n\n// ---------------------------------------------------------------------------\n// Public API\n// ---------------------------------------------------------------------------\n\n/** Demux the video track (+ optional audio) of a container. */\nexport function demuxContainerVideo(\n family: ContainerFamily,\n bytes: Uint8Array,\n): ContainerVideoDemux {\n return wrap(() => {\n switch (family) {\n case 'mp4': {\n const file = parseMp4(bytes);\n return { video: mp4Video(file), audio: mp4Audio(file) };\n }\n case 'webm': {\n const file = parseWebm(bytes);\n return { video: webmVideo(file), audio: webmAudio(file) };\n }\n case 'mkv': {\n const file = parseMkv(bytes);\n return { video: mkvVideo(file), audio: mkvAudio(file) };\n }\n }\n });\n}\n\n/** Demux JUST the audio track of a container (for the audio matrix). */\nexport function demuxContainerAudioTrack(\n family: ContainerFamily,\n bytes: Uint8Array,\n): EncodedAudioTrack {\n return wrap(() => {\n let audio: EncodedAudioTrack | null;\n switch (family) {\n case 'mp4':\n audio = mp4Audio(parseMp4(bytes));\n break;\n case 'webm':\n audio = webmAudio(parseWebm(bytes));\n break;\n case 'mkv':\n audio = mkvAudio(parseMkv(bytes));\n break;\n }\n if (!audio) {\n throw new TranscodeDemuxError(\n `${family} container has no audio track decodable here (only AAC/Opus audio is supported)`,\n );\n }\n return audio;\n });\n}\n\n// ---------------------------------------------------------------------------\n// mp4\n// ---------------------------------------------------------------------------\n\nfunction mp4Video(file: Mp4File): EncodedVideoTrack {\n const track = findVideoTrack(file);\n if (!track || track.sampleEntry.kind !== 'video') {\n throw new TranscodeDemuxError('mp4 has no video track');\n }\n const entry = track.sampleEntry.entry;\n const config: VideoDecoderConfig = {\n codec: entry.codecString,\n description: entry.codecConfig.bytes,\n codedWidth: entry.width,\n codedHeight: entry.height,\n };\n\n const chunks: EncodedVideoChunkSpec[] = [];\n for (const s of iterateVideoSamples(track, file.fileBytes)) {\n chunks.push({\n type: s.isKeyframe ? 'key' : 'delta',\n timestampUs: Math.round(s.presentationTimeUs),\n durationUs: Math.round(s.durationUs),\n data: s.data,\n });\n }\n if (chunks.length === 0) throw new TranscodeDemuxError('mp4 video track has no samples');\n\n return {\n config,\n chunks,\n width: entry.width,\n height: entry.height,\n frameRate: estimateFrameRate(chunks),\n codec: mp4VideoCodec(entry.format),\n };\n}\n\nfunction mp4Audio(file: Mp4File): EncodedAudioTrack | null {\n const track: Mp4Track | null = findAudioTrack(file);\n if (!track || track.sampleEntry.kind !== 'audio') return null;\n const entry = track.sampleEntry.entry;\n const codec = deriveCodecString(entry.objectTypeIndication, entry.decoderSpecificInfo);\n const numberOfChannels = Math.max(1, entry.channelCount);\n const sampleRate = entry.sampleRate;\n\n const chunks: EncodedChunkSpec[] = [];\n for (const s of iterateAudioSamplesAuto(file, track)) {\n chunks.push({\n type: 'key',\n timestampUs: Math.round(s.timestampUs),\n durationUs: Math.round(s.durationUs),\n data: s.data,\n });\n }\n if (chunks.length === 0) return null;\n\n return {\n config: { codec, description: entry.decoderSpecificInfo, sampleRate, numberOfChannels },\n chunks,\n sampleRate,\n numberOfChannels,\n };\n}\n\nfunction mp4VideoCodec(format: string): VideoSideCodec {\n if (format.startsWith('avc')) return 'h264';\n if (format.startsWith('hev') || format.startsWith('hvc')) return 'hevc';\n if (format === 'vp09') return 'vp9';\n if (format === 'av01') return 'av1';\n return 'h264';\n}\n\n// ---------------------------------------------------------------------------\n// webm\n// ---------------------------------------------------------------------------\n\nfunction webmVideo(file: WebmFile): EncodedVideoTrack {\n const track = file.tracks.find((t): t is WebmVideoTrack => t.trackType === 1);\n if (!track) throw new TranscodeDemuxError('webm has no video track');\n const { codec, config } = webmVideoConfig(track);\n\n const chunks = collectVideoChunks(iterateWebmVideoChunks(file, track.trackNumber));\n if (chunks.length === 0) throw new TranscodeDemuxError('webm video track has no blocks');\n\n return {\n config,\n chunks,\n width: track.pixelWidth,\n height: track.pixelHeight,\n frameRate: estimateFrameRate(chunks),\n codec,\n };\n}\n\nfunction webmVideoConfig(track: WebmVideoTrack): {\n codec: VideoSideCodec;\n config: VideoDecoderConfig;\n} {\n const base = { codedWidth: track.pixelWidth, codedHeight: track.pixelHeight };\n switch (track.codecId) {\n case 'V_VP8':\n return { codec: 'vp8', config: { codec: 'vp8', ...base } };\n case 'V_VP9':\n return { codec: 'vp9', config: { codec: 'vp09.00.10.08', ...base } };\n case 'V_AV01':\n return {\n codec: 'av1',\n config: {\n codec: 'av01.0.04M.08',\n ...(track.codecPrivate ? { description: track.codecPrivate } : {}),\n ...base,\n },\n };\n }\n}\n\nfunction webmAudio(file: WebmFile): EncodedAudioTrack | null {\n const track = file.tracks.find((t): t is WebmAudioTrack => t.trackType === 2);\n if (!track) return null;\n if (track.codecId !== 'A_OPUS') return null; // Vorbis-in-webm decode deferred.\n const numberOfChannels = Math.max(1, track.channels);\n const chunks = deriveAudioDurations(\n collectAudioChunks(iterateWebmAudioChunks(file, track.trackNumber)),\n );\n if (chunks.length === 0) return null;\n\n return {\n config: {\n codec: 'opus',\n description: track.codecPrivate,\n sampleRate: OPUS_RATE,\n numberOfChannels,\n },\n chunks,\n sampleRate: OPUS_RATE,\n numberOfChannels,\n };\n}\n\n// ---------------------------------------------------------------------------\n// mkv\n// ---------------------------------------------------------------------------\n\nfunction mkvVideo(file: MkvFile): EncodedVideoTrack {\n const track = file.tracks.find((t): t is MkvVideoTrack => t.trackType === 1);\n if (!track) throw new TranscodeDemuxError('mkv has no video track');\n const config: VideoDecoderConfig = {\n codec: track.webcodecsCodecString,\n ...(mkvNeedsDescription(track.codecId) && track.codecPrivate\n ? { description: track.codecPrivate }\n : {}),\n codedWidth: track.pixelWidth,\n codedHeight: track.pixelHeight,\n };\n\n const chunks = collectVideoChunks(iterateMkvVideoChunks(file, track.trackNumber));\n if (chunks.length === 0) throw new TranscodeDemuxError('mkv video track has no blocks');\n\n return {\n config,\n chunks,\n width: track.pixelWidth,\n height: track.pixelHeight,\n frameRate: estimateFrameRate(chunks),\n codec: mkvVideoCodec(track.codecId),\n };\n}\n\nfunction mkvAudio(file: MkvFile): EncodedAudioTrack | null {\n const track = file.tracks.find((t): t is MkvAudioTrack => t.trackType === 2);\n if (!track) return null;\n\n let config: AudioDecoderConfig;\n let sampleRate: number;\n const numberOfChannels = Math.max(1, track.channels);\n if (track.codecId === 'A_OPUS') {\n sampleRate = OPUS_RATE;\n config = { codec: 'opus', description: track.codecPrivate, sampleRate, numberOfChannels };\n } else if (track.codecId === 'A_AAC') {\n sampleRate = track.samplingFrequency;\n config = {\n codec: track.webcodecsCodecString,\n description: track.codecPrivate,\n sampleRate,\n numberOfChannels,\n };\n } else {\n return null; // Vorbis / MP3 / FLAC-in-mkv decode deferred.\n }\n\n const chunks = deriveAudioDurations(\n collectAudioChunks(iterateMkvAudioChunks(file, track.trackNumber)),\n );\n if (chunks.length === 0) return null;\n\n return { config, chunks, sampleRate, numberOfChannels };\n}\n\nfunction mkvNeedsDescription(codecId: MkvVideoTrack['codecId']): boolean {\n return codecId === 'V_MPEG4/ISO/AVC' || codecId === 'V_MPEGH/ISO/HEVC' || codecId === 'V_AV01';\n}\n\nfunction mkvVideoCodec(codecId: MkvVideoTrack['codecId']): VideoSideCodec {\n switch (codecId) {\n case 'V_MPEG4/ISO/AVC':\n return 'h264';\n case 'V_MPEGH/ISO/HEVC':\n return 'hevc';\n case 'V_VP8':\n return 'vp8';\n case 'V_VP9':\n return 'vp9';\n case 'V_AV01':\n return 'av1';\n }\n}\n\n// ---------------------------------------------------------------------------\n// Shared helpers\n// ---------------------------------------------------------------------------\n\nfunction collectVideoChunks(\n gen: Generator<WebmVideoChunk | MkvVideoChunk>,\n): EncodedVideoChunkSpec[] {\n const raw = [...gen];\n const specs: EncodedVideoChunkSpec[] = [];\n for (let i = 0; i < raw.length; i++) {\n const c = raw[i];\n if (!c) continue;\n const next = raw[i + 1];\n const durationUs = next\n ? Math.max(1, next.timestampUs - c.timestampUs)\n : (specs.at(-1)?.durationUs ?? NOMINAL_AUDIO_FRAME_US);\n specs.push({ type: c.type, timestampUs: c.timestampUs, durationUs, data: c.data });\n }\n return specs;\n}\n\nfunction collectAudioChunks(gen: Generator<WebmAudioChunk | MkvAudioChunk>): EncodedChunkSpec[] {\n return [...gen].map((c) => ({\n type: 'key' as const,\n timestampUs: c.timestampUs,\n durationUs: 0, // filled by deriveAudioDurations\n data: c.data,\n }));\n}\n\n/** Fill each audio chunk's duration from the delta to the next chunk. */\nfunction deriveAudioDurations(chunks: EncodedChunkSpec[]): EncodedChunkSpec[] {\n return chunks.map((c, i) => {\n const next = chunks[i + 1];\n const durationUs = next\n ? Math.max(1, next.timestampUs - c.timestampUs)\n : chunks[i - 1]\n ? Math.max(1, c.timestampUs - (chunks[i - 1]?.timestampUs ?? 0))\n : NOMINAL_AUDIO_FRAME_US;\n return { ...c, durationUs };\n });\n}\n\nfunction estimateFrameRate(chunks: readonly EncodedVideoChunkSpec[]): number {\n if (chunks.length < 2) return 0;\n const first = chunks[0];\n const last = chunks[chunks.length - 1];\n if (!first || !last) return 0;\n const spanUs = last.timestampUs + last.durationUs - first.timestampUs;\n if (spanUs <= 0) return 0;\n return Math.round((chunks.length / spanUs) * US_PER_SEC);\n}\n\nfunction wrap<T>(run: () => T): T {\n try {\n return run();\n } catch (err) {\n if (err instanceof TranscodeDemuxError) throw err;\n throw new TranscodeDemuxError(err instanceof Error ? err.message : String(err), { cause: err });\n }\n}\n","/**\n * Demux: parse an input container into either raw PCM (wav) or a list of\n * encoded chunks + an `AudioDecoderConfig` for the AudioDecoder.\n *\n * All audio parsers here are eager (they return `frames[]`/`packets[]`), which\n * suits the buffer-all v1. mp3 and aac carry no explicit PTS, so timestamps are\n * derived from samples-per-frame; flac uses `sampleNumber`; opus uses granule.\n *\n * See docs/design-notes/transcode.md §B (demux capability audit).\n */\n\nimport { buildAudioSpecificConfig, parseAdts } from '@catlabtech/webcvt-container-aac';\nimport {\n BLOCK_TYPE_STREAMINFO,\n type FlacStreamInfo,\n STREAMINFO_SIZE,\n encodeBlockHeader,\n encodeStreamInfo,\n parseFlac,\n} from '@catlabtech/webcvt-container-flac';\nimport { parseMp3 } from '@catlabtech/webcvt-container-mp3';\nimport { parseOgg } from '@catlabtech/webcvt-container-ogg';\nimport { parseWav } from '@catlabtech/webcvt-container-wav';\nimport { TranscodeDemuxError } from './errors.ts';\nimport type { ContainerFamily, SideCodec } from './matrix.ts';\nimport { type DecodedAudio, wavToDecoded } from './pcm.ts';\nimport { demuxContainerAudioTrack } from './video-demux.ts';\n\n/** One encoded access unit to hand to the AudioDecoder. */\nexport interface EncodedChunkSpec {\n readonly type: 'key' | 'delta';\n readonly timestampUs: number;\n readonly durationUs: number;\n readonly data: Uint8Array;\n}\n\nexport type DemuxResult =\n | { readonly kind: 'pcm'; readonly decoded: DecodedAudio }\n | {\n readonly kind: 'encoded';\n readonly config: AudioDecoderConfig;\n readonly chunks: EncodedChunkSpec[];\n readonly sampleRate: number;\n readonly numberOfChannels: number;\n };\n\nconst US_PER_SEC = 1_000_000;\n\n/** Demux `bytes` for the given input codec. */\nexport function demuxAudio(inputCodec: SideCodec, bytes: Uint8Array): DemuxResult {\n try {\n switch (inputCodec) {\n case 'pcm':\n return { kind: 'pcm', decoded: wavToDecoded(parseWav(bytes)) };\n case 'mp3':\n return demuxMp3(bytes);\n case 'aac':\n return demuxAac(bytes);\n case 'flac':\n return demuxFlac(bytes);\n case 'opus':\n return demuxOpus(bytes);\n default:\n throw new TranscodeDemuxError(`no demuxer for input codec \"${inputCodec}\"`);\n }\n } catch (err) {\n if (err instanceof TranscodeDemuxError) throw err;\n throw new TranscodeDemuxError(err instanceof Error ? err.message : String(err), { cause: err });\n }\n}\n\n/**\n * Demux a container's (mp4/webm/mkv) audio track into encoded chunks + an\n * `AudioDecoderConfig`, so a container's audio can be routed into the existing\n * audio matrix (e.g. mp4/m4a AAC → wav, webm/mkv Opus → opus). The heavy\n * lifting lives in {@link demuxContainerAudioTrack}; this adapts it to the\n * shared {@link DemuxResult} shape the decode driver consumes.\n */\nexport function demuxContainerAudio(family: ContainerFamily, bytes: Uint8Array): DemuxResult {\n const track = demuxContainerAudioTrack(family, bytes);\n return {\n kind: 'encoded',\n config: track.config,\n chunks: track.chunks,\n sampleRate: track.sampleRate,\n numberOfChannels: track.numberOfChannels,\n };\n}\n\n// ---------------------------------------------------------------------------\n// mp3\n// ---------------------------------------------------------------------------\n\nfunction demuxMp3(bytes: Uint8Array): DemuxResult {\n const { frames } = parseMp3(bytes);\n if (frames.length === 0) throw new TranscodeDemuxError('mp3 has no audio frames');\n const first = frames[0]?.header;\n if (!first) throw new TranscodeDemuxError('mp3 has no audio frames');\n const sampleRate = first.sampleRate;\n const numberOfChannels = first.channelMode === 'mono' ? 1 : 2;\n\n const chunks: EncodedChunkSpec[] = [];\n let sample = 0;\n for (const frame of frames) {\n const spf = frame.header.samplesPerFrame;\n chunks.push({\n type: 'key',\n timestampUs: Math.round((sample / sampleRate) * US_PER_SEC),\n durationUs: Math.round((spf / sampleRate) * US_PER_SEC),\n data: frame.data,\n });\n sample += spf;\n }\n\n return {\n kind: 'encoded',\n config: { codec: 'mp3', sampleRate, numberOfChannels },\n chunks,\n sampleRate,\n numberOfChannels,\n };\n}\n\n// ---------------------------------------------------------------------------\n// aac (ADTS)\n// ---------------------------------------------------------------------------\n\nconst AAC_SAMPLES_PER_FRAME = 1024;\n\nfunction demuxAac(bytes: Uint8Array): DemuxResult {\n const { frames } = parseAdts(bytes);\n if (frames.length === 0) throw new TranscodeDemuxError('aac has no ADTS frames');\n const header = frames[0]?.header;\n if (!header) throw new TranscodeDemuxError('aac has no ADTS frames');\n const sampleRate = header.sampleRate;\n const numberOfChannels = Math.max(1, header.channelConfiguration);\n const description = buildAudioSpecificConfig(header);\n\n const chunks: EncodedChunkSpec[] = [];\n let sample = 0;\n for (const frame of frames) {\n const headerSize = frame.header.hasCrc ? 9 : 7;\n chunks.push({\n type: 'key',\n timestampUs: Math.round((sample / sampleRate) * US_PER_SEC),\n durationUs: Math.round((AAC_SAMPLES_PER_FRAME / sampleRate) * US_PER_SEC),\n data: frame.data.subarray(headerSize), // raw AU (strip ADTS header)\n });\n sample += AAC_SAMPLES_PER_FRAME;\n }\n\n return {\n kind: 'encoded',\n config: { codec: 'mp4a.40.2', sampleRate, numberOfChannels, description },\n chunks,\n sampleRate,\n numberOfChannels,\n };\n}\n\n// ---------------------------------------------------------------------------\n// flac\n// ---------------------------------------------------------------------------\n\nfunction demuxFlac(bytes: Uint8Array): DemuxResult {\n const { streamInfo, frames } = parseFlac(bytes);\n if (frames.length === 0) throw new TranscodeDemuxError('flac has no frames');\n const sampleRate = streamInfo.sampleRate;\n const numberOfChannels = streamInfo.channels;\n const description = buildFlacDescription(streamInfo);\n\n const chunks: EncodedChunkSpec[] = [];\n for (const frame of frames) {\n chunks.push({\n type: 'key',\n timestampUs: Math.round((frame.sampleNumber / sampleRate) * US_PER_SEC),\n durationUs: Math.round((frame.blockSize / sampleRate) * US_PER_SEC),\n data: frame.data,\n });\n }\n\n return {\n kind: 'encoded',\n config: { codec: 'flac', sampleRate, numberOfChannels, description },\n chunks,\n sampleRate,\n numberOfChannels,\n };\n}\n\n/**\n * WebCodecs FLAC decoder `description`: the `fLaC` signature followed by the\n * STREAMINFO metadata block (header + 34-byte body). Reuses container-flac's\n * own STREAMINFO encoder so the bytes match the parser exactly.\n */\nfunction buildFlacDescription(info: FlacStreamInfo): Uint8Array {\n const magic = new Uint8Array([0x66, 0x4c, 0x61, 0x43]); // \"fLaC\"\n const header = encodeBlockHeader(true, BLOCK_TYPE_STREAMINFO, STREAMINFO_SIZE);\n const body = encodeStreamInfo(info);\n const out = new Uint8Array(magic.length + header.length + body.length);\n out.set(magic, 0);\n out.set(header, magic.length);\n out.set(body, magic.length + header.length);\n return out;\n}\n\n// ---------------------------------------------------------------------------\n// opus-in-ogg\n// ---------------------------------------------------------------------------\n\nconst OPUS_RATE = 48_000;\nconst OPUS_NOMINAL_FRAME = 960; // 20 ms @ 48 kHz — fallback when granule is absent.\n\nfunction demuxOpus(bytes: Uint8Array): DemuxResult {\n const { streams } = parseOgg(bytes);\n const stream = streams[0];\n if (!stream) throw new TranscodeDemuxError('ogg has no logical stream');\n if (stream.codec !== 'opus') {\n throw new TranscodeDemuxError(\n `ogg stream codec \"${stream.codec}\" is not decodable here (only opus-in-ogg is supported)`,\n );\n }\n const numberOfChannels = Math.max(1, stream.channels);\n\n const chunks: EncodedChunkSpec[] = [];\n let sample = 0;\n for (const pkt of stream.packets) {\n // Prefer granule (48 kHz samples at packet end) to derive duration; else\n // fall back to a nominal 20 ms frame.\n let end = sample + OPUS_NOMINAL_FRAME;\n if (pkt.granulePosition >= 0n) {\n const g = Number(pkt.granulePosition);\n if (g > sample) end = g;\n }\n const durationSamples = Math.max(1, end - sample);\n chunks.push({\n type: 'key',\n timestampUs: Math.round((sample / OPUS_RATE) * US_PER_SEC),\n durationUs: Math.round((durationSamples / OPUS_RATE) * US_PER_SEC),\n data: pkt.data,\n });\n sample = end;\n }\n\n return {\n kind: 'encoded',\n config: {\n codec: 'opus',\n sampleRate: OPUS_RATE,\n numberOfChannels,\n description: stream.identification,\n },\n chunks,\n sampleRate: OPUS_RATE,\n numberOfChannels,\n };\n}\n","/**\n * Encode sinks: consume planar-float {@link DecodedAudio} and produce the\n * output container bytes.\n *\n * - **wav** — interleave to int16 → `serializeWav`. No WebCodecs.\n * - **opus-in-ogg / opus-in-webm** — `AudioEncoder{codec:'opus'}` → collect\n * `EncodedAudioChunk`s → build an `OggFile` / `WebmFile` (OpusHead as the\n * Ogg identification packet / WebM `CodecPrivate`) → serialize.\n * - **aac (ADTS)** — `AudioEncoder{codec:'mp4a.40.2', aac:{format:'adts'}}`;\n * the encoder emits ADTS-framed chunks, so concatenation is the file.\n * - **flac** — `AudioEncoder{codec:'flac'}`; the FLAC stream header arrives as\n * the first chunk's `decoderConfig.description`, frames as chunk data\n * (probe-gated, best-effort — see notes).\n *\n * Encoders take ownership of each `AudioData` and always `close()` it; the\n * encoder is `close()`-d in `finally` and via an abort listener.\n */\n\nimport { WebCodecsAudioEncoder } from '@catlabtech/webcvt-codec-webcodecs';\nimport {\n type OggFile,\n buildOpusHead,\n buildOpusTags,\n serializeOgg,\n} from '@catlabtech/webcvt-container-ogg';\nimport { type WavFile, serializeWav } from '@catlabtech/webcvt-container-wav';\nimport {\n type WebmAudioTrack,\n type WebmCluster,\n type WebmFile,\n serializeWebm,\n} from '@catlabtech/webcvt-container-webm';\nimport { asCodecError, throwIfAborted } from './abort.ts';\nimport { TranscodeCodecError } from './errors.ts';\nimport { type DecodedAudio, buildAudioData, interleaveInt16 } from './pcm.ts';\n\nexport interface EncodeContext {\n readonly signal?: AbortSignal;\n readonly onProgress?: (fraction: number) => void;\n}\n\nexport interface EncodeOptions {\n /** Encoder bitrate in bits/s (opus/aac). Omitted for lossless. */\n readonly bitrate?: number;\n}\n\n// ---------------------------------------------------------------------------\n// WAV sink (pure PCM — no WebCodecs)\n// ---------------------------------------------------------------------------\n\n/** Interleave to 16-bit PCM and serialize a canonical RIFF/WAV file. */\nexport function encodeWav(decoded: DecodedAudio): Uint8Array {\n const audioData = interleaveInt16(decoded);\n const channels = Math.max(1, decoded.numberOfChannels);\n const bitsPerSample = 16 as const;\n const blockAlign = (channels * bitsPerSample) / 8;\n const file: WavFile = {\n format: {\n audioFormat: 1,\n channels,\n sampleRate: decoded.sampleRate,\n bitsPerSample,\n blockAlign,\n byteRate: decoded.sampleRate * blockAlign,\n },\n audioData,\n };\n return serializeWav(file);\n}\n\n// ---------------------------------------------------------------------------\n// Shared AudioEncoder driver\n// ---------------------------------------------------------------------------\n\nexport interface EncodedOut {\n readonly data: Uint8Array;\n readonly timestampUs: number;\n readonly durationUs: number;\n}\n\ninterface EncodeChunksResult {\n readonly chunks: EncodedOut[];\n /** First chunk's `decoderConfig.description`, when the encoder supplies one. */\n readonly description?: Uint8Array;\n}\n\n/**\n * Feed `decoded` to an `AudioEncoder` in ~20 ms windows and collect the emitted\n * chunks (bytes + timing) plus the setup description from the first chunk.\n */\nasync function encodeAudioChunks(\n decoded: DecodedAudio,\n config: AudioEncoderConfig,\n ctx: EncodeContext,\n): Promise<EncodeChunksResult> {\n throwIfAborted(ctx.signal);\n\n const out: EncodedOut[] = [];\n let description: Uint8Array | undefined;\n\n const encoder = new WebCodecsAudioEncoder({ config }, (chunk, metadata) => {\n const bytes = new Uint8Array(chunk.byteLength);\n chunk.copyTo(bytes);\n out.push({\n data: bytes,\n timestampUs: chunk.timestamp,\n durationUs: chunk.duration ?? 0,\n });\n const desc = metadata?.decoderConfig?.description;\n if (description === undefined && desc !== undefined) {\n description = new Uint8Array(\n desc instanceof ArrayBuffer ? desc : (desc as ArrayBufferView).buffer,\n desc instanceof ArrayBuffer ? 0 : (desc as ArrayBufferView).byteOffset,\n desc instanceof ArrayBuffer ? desc.byteLength : (desc as ArrayBufferView).byteLength,\n ).slice();\n }\n });\n\n const onAbort = (): void => encoder.close();\n ctx.signal?.addEventListener('abort', onAbort, { once: true });\n\n try {\n const framesPerChunk = Math.max(1, Math.round(decoded.sampleRate / 50)); // 20 ms\n const total = decoded.numberOfFrames || 1;\n for (let start = 0; start < decoded.numberOfFrames; start += framesPerChunk) {\n throwIfAborted(ctx.signal);\n const count = Math.min(framesPerChunk, decoded.numberOfFrames - start);\n const timestampUs = Math.round((start / decoded.sampleRate) * 1_000_000);\n encoder.encode(buildAudioData(decoded, start, count, timestampUs));\n ctx.onProgress?.(Math.min(1, (start + count) / total));\n }\n await encoder.flush();\n throwIfAborted(ctx.signal);\n } catch (err) {\n throw asCodecError(err, 'encode');\n } finally {\n ctx.signal?.removeEventListener('abort', onAbort);\n encoder.close();\n }\n\n return { chunks: out, description };\n}\n\n// ---------------------------------------------------------------------------\n// Opus encode + mux\n// ---------------------------------------------------------------------------\n\nexport const OPUS_RATE = 48_000;\nexport const OPUS_PRE_SKIP = 3840; // libopus default pre-skip @48 kHz.\nconst OGG_SERIAL = 1;\nconst WEBM_TIMECODE_SCALE = 1_000_000; // 1 ms ticks.\nconst WEBM_CLUSTER_MAX_TICKS = 30_000; // keep per-block int16 delta well within range.\n\n/**\n * Encode planar-float PCM to Opus chunks (bytes + timing). Exported so the video\n * pipeline can reuse the exact audio-encode path for a webm/mkv Opus track.\n */\nexport function encodeOpusChunks(\n decoded: DecodedAudio,\n opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<EncodedOut[]> {\n return encodeOpus(decoded, opts, ctx);\n}\n\nasync function encodeOpus(\n decoded: DecodedAudio,\n opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<EncodedOut[]> {\n const config: AudioEncoderConfig = {\n codec: 'opus',\n sampleRate: decoded.sampleRate,\n numberOfChannels: Math.max(1, decoded.numberOfChannels),\n ...(opts.bitrate ? { bitrate: opts.bitrate } : {}),\n };\n const { chunks } = await encodeAudioChunks(decoded, config, ctx);\n return chunks;\n}\n\n/** any decodable audio → opus-in-ogg. */\nexport async function encodeOpusOgg(\n decoded: DecodedAudio,\n opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<Uint8Array> {\n const channels = Math.max(1, decoded.numberOfChannels);\n const chunks = await encodeOpus(decoded, opts, ctx);\n const identification = buildOpusHead({\n channelCount: channels,\n preSkip: OPUS_PRE_SKIP,\n inputSampleRate: decoded.sampleRate,\n });\n const comments = buildOpusTags('webcvt-transcode', [\n { key: 'ENCODER', value: 'webcvt-transcode' },\n ]);\n\n const packets = chunks.map((c) => ({\n data: c.data,\n // Opus granule = decoded 48 kHz samples (incl. pre-skip) at packet end.\n granulePosition: BigInt(\n Math.round(((c.timestampUs + c.durationUs) / 1_000_000) * OPUS_RATE) + OPUS_PRE_SKIP,\n ),\n serialNumber: OGG_SERIAL,\n }));\n\n const file: OggFile = {\n streams: [\n {\n serialNumber: OGG_SERIAL,\n codec: 'opus',\n identification,\n comments,\n setup: undefined,\n packets,\n preSkip: OPUS_PRE_SKIP,\n sampleRate: OPUS_RATE,\n channels,\n },\n ],\n };\n return serializeOgg(file);\n}\n\n/** any decodable audio → opus-in-webm. */\nexport async function encodeOpusWebm(\n decoded: DecodedAudio,\n opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<Uint8Array> {\n const channels = Math.max(1, decoded.numberOfChannels);\n const chunks = await encodeOpus(decoded, opts, ctx);\n const codecPrivate = buildOpusHead({\n channelCount: channels,\n preSkip: OPUS_PRE_SKIP,\n inputSampleRate: decoded.sampleRate,\n });\n\n // Split clusters so every SimpleBlock's timecode delta (ticks) fits int16.\n const clusters: WebmCluster[] = [];\n let current: WebmCluster | null = null;\n for (const c of chunks) {\n const tick = BigInt(Math.round(c.timestampUs / 1000)); // µs → ms ticks\n if (current === null || tick - current.timecode > BigInt(WEBM_CLUSTER_MAX_TICKS)) {\n current = { fileOffset: 0, timecode: tick, blocks: [] };\n clusters.push(current);\n }\n current.blocks.push({\n trackNumber: 1,\n timestampNs: tick * BigInt(WEBM_TIMECODE_SCALE),\n keyframe: true,\n invisible: false,\n discardable: false,\n frames: [c.data],\n });\n }\n\n const track: WebmAudioTrack = {\n trackNumber: 1,\n trackUid: 1n,\n trackType: 2,\n codecId: 'A_OPUS',\n codecPrivate,\n samplingFrequency: OPUS_RATE,\n channels,\n };\n\n const file: WebmFile = {\n ebmlHeader: {\n ebmlVersion: 1,\n ebmlReadVersion: 1,\n ebmlMaxIdLength: 4,\n ebmlMaxSizeLength: 8,\n docType: 'webm',\n docTypeVersion: 4,\n docTypeReadVersion: 2,\n },\n segmentPayloadOffset: 0, // placeholder — ignored on write.\n info: {\n timecodeScale: WEBM_TIMECODE_SCALE,\n muxingApp: 'webcvt-transcode',\n writingApp: 'webcvt-transcode',\n },\n tracks: [track],\n clusters,\n fileBytes: new Uint8Array(0), // placeholder — ignored on write.\n };\n return serializeWebm(file);\n}\n\n// ---------------------------------------------------------------------------\n// AAC (encoder-native ADTS)\n// ---------------------------------------------------------------------------\n\n/** any decodable audio → aac (ADTS). Encoder emits ADTS frames; concat = file. */\nexport async function encodeAac(\n decoded: DecodedAudio,\n opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<Uint8Array> {\n const config = {\n codec: 'mp4a.40.2',\n sampleRate: decoded.sampleRate,\n numberOfChannels: Math.max(1, decoded.numberOfChannels),\n ...(opts.bitrate ? { bitrate: opts.bitrate } : {}),\n // Ask the encoder for self-framed ADTS output so concatenation is a valid\n // .aac file (no container muxer needed).\n aac: { format: 'adts' as const },\n } as AudioEncoderConfig;\n\n const { chunks } = await encodeAudioChunks(decoded, config, ctx);\n return concatBytes(chunks.map((c) => c.data));\n}\n\n// ---------------------------------------------------------------------------\n// FLAC (probe-gated, best-effort)\n// ---------------------------------------------------------------------------\n\n/**\n * any decodable audio → flac. The WebCodecs FLAC encoder delivers the stream\n * header (`fLaC` + STREAMINFO) via the first chunk's `decoderConfig.description`\n * and the FLAC frames as chunk data, so the file is `description ++ frames`.\n *\n * This path is gated by the encoder probe in `canHandle`; if FLAC encode is\n * unsupported (most non-Chromium runtimes) routing falls through to ffmpeg-wasm\n * and this is never reached. The exact chunk→frame mapping is unverified in\n * Node (no real encoder) — see docs/design-notes/transcode.md §Needs verification.\n */\nexport async function encodeFlac(\n decoded: DecodedAudio,\n _opts: EncodeOptions,\n ctx: EncodeContext,\n): Promise<Uint8Array> {\n const config: AudioEncoderConfig = {\n codec: 'flac',\n sampleRate: decoded.sampleRate,\n numberOfChannels: Math.max(1, decoded.numberOfChannels),\n };\n const { chunks, description } = await encodeAudioChunks(decoded, config, ctx);\n if (description === undefined) {\n throw new TranscodeCodecError(\n 'flac encoder did not supply a stream header (decoderConfig.description); cannot mux FLAC',\n );\n }\n return concatBytes([description, ...chunks.map((c) => c.data)]);\n}\n\n// ---------------------------------------------------------------------------\n// Helpers\n// ---------------------------------------------------------------------------\n\nfunction concatBytes(parts: Uint8Array[]): Uint8Array {\n const total = parts.reduce((s, p) => s + p.length, 0);\n const out = new Uint8Array(total);\n let off = 0;\n for (const p of parts) {\n out.set(p, off);\n off += p.length;\n }\n return out;\n}\n","/**\n * Static v1 capability matrix + codec/bitrate resolution for the audio\n * transcode paths.\n *\n * `canHandle` is two-stage: this frozen matrix is the cheap O(1) first gate\n * (rejects off-matrix pairs like `→ mp3`/`→ mp4` and every video pair before\n * any `isConfigSupported` round-trip), and the concrete both-sided codec probe\n * is the second gate (see backend.ts). See docs/design-notes/transcode.md §D.\n */\n\n/** Codec token for a demux/mux side. `'pcm'` = raw PCM (wav), no WebCodecs. */\nexport type SideCodec = 'pcm' | 'opus' | 'aac' | 'flac' | 'mp3';\n\n/** Output container family we can mux from scratch. */\nexport type OutputContainer = 'wav' | 'ogg' | 'webm' | 'aac' | 'flac';\n\nexport interface OutputTarget {\n readonly container: OutputContainer;\n /** Codec the WebCodecs encoder must produce (`'pcm'` = interleave, no encode). */\n readonly codec: SideCodec;\n}\n\n// ---------------------------------------------------------------------------\n// Decodable audio inputs → the codec fed to the AudioDecoder (or 'pcm').\n// ---------------------------------------------------------------------------\n\n/**\n * MIME → input codec for the containers this stage can demux. `audio/ogg` is\n * assumed to carry Opus (Vorbis-in-Ogg decode needs a multi-packet setup\n * description and is deferred). mp4/m4a/webm/mkv audio-track extraction is\n * deferred to the video stage (shared demuxers).\n */\nexport const INPUT_CODECS: Readonly<Record<string, SideCodec>> = Object.freeze({\n 'audio/wav': 'pcm',\n 'audio/x-wav': 'pcm',\n 'audio/wave': 'pcm',\n 'audio/mpeg': 'mp3',\n 'audio/mp3': 'mp3',\n 'audio/x-mpeg': 'mp3',\n 'audio/aac': 'aac',\n 'audio/x-aac': 'aac',\n 'audio/ogg': 'opus',\n 'audio/opus': 'opus',\n 'audio/flac': 'flac',\n 'audio/x-flac': 'flac',\n});\n\n// ---------------------------------------------------------------------------\n// Muxable audio outputs → container + encoder codec.\n// ---------------------------------------------------------------------------\n\n/**\n * MIME → output target. `→ mp3` and `→ ogg(vorbis)` are intentionally absent\n * (no WebCodecs encoder); `→ mp4/m4a` is absent (no from-scratch mp4 muxer yet).\n */\nexport const OUTPUT_TARGETS: Readonly<Record<string, OutputTarget>> = Object.freeze({\n 'audio/wav': { container: 'wav', codec: 'pcm' },\n 'audio/x-wav': { container: 'wav', codec: 'pcm' },\n 'audio/wave': { container: 'wav', codec: 'pcm' },\n 'audio/ogg': { container: 'ogg', codec: 'opus' },\n 'audio/opus': { container: 'ogg', codec: 'opus' },\n 'audio/webm': { container: 'webm', codec: 'opus' },\n 'audio/aac': { container: 'aac', codec: 'aac' },\n 'audio/flac': { container: 'flac', codec: 'flac' },\n});\n\n/**\n * Frozen set of `\"${inputMime}|${outputMime}\"` keys — the full product of every\n * decodable input against every muxable output. This is the static gate.\n */\nexport const TRANSCODE_MATRIX: ReadonlySet<string> = Object.freeze(\n new Set<string>(\n Object.keys(INPUT_CODECS).flatMap((inMime) =>\n Object.keys(OUTPUT_TARGETS).map((outMime) => `${inMime}|${outMime}`),\n ),\n ),\n);\n\n/** Build the matrix key for a pair. */\nexport function matrixKey(inputMime: string, outputMime: string): string {\n return `${inputMime}|${outputMime}`;\n}\n\n/** Lookup the input codec for a MIME, or `undefined` if not decodable here. */\nexport function inputCodecFor(mime: string): SideCodec | undefined {\n return INPUT_CODECS[mime];\n}\n\n/** Lookup the output target for a MIME, or `undefined` if not muxable here. */\nexport function outputTargetFor(mime: string): OutputTarget | undefined {\n return OUTPUT_TARGETS[mime];\n}\n\n// ---------------------------------------------------------------------------\n// Video + container-track matrix (video stage).\n//\n// These are DELIBERATELY separate from INPUT_CODECS / OUTPUT_TARGETS /\n// TRANSCODE_MATRIX above (which stay the frozen audio-only matrix). A container\n// input (mp4/webm/mkv) is not a single audio codec — it may carry a video and\n// an audio track — so it is routed through its own lookups. `canHandle`/`convert`\n// consult BOTH the audio matrix and these container lookups.\n// ---------------------------------------------------------------------------\n\n/** Demuxable container family (its sample/block iterators feed the pipeline). */\nexport type ContainerFamily = 'mp4' | 'webm' | 'mkv';\n\n/** WebCodecs video codec token for a demux/mux side. */\nexport type VideoSideCodec = 'h264' | 'vp9' | 'vp8' | 'av1' | 'hevc';\n\n/**\n * Container input MIME → family. Covers the video containers (video/mp4,\n * video/webm, video/x-matroska) and the audio-only projections that share a\n * demuxer (audio/mp4 = m4a, audio/webm). A container input can be transcoded\n * to a video target (its video track) OR have its audio track routed into the\n * existing audio matrix. Kept out of INPUT_CODECS so `inputCodecFor('video/mp4')`\n * stays `undefined` (a container is not one audio codec).\n */\nexport const CONTAINER_INPUTS: Readonly<Record<string, ContainerFamily>> = Object.freeze({\n 'video/mp4': 'mp4',\n 'audio/mp4': 'mp4', // m4a — audio-only track extraction\n 'video/webm': 'webm',\n 'audio/webm': 'webm', // webm audio-only extraction\n 'video/x-matroska': 'mkv',\n 'video/mkv': 'mkv',\n});\n\n/** Output container family for a video target (VP9 default, VP8 fallback). */\nexport interface VideoTarget {\n readonly container: 'webm' | 'mkv';\n}\n\n/**\n * Video output MIME → container. `→ mp4` is intentionally absent (no from-scratch\n * mp4 muxer yet — deferred to a later stage). The encoder codec (VP9 preferred,\n * VP8 fallback) is resolved at convert time by probing, not fixed here.\n */\nexport const VIDEO_TARGETS: Readonly<Record<string, VideoTarget>> = Object.freeze({\n 'video/webm': { container: 'webm' },\n 'video/x-matroska': { container: 'mkv' },\n 'video/mkv': { container: 'mkv' },\n});\n\n/**\n * Assumed audio-track codec per container family, used only by `canHandle`\n * probing (the true codec is known after demux). mp4 audio is ~always AAC and\n * webm/mkv audio is ~always Opus. A mismatch (e.g. Vorbis-in-webm) is rejected\n * at demux — the same optimistic-probe contract the ogg(opus) input already uses.\n */\nexport const CONTAINER_AUDIO_CODEC: Readonly<Record<ContainerFamily, Exclude<SideCodec, 'pcm'>>> =\n Object.freeze({ mp4: 'aac', webm: 'opus', mkv: 'opus' });\n\n/** Assumed video-track codec per family, for `canHandle` decoder probing. */\nexport const CONTAINER_VIDEO_CODEC: Readonly<Record<ContainerFamily, VideoSideCodec>> =\n Object.freeze({ mp4: 'h264', webm: 'vp9', mkv: 'h264' });\n\n/** Lookup the container family for an input MIME, or `undefined`. */\nexport function containerFamilyFor(mime: string): ContainerFamily | undefined {\n return CONTAINER_INPUTS[mime];\n}\n\n/** Lookup the video output target for a MIME, or `undefined`. */\nexport function videoTargetFor(mime: string): VideoTarget | undefined {\n return VIDEO_TARGETS[mime];\n}\n\n// ---------------------------------------------------------------------------\n// Options → encoder config mapping\n// ---------------------------------------------------------------------------\n\n/** Default quality when `ConvertOptions.quality` is omitted (mid-ladder). */\nexport const DEFAULT_QUALITY = 0.7;\n\n/**\n * Buffer-all input cap. Every serializer takes a complete in-memory model, so a\n * hard cap protects against OOM. 256 MiB per the design note's conservative\n * default; revisit when a streaming muxer exists (decoded PCM expands well\n * beyond the compressed input size).\n */\nexport const MAX_INPUT_BYTES = 256 * 1024 * 1024;\n\n/**\n * Map `quality` (0–1) to a codec bitrate in bits/s. The ladder is piecewise so\n * the default `quality = 0.7` lands exactly on the mid-ladder figure from the\n * design note (Opus 128 kbps, AAC 128 kbps stereo), with q=0 and q=1 at the\n * ladder ends. Mono is scaled to ~0.6×. Lossless FLAC returns `undefined`\n * (bitrate is not applicable).\n */\nexport function resolveBitrate(\n codec: SideCodec,\n quality: number,\n channels: number,\n): number | undefined {\n const q = clamp01(quality);\n let stereo: number;\n switch (codec) {\n case 'opus':\n // 64 → 128 → 256 kbps across q 0 → 0.7 → 1.\n stereo = ladder(q, 64_000, 128_000, 256_000);\n break;\n case 'aac':\n // 96 → 128 → 256 kbps across q 0 → 0.7 → 1.\n stereo = ladder(q, 96_000, 128_000, 256_000);\n break;\n default:\n return undefined; // pcm / flac: no bitrate.\n }\n const scale = channels <= 1 ? 0.6 : 1;\n return Math.round(stereo * scale);\n}\n\n/**\n * Map video geometry + `quality` to a VP9/VP8 bitrate in bits/s. Resolution\n * drives the base (design note ladder: 720p ≈ 3 Mbps for VP9), scaled ±40% by\n * quality around the default `0.7`, with VP8 at ≈1.3× VP9. Clamped to a sane\n * floor so tiny test clips still get a positive bitrate.\n */\nexport function resolveVideoBitrate(\n width: number,\n height: number,\n quality: number,\n codec: 'vp9' | 'vp8',\n): number {\n const q = clamp01(quality);\n const pixels = Math.max(1, width) * Math.max(1, height);\n const ratio = pixels / (1280 * 720);\n const base = VP9_720P_BPS * ratio;\n const scale =\n q <= DEFAULT_QUALITY\n ? 0.6 + (0.4 * q) / DEFAULT_QUALITY // q 0 → 0.6, q 0.7 → 1.0\n : 1.0 + (0.4 * (q - DEFAULT_QUALITY)) / (1 - DEFAULT_QUALITY); // q 1 → 1.4\n const codecFactor = codec === 'vp8' ? 1.3 : 1;\n return Math.max(VIDEO_MIN_BPS, Math.round(base * scale * codecFactor));\n}\n\n// ---------------------------------------------------------------------------\n// Internal helpers\n// ---------------------------------------------------------------------------\n\n/** VP9 base bitrate at 720p, quality 0.7. */\nconst VP9_720P_BPS = 3_000_000;\n/** Floor so tiny clips still get a positive, encodable bitrate. */\nconst VIDEO_MIN_BPS = 100_000;\n\nfunction clamp01(v: number): number {\n if (Number.isNaN(v)) return DEFAULT_QUALITY;\n return Math.max(0, Math.min(1, v));\n}\n\n/**\n * Piecewise-linear ladder through (0, lo), (0.7, mid), (1, hi). The knee at 0.7\n * makes the default quality resolve to `mid` exactly.\n */\nfunction ladder(q: number, lo: number, mid: number, hi: number): number {\n const KNEE = DEFAULT_QUALITY;\n if (q <= KNEE) return lo + ((mid - lo) * q) / KNEE;\n return mid + ((hi - mid) * (q - KNEE)) / (1 - KNEE);\n}\n","/**\n * Per-session memoisation of WebCodecs `isConfigSupported` probes.\n *\n * `canHandle` runs for every registry lookup, so an uncached backend would fire\n * an `isConfigSupported` round-trip on each call. Codec support does not change\n * within a session, so results are cached forever (keyed by codec token +\n * direction). A `WebCodecsNotSupportedError` — the signal for a runtime with no\n * WebCodecs (Node, Safari < 26 for audio) — is caught and cached as `false`, so\n * the backend falls through to ffmpeg-wasm cleanly instead of throwing.\n */\n\nimport {\n type AudioCodecName,\n type ProbeResult,\n type VideoCodecName,\n WebCodecsNotSupportedError,\n probeAudioCodec,\n probeAudioDecoder,\n probeVideoCodec,\n probeVideoDecoder,\n} from '@catlabtech/webcvt-codec-webcodecs';\n\nexport interface ProbeHints {\n readonly sampleRate?: number;\n readonly numberOfChannels?: number;\n}\n\nexport interface VideoProbeHints {\n readonly width?: number;\n readonly height?: number;\n readonly bitrate?: number;\n readonly framerate?: number;\n readonly hardwareAcceleration?: HardwareAcceleration;\n}\n\n/**\n * Caches audio decode/encode capability probes for one backend instance\n * (== one session). Not shared across instances so tests stay isolated.\n */\nexport class ProbeCache {\n readonly #decode = new Map<string, boolean>();\n readonly #encode = new Map<string, boolean>();\n readonly #videoDecode = new Map<string, boolean>();\n readonly #videoEncode = new Map<string, ProbeResult | null>();\n\n /** True iff the current runtime can DECODE `codec`. Memoised. */\n async canDecode(codec: AudioCodecName, hints: ProbeHints = {}): Promise<boolean> {\n const cached = this.#decode.get(codec);\n if (cached !== undefined) return cached;\n const ok = await probeSafely(() => probeAudioDecoder({ codec, ...hints }));\n this.#decode.set(codec, ok);\n return ok;\n }\n\n /** True iff the current runtime can ENCODE `codec` (probeAudioCodec). Memoised. */\n async canEncode(codec: AudioCodecName, hints: ProbeHints = {}): Promise<boolean> {\n const cached = this.#encode.get(codec);\n if (cached !== undefined) return cached;\n const ok = await probeSafely(() => probeAudioCodec({ codec, ...hints }));\n this.#encode.set(codec, ok);\n return ok;\n }\n\n /** True iff the current runtime can DECODE video `codec`. Memoised. */\n async canDecodeVideo(codec: VideoCodecName): Promise<boolean> {\n const cached = this.#videoDecode.get(codec);\n if (cached !== undefined) return cached;\n const ok = await probeSafely(() => probeVideoDecoder({ codec }));\n this.#videoDecode.set(codec, ok);\n return ok;\n }\n\n /**\n * Full encode probe for a video `codec` at the given geometry — returns the\n * {@link ProbeResult} (with `hardwareAccelerated`) or `null` when unsupported\n * / no WebCodecs. Memoised by codec + geometry so `canHandle` (default dims)\n * and the pipeline (real dims) each cache their own lookup.\n */\n async probeEncodeVideo(\n codec: VideoCodecName,\n hints: VideoProbeHints = {},\n ): Promise<ProbeResult | null> {\n const key = `${codec}:${hints.width ?? 0}x${hints.height ?? 0}`;\n const cached = this.#videoEncode.get(key);\n if (cached !== undefined) return cached;\n let result: ProbeResult | null;\n try {\n const r = await probeVideoCodec({ codec, ...hints });\n result = r.supported ? r : null;\n } catch (err) {\n if (err instanceof WebCodecsNotSupportedError) result = null;\n else throw err;\n }\n this.#videoEncode.set(key, result);\n return result;\n }\n\n /** True iff the current runtime can ENCODE video `codec`. Memoised. */\n async canEncodeVideo(codec: VideoCodecName, hints: VideoProbeHints = {}): Promise<boolean> {\n return (await this.probeEncodeVideo(codec, hints)) !== null;\n }\n}\n\nasync function probeSafely(run: () => Promise<{ supported: boolean }>): Promise<boolean> {\n try {\n const result = await run();\n return result.supported;\n } catch (err) {\n if (err instanceof WebCodecsNotSupportedError) return false;\n throw err;\n }\n}\n","/**\n * Video transcode pipeline: mp4/webm/mkv → webm/mkv (VP9|VP8 + Opus).\n *\n * Flow: demux both tracks → decode H.264/VP8/VP9 video to `VideoFrame`s and feed\n * them straight into a VP9 (fallback VP8) `VideoEncoder`, closing every frame the\n * moment it is handed off (the wrapper's `finally` closes it — VideoFrames leak\n * GPU surfaces otherwise). If the source carries audio AND both the audio decoder\n * and the Opus encoder probe as supported, the audio track is decoded to PCM and\n * re-encoded to Opus (reusing the audio stage); otherwise a **video-only** webm/mkv\n * is emitted (Safari-style video-only WebCodecs) and surfaced via\n * {@link VideoTranscodeResult.audioIncluded}.\n *\n * The muxer is buffer-all: encoded chunks accumulate into a model, then one\n * `serializeWebm`/`serializeMkv` call. Clusters split on video keyframes and\n * whenever a per-block tick delta would approach the int16 SimpleBlock limit.\n *\n * See docs/design-notes/transcode.md §D (category 1, flagship) and §E.\n */\n\nimport { WebCodecsVideoDecoder, WebCodecsVideoEncoder } from '@catlabtech/webcvt-codec-webcodecs';\nimport {\n type MkvAudioTrack,\n type MkvCluster,\n type MkvFile,\n type MkvVideoTrack,\n serializeMkv,\n} from '@catlabtech/webcvt-container-mkv';\nimport { buildOpusHead } from '@catlabtech/webcvt-container-ogg';\nimport {\n type WebmAudioTrack,\n type WebmCluster,\n type WebmFile,\n type WebmVideoTrack,\n serializeWebm,\n} from '@catlabtech/webcvt-container-webm';\nimport { asCodecError, throwIfAborted } from './abort.ts';\nimport { decodeToPcm } from './decode.ts';\nimport { type EncodedOut, OPUS_PRE_SKIP, OPUS_RATE, encodeOpusChunks } from './encode.ts';\nimport { TranscodeCodecError } from './errors.ts';\nimport { type ContainerFamily, resolveVideoBitrate } from './matrix.ts';\nimport type { ProbeCache } from './probe-cache.ts';\nimport {\n type ContainerVideoDemux,\n type EncodedAudioTrack,\n demuxContainerVideo,\n} from './video-demux.ts';\n\n// ---------------------------------------------------------------------------\n// Public API\n// ---------------------------------------------------------------------------\n\nexport interface VideoTranscodeOptions {\n readonly quality: number;\n readonly signal?: AbortSignal;\n readonly hardwareAcceleration?: HardwareAcceleration;\n /** (percent 0–100, phase) — already monotone. */\n readonly onProgress?: (percent: number, phase: string) => void;\n}\n\nexport interface VideoTranscodeResult {\n readonly bytes: Uint8Array;\n readonly hardwareAccelerated: boolean;\n /** False → the source had audio we could not transcode; output is video-only. */\n readonly audioIncluded: boolean;\n}\n\nconst KEYFRAME_INTERVAL_US = 2_000_000; // request a keyframe at most every 2 s.\nconst WEBM_TIMECODE_SCALE = 1_000_000; // 1 ms ticks.\nconst CLUSTER_MAX_TICKS = 30_000; // keep every SimpleBlock delta well inside int16.\nconst VIDEO_TRACK = 1;\nconst AUDIO_TRACK = 2;\n\n/**\n * Transcode a container's video (and, when possible, audio) track to a fresh\n * webm/mkv with VP9|VP8 video + Opus audio.\n */\nexport async function transcodeVideo(\n family: ContainerFamily,\n outputContainer: 'webm' | 'mkv',\n bytes: Uint8Array,\n probes: ProbeCache,\n opts: VideoTranscodeOptions,\n): Promise<VideoTranscodeResult> {\n const { signal } = opts;\n const emit = monotone(opts.onProgress);\n\n emit(0, 'demux');\n const demux = demuxContainerVideo(family, bytes);\n emit(5, 'demux');\n throwIfAborted(signal);\n\n // Choose the video encoder: prefer VP9, fall back to VP8. Both are probed at\n // the real geometry so we also learn hardware acceleration.\n const { videoCodec, codecString, hardwareAccelerated } = await selectVideoEncoder(\n demux.video,\n probes,\n opts.hardwareAcceleration,\n );\n\n // Decide up front whether the audio track can ride along (probe-gated).\n const audioPlan = await planAudio(demux.audio, probes);\n const videoCap = audioPlan ? 60 : 90; // reserve the tail for audio when present.\n\n emit(5, 'decode');\n const videoOut = await encodeVideoTrack(demux, videoCodec, codecString, {\n signal,\n hardwareAcceleration: opts.hardwareAcceleration,\n quality: opts.quality,\n onEncoded: (fraction) => emit(5 + fraction * (videoCap - 5), 'encode'),\n });\n throwIfAborted(signal);\n\n // Audio sub-pipeline (decode → Opus), reusing the audio stage.\n let audioOut: EncodedOut[] | null = null;\n let audioChannels = 2;\n if (audioPlan) {\n emit(videoCap, 'decode');\n const decoded = await decodeToPcm(\n {\n kind: 'encoded',\n config: audioPlan.track.config,\n chunks: audioPlan.track.chunks,\n sampleRate: audioPlan.track.sampleRate,\n numberOfChannels: audioPlan.track.numberOfChannels,\n },\n { signal, onProgress: (f) => emit(videoCap + f * 10, 'decode') },\n );\n throwIfAborted(signal);\n audioChannels = Math.max(1, decoded.numberOfChannels);\n audioOut = await encodeOpusChunks(\n decoded,\n {},\n { signal, onProgress: (f) => emit(70 + f * 20, 'encode') },\n );\n throwIfAborted(signal);\n }\n\n emit(90, 'mux');\n const outBytes =\n outputContainer === 'mkv'\n ? buildMkv(\n demux.video.width,\n demux.video.height,\n videoCodec,\n videoOut,\n audioOut,\n audioChannels,\n )\n : buildWebm(\n demux.video.width,\n demux.video.height,\n videoCodec,\n videoOut,\n audioOut,\n audioChannels,\n );\n throwIfAborted(signal);\n\n emit(100, 'done');\n return { bytes: outBytes, hardwareAccelerated, audioIncluded: audioOut !== null };\n}\n\n// ---------------------------------------------------------------------------\n// Encoder selection\n// ---------------------------------------------------------------------------\n\nasync function selectVideoEncoder(\n video: ContainerVideoDemux['video'],\n probes: ProbeCache,\n hardwareAcceleration: HardwareAcceleration | undefined,\n): Promise<{ videoCodec: 'vp9' | 'vp8'; codecString: string; hardwareAccelerated: boolean }> {\n const hints = {\n width: video.width,\n height: video.height,\n ...(hardwareAcceleration ? { hardwareAcceleration } : {}),\n };\n const vp9 = await probes.probeEncodeVideo('vp9', hints);\n if (vp9) {\n return {\n videoCodec: 'vp9',\n codecString: 'vp09.00.10.08',\n hardwareAccelerated: vp9.hardwareAccelerated,\n };\n }\n const vp8 = await probes.probeEncodeVideo('vp8', hints);\n if (vp8) {\n return { videoCodec: 'vp8', codecString: 'vp8', hardwareAccelerated: vp8.hardwareAccelerated };\n }\n throw new TranscodeCodecError('no VP9/VP8 video encoder is supported in this runtime');\n}\n\ninterface AudioPlan {\n readonly track: EncodedAudioTrack;\n}\n\nasync function planAudio(\n audio: EncodedAudioTrack | null,\n probes: ProbeCache,\n): Promise<AudioPlan | null> {\n if (!audio) return null;\n const inputCodec = audioCodecName(audio.config.codec);\n if (!inputCodec) return null;\n if (!(await probes.canDecode(inputCodec))) return null;\n if (!(await probes.canEncode('opus'))) return null;\n return { track: audio };\n}\n\nfunction audioCodecName(codec: string): 'aac' | 'opus' | 'flac' | null {\n if (codec === 'opus') return 'opus';\n if (codec.startsWith('mp4a') || codec === 'aac') return 'aac';\n if (codec === 'flac') return 'flac';\n return null;\n}\n\n// ---------------------------------------------------------------------------\n// Video decode → encode\n// ---------------------------------------------------------------------------\n\ninterface VideoOut {\n readonly data: Uint8Array;\n readonly timestampUs: number;\n readonly keyframe: boolean;\n}\n\ninterface VideoEncodeCtx {\n readonly signal?: AbortSignal;\n readonly hardwareAcceleration?: HardwareAcceleration;\n readonly quality: number;\n readonly onEncoded: (fraction: number) => void;\n}\n\nasync function encodeVideoTrack(\n demux: ContainerVideoDemux,\n videoCodec: 'vp9' | 'vp8',\n codecString: string,\n ctx: VideoEncodeCtx,\n): Promise<VideoOut[]> {\n const { video } = demux;\n const { width, height } = video;\n const frameRate = video.frameRate > 0 ? video.frameRate : 30;\n const bitrate = resolveVideoBitrate(width, height, ctx.quality, videoCodec);\n const total = video.chunks.length || 1;\n\n const out: VideoOut[] = [];\n const encoder = new WebCodecsVideoEncoder(\n {\n config: {\n codec: codecString,\n width,\n height,\n bitrate,\n framerate: frameRate,\n ...(ctx.hardwareAcceleration ? { hardwareAcceleration: ctx.hardwareAcceleration } : {}),\n },\n },\n (chunk) => {\n const data = new Uint8Array(chunk.byteLength);\n chunk.copyTo(data);\n out.push({ data, timestampUs: chunk.timestamp, keyframe: chunk.type === 'key' });\n ctx.onEncoded(Math.min(1, out.length / total));\n },\n );\n\n let frameIndex = 0;\n let lastKeyframeUs = Number.NEGATIVE_INFINITY;\n const decoder = new WebCodecsVideoDecoder({ config: video.config }, (frame) => {\n // If an abort raced in (the signal listener closes the encoder mid-flush),\n // don't feed the now-closed encoder — just release the frame. The feed loop\n // then throws the AbortError on its next `throwIfAborted`.\n if (ctx.signal?.aborted) {\n frame.close();\n return;\n }\n const ts = frame.timestamp;\n const keyFrame = frameIndex === 0 || ts - lastKeyframeUs >= KEYFRAME_INTERVAL_US;\n if (keyFrame) lastKeyframeUs = ts;\n frameIndex++;\n // The encoder wrapper takes ownership and closes the frame in its finally,\n // so we never reuse it and no GPU surface leaks.\n encoder.encode(frame, { keyFrame });\n });\n\n const onAbort = (): void => {\n decoder.close();\n encoder.close();\n };\n ctx.signal?.addEventListener('abort', onAbort, { once: true });\n\n try {\n for (let i = 0; i < video.chunks.length; i++) {\n throwIfAborted(ctx.signal);\n const c = video.chunks[i];\n if (!c) continue;\n decoder.decode(\n new EncodedVideoChunk({\n type: c.type,\n timestamp: c.timestampUs,\n duration: c.durationUs,\n data: c.data,\n }),\n );\n }\n await decoder.flush();\n throwIfAborted(ctx.signal);\n await encoder.flush();\n throwIfAborted(ctx.signal);\n } catch (err) {\n throw asCodecError(err, 'video');\n } finally {\n ctx.signal?.removeEventListener('abort', onAbort);\n decoder.close();\n encoder.close();\n }\n\n return out;\n}\n\n// ---------------------------------------------------------------------------\n// Mux — shared cluster builder + webm/mkv adapters\n// ---------------------------------------------------------------------------\n\ninterface OutBlock {\n readonly trackNumber: number;\n readonly timestampUs: number;\n readonly keyframe: boolean;\n readonly data: Uint8Array;\n}\n\ninterface BuiltCluster {\n readonly fileOffset: number;\n readonly timecode: bigint;\n readonly blocks: Array<{\n trackNumber: number;\n timestampNs: bigint;\n keyframe: boolean;\n invisible: boolean;\n discardable: boolean;\n frames: Uint8Array[];\n }>;\n}\n\n/**\n * Merge video + audio blocks in timestamp order and split into clusters. A new\n * cluster begins on a video keyframe or whenever the tick delta from the current\n * cluster start would approach the int16 SimpleBlock limit.\n */\nfunction buildClusters(videoOut: VideoOut[], audioOut: EncodedOut[] | null): BuiltCluster[] {\n const blocks: OutBlock[] = [];\n for (const v of videoOut) {\n blocks.push({\n trackNumber: VIDEO_TRACK,\n timestampUs: v.timestampUs,\n keyframe: v.keyframe,\n data: v.data,\n });\n }\n if (audioOut) {\n for (const a of audioOut) {\n blocks.push({\n trackNumber: AUDIO_TRACK,\n timestampUs: a.timestampUs,\n keyframe: true,\n data: a.data,\n });\n }\n }\n // Stable sort by timestamp; video (pushed first) wins ties so a cluster opens\n // on the keyframe rather than a co-timed audio block.\n blocks.sort((x, y) => x.timestampUs - y.timestampUs);\n\n const clusters: BuiltCluster[] = [];\n let current: BuiltCluster | null = null;\n for (const b of blocks) {\n const tick = BigInt(Math.round(b.timestampUs / 1000)); // µs → 1 ms ticks\n const startNew =\n current === null ||\n (b.trackNumber === VIDEO_TRACK && b.keyframe) ||\n tick - current.timecode > BigInt(CLUSTER_MAX_TICKS);\n if (startNew) {\n current = { fileOffset: 0, timecode: tick, blocks: [] };\n clusters.push(current);\n }\n current?.blocks.push({\n trackNumber: b.trackNumber,\n timestampNs: tick * BigInt(WEBM_TIMECODE_SCALE),\n keyframe: b.keyframe,\n invisible: false,\n discardable: false,\n frames: [b.data],\n });\n }\n return clusters;\n}\n\nfunction buildWebm(\n width: number,\n height: number,\n videoCodec: 'vp9' | 'vp8',\n videoOut: VideoOut[],\n audioOut: EncodedOut[] | null,\n audioChannels: number,\n): Uint8Array {\n const videoTrack: WebmVideoTrack = {\n trackNumber: VIDEO_TRACK,\n trackUid: 1n,\n trackType: 1,\n codecId: videoCodec === 'vp8' ? 'V_VP8' : 'V_VP9',\n pixelWidth: width,\n pixelHeight: height,\n };\n const tracks: WebmFile['tracks'] = [videoTrack];\n if (audioOut) {\n const audioTrack: WebmAudioTrack = {\n trackNumber: AUDIO_TRACK,\n trackUid: 2n,\n trackType: 2,\n codecId: 'A_OPUS',\n codecPrivate: opusHead(audioChannels),\n samplingFrequency: OPUS_RATE,\n channels: audioChannels,\n };\n tracks.push(audioTrack);\n }\n\n const file: WebmFile = {\n ebmlHeader: {\n ebmlVersion: 1,\n ebmlReadVersion: 1,\n ebmlMaxIdLength: 4,\n ebmlMaxSizeLength: 8,\n docType: 'webm',\n docTypeVersion: 4,\n docTypeReadVersion: 2,\n },\n segmentPayloadOffset: 0,\n info: {\n timecodeScale: WEBM_TIMECODE_SCALE,\n muxingApp: 'webcvt-transcode',\n writingApp: 'webcvt-transcode',\n },\n tracks,\n clusters: buildClusters(videoOut, audioOut) as WebmCluster[],\n fileBytes: new Uint8Array(0),\n };\n return serializeWebm(file);\n}\n\nfunction buildMkv(\n width: number,\n height: number,\n videoCodec: 'vp9' | 'vp8',\n videoOut: VideoOut[],\n audioOut: EncodedOut[] | null,\n audioChannels: number,\n): Uint8Array {\n const videoTrack: MkvVideoTrack = {\n trackNumber: VIDEO_TRACK,\n trackUid: 1n,\n trackType: 1,\n codecId: videoCodec === 'vp8' ? 'V_VP8' : 'V_VP9',\n pixelWidth: width,\n pixelHeight: height,\n webcodecsCodecString: videoCodec === 'vp8' ? 'vp8' : 'vp09.00.10.08',\n };\n const tracks: MkvFile['tracks'] = [videoTrack];\n if (audioOut) {\n const audioTrack: MkvAudioTrack = {\n trackNumber: AUDIO_TRACK,\n trackUid: 2n,\n trackType: 2,\n codecId: 'A_OPUS',\n codecPrivate: opusHead(audioChannels),\n samplingFrequency: OPUS_RATE,\n channels: audioChannels,\n webcodecsCodecString: 'opus',\n };\n tracks.push(audioTrack);\n }\n\n const file: MkvFile = {\n ebmlHeader: {\n ebmlVersion: 1,\n ebmlReadVersion: 1,\n ebmlMaxIdLength: 4,\n ebmlMaxSizeLength: 8,\n docType: 'matroska',\n docTypeVersion: 4,\n docTypeReadVersion: 2,\n },\n segmentPayloadOffset: 0,\n info: {\n timecodeScale: WEBM_TIMECODE_SCALE,\n muxingApp: 'webcvt-transcode',\n writingApp: 'webcvt-transcode',\n },\n tracks,\n clusters: buildClusters(videoOut, audioOut) as MkvCluster[],\n chapters: [],\n tags: [],\n fileBytes: new Uint8Array(0),\n };\n return serializeMkv(file);\n}\n\nfunction opusHead(channels: number): Uint8Array {\n return buildOpusHead({\n channelCount: channels,\n preSkip: OPUS_PRE_SKIP,\n inputSampleRate: OPUS_RATE,\n });\n}\n\n// ---------------------------------------------------------------------------\n// Monotone progress\n// ---------------------------------------------------------------------------\n\nfunction monotone(\n onProgress: ((percent: number, phase: string) => void) | undefined,\n): (percent: number, phase: string) => void {\n let last = 0;\n return (percent, phase) => {\n const p = Math.max(last, Math.min(100, percent));\n last = p;\n onProgress?.(p, phase);\n };\n}\n","/**\n * TranscodeBackend — the WebCodecs-first audio transcode backend.\n *\n * `canHandle` is two-stage + cached: (1) the static {@link TRANSCODE_MATRIX}\n * gate rejects off-matrix pairs in O(1) with no probing; (2) both-sided\n * concrete codec probes (`probeAudioDecoder` for the input, `probeAudioCodec`\n * for the output) confirm the runtime actually supports the pair — this is what\n * makes Safari-audio and no-WebCodecs runtimes fall through to ffmpeg-wasm.\n * Probe results are memoised per session.\n *\n * `convert` runs the four-phase pipeline demux → decode → encode → mux, enforces\n * the input cap first, honours `options.signal`, and closes codecs on exit.\n */\n\nimport type { AudioCodecName } from '@catlabtech/webcvt-codec-webcodecs';\nimport type {\n Backend,\n BackendRegistry,\n ConvertOptions,\n ConvertResult,\n FormatDescriptor,\n} from '@catlabtech/webcvt-core';\nimport { defaultRegistry } from '@catlabtech/webcvt-core';\nimport { throwIfAborted } from './abort.ts';\nimport { decodeToPcm } from './decode.ts';\nimport { demuxAudio, demuxContainerAudio } from './demux.ts';\nimport {\n type EncodeContext,\n encodeAac,\n encodeFlac,\n encodeOpusOgg,\n encodeOpusWebm,\n encodeWav,\n} from './encode.ts';\nimport { TranscodeInputTooLargeError, TranscodeUnsupportedError } from './errors.ts';\nimport {\n CONTAINER_AUDIO_CODEC,\n CONTAINER_VIDEO_CODEC,\n type ContainerFamily,\n MAX_INPUT_BYTES,\n type OutputTarget,\n type SideCodec,\n TRANSCODE_MATRIX,\n type VideoTarget,\n containerFamilyFor,\n inputCodecFor,\n matrixKey,\n outputTargetFor,\n resolveBitrate,\n videoTargetFor,\n} from './matrix.ts';\nimport type { DecodedAudio } from './pcm.ts';\nimport { ProbeCache } from './probe-cache.ts';\nimport { transcodeVideo } from './video.ts';\n\nconst DEFAULT_QUALITY = 0.7;\n\nexport class TranscodeBackend implements Backend {\n readonly name = 'webcodecs-transcode';\n readonly priority = 0;\n\n readonly #probes = new ProbeCache();\n\n async canHandle(input: FormatDescriptor, output: FormatDescriptor): Promise<boolean> {\n // Audio-only matrix (wav/mp3/aac/flac/ogg → wav/opus/webm/aac/flac).\n if (TRANSCODE_MATRIX.has(matrixKey(input.mime, output.mime))) {\n return this.#canHandleAudioMatrix(input.mime, output.mime);\n }\n // Container inputs (mp4/m4a/webm/mkv): video transcode OR audio-track\n // extraction into the audio matrix. Kept separate so the frozen audio\n // matrix (and `inputCodecFor('video/mp4')`) stays unchanged.\n const family = containerFamilyFor(input.mime);\n if (family) return this.#canHandleContainer(family, input, output);\n return false;\n }\n\n /** Stage-2 both-sided probe for the frozen audio matrix. */\n async #canHandleAudioMatrix(inputMime: string, outputMime: string): Promise<boolean> {\n const inputCodec = inputCodecFor(inputMime);\n const target = outputTargetFor(outputMime);\n if (!inputCodec || !target) return false;\n\n // A pcm side (wav) needs no codec — skip it.\n if (inputCodec !== 'pcm') {\n if (!(await this.#probes.canDecode(asAudioCodec(inputCodec)))) return false;\n }\n if (target.codec !== 'pcm') {\n if (!(await this.#probes.canEncode(asAudioCodec(target.codec)))) return false;\n }\n return true;\n }\n\n /** Container input: gate a video-target or an audio-track-extraction pair. */\n async #canHandleContainer(\n family: ContainerFamily,\n input: FormatDescriptor,\n output: FormatDescriptor,\n ): Promise<boolean> {\n // Video output → transcode the container's video track. Requires a\n // video-bearing input (m4a / audio-only containers cannot supply video).\n const videoTarget = videoTargetFor(output.mime);\n if (videoTarget && input.category === 'video') {\n if (!(await this.#probes.canDecodeVideo(CONTAINER_VIDEO_CODEC[family]))) return false;\n const canVp9 = await this.#probes.canEncodeVideo('vp9');\n const canVp8 = canVp9 || (await this.#probes.canEncodeVideo('vp8'));\n return canVp9 || canVp8;\n }\n\n // Audio output → extract the container's audio track into the audio matrix.\n const target = outputTargetFor(output.mime);\n if (target && output.mime !== input.mime) {\n if (!(await this.#probes.canDecode(CONTAINER_AUDIO_CODEC[family]))) return false;\n if (target.codec !== 'pcm') {\n if (!(await this.#probes.canEncode(asAudioCodec(target.codec)))) return false;\n }\n return true;\n }\n return false;\n }\n\n async convert(\n input: Blob,\n output: FormatDescriptor,\n options: ConvertOptions,\n ): Promise<ConvertResult> {\n const startMs = Date.now();\n\n // Input cap FIRST — reject before touching the bytes so a hostile `.size`\n // never triggers an allocation.\n if (input.size > MAX_INPUT_BYTES) {\n throw new TranscodeInputTooLargeError(input.size, MAX_INPUT_BYTES);\n }\n\n const signal = options.signal;\n throwIfAborted(signal);\n\n // Backends dispatch on Blob.type (core re-types the blob to the resolved\n // input MIME before calling convert). Fall back to options.inputFormat.\n const inputMime = resolveInputMime(input, options);\n\n // 1. Audio-only matrix (wav/mp3/aac/flac/ogg audio).\n const inputCodec = inputCodecFor(inputMime);\n const target = outputTargetFor(output.mime);\n if (inputCodec && target && TRANSCODE_MATRIX.has(matrixKey(inputMime, output.mime))) {\n const bytes = new Uint8Array(await input.arrayBuffer());\n const outputBytes = await this.#runAudio(demuxAudio(inputCodec, bytes), target, options);\n return this.#audioResult(outputBytes, output, startMs);\n }\n\n // 2. Container inputs (mp4/m4a/webm/mkv).\n const family = containerFamilyFor(inputMime);\n if (family) {\n const bytes = new Uint8Array(await input.arrayBuffer());\n const videoTarget = videoTargetFor(output.mime);\n if (videoTarget && output.category === 'video') {\n return this.#convertVideo(family, videoTarget, bytes, output, options, startMs);\n }\n if (target) {\n const outputBytes = await this.#runAudio(\n demuxContainerAudio(family, bytes),\n target,\n options,\n );\n return this.#audioResult(outputBytes, output, startMs);\n }\n }\n\n throw new TranscodeUnsupportedError(inputMime, output.mime);\n }\n\n /** Shared audio demux→decode→encode→mux for both audio-matrix and container-audio. */\n async #runAudio(\n demux: ReturnType<typeof demuxAudio>,\n target: OutputTarget,\n options: ConvertOptions,\n ): Promise<Uint8Array> {\n const signal = options.signal;\n const report = (percent: number, phase: string): void =>\n options.onProgress?.({ percent, phase });\n\n report(10, 'demux');\n throwIfAborted(signal);\n\n // Decode (10 → 45%).\n const decoded = await decodeToPcm(demux, {\n signal,\n onProgress: (f) => report(10 + f * 35, 'decode'),\n });\n report(45, 'decode');\n throwIfAborted(signal);\n\n // Encode (45 → 90%) then mux, folded into the sink.\n const quality = options.quality ?? DEFAULT_QUALITY;\n const bitrate = resolveBitrate(target.codec, quality, decoded.numberOfChannels);\n const encodeCtx: EncodeContext = {\n signal,\n onProgress: (f) => report(45 + f * 45, 'encode'),\n };\n const outputBytes = await runSink(target, decoded, { bitrate }, encodeCtx);\n report(90, 'mux');\n throwIfAborted(signal);\n report(100, 'done');\n return outputBytes;\n }\n\n #audioResult(outputBytes: Uint8Array, output: FormatDescriptor, startMs: number): ConvertResult {\n return {\n blob: new Blob([outputBytes.buffer as ArrayBuffer], { type: output.mime }),\n format: output,\n durationMs: Date.now() - startMs,\n backend: this.name,\n hardwareAccelerated: false, // audio decode/encode is always software today.\n };\n }\n\n /** Video transcode dispatch (mp4/webm/mkv → webm/mkv, VP9|VP8 + Opus). */\n async #convertVideo(\n family: ContainerFamily,\n videoTarget: VideoTarget,\n bytes: Uint8Array,\n output: FormatDescriptor,\n options: ConvertOptions,\n startMs: number,\n ): Promise<ConvertResult> {\n const result = await transcodeVideo(family, videoTarget.container, bytes, this.#probes, {\n quality: options.quality ?? DEFAULT_QUALITY,\n signal: options.signal,\n ...(options.hardwareAcceleration\n ? { hardwareAcceleration: mapHwAccel(options.hardwareAcceleration) }\n : {}),\n onProgress: (percent, phase) => options.onProgress?.({ percent, phase }),\n });\n\n // Video-only fallback surface: when the source had audio we could not\n // transcode (Safari-style video-only WebCodecs, or a non-decodable audio\n // codec), the output is video-only. There is no ConvertResult field for\n // this by design — it is surfaced via a console warning + the docs/report.\n if (!result.audioIncluded) {\n console.warn(\n '[webcvt-transcode] emitted a VIDEO-ONLY output: the source audio track could not be ' +\n 'transcoded (no WebCodecs AudioDecoder/AudioEncoder, or an unsupported audio codec).',\n );\n }\n\n return {\n blob: new Blob([result.bytes.buffer as ArrayBuffer], { type: output.mime }),\n format: output,\n durationMs: Date.now() - startMs,\n backend: this.name,\n hardwareAccelerated: result.hardwareAccelerated,\n };\n }\n}\n\n/**\n * Construct a TranscodeBackend and register it with the given registry (or\n * core's `defaultRegistry` when omitted). Returns the backend so the caller can\n * later `registry.unregister('webcodecs-transcode')`. Nothing registers on\n * import — call this explicitly.\n */\nexport function registerTranscodeBackend(\n registry: BackendRegistry = defaultRegistry,\n): TranscodeBackend {\n const backend = new TranscodeBackend();\n registry.register(backend);\n return backend;\n}\n\n// ---------------------------------------------------------------------------\n// Internal helpers\n// ---------------------------------------------------------------------------\n\nfunction runSink(\n target: OutputTarget,\n decoded: DecodedAudio,\n opts: { bitrate?: number },\n ctx: EncodeContext,\n): Promise<Uint8Array> | Uint8Array {\n switch (target.container) {\n case 'wav':\n return encodeWav(decoded);\n case 'ogg':\n return encodeOpusOgg(decoded, opts, ctx);\n case 'webm':\n return encodeOpusWebm(decoded, opts, ctx);\n case 'aac':\n return encodeAac(decoded, opts, ctx);\n case 'flac':\n return encodeFlac(decoded, opts, ctx);\n default:\n throw new TranscodeUnsupportedError('audio', target.container);\n }\n}\n\nfunction resolveInputMime(input: Blob, options: ConvertOptions): string {\n if (options.inputFormat !== undefined) {\n if (typeof options.inputFormat === 'string') {\n // Extension or MIME string — only MIME is meaningful for demuxer routing.\n if (options.inputFormat.includes('/')) return options.inputFormat;\n } else {\n return options.inputFormat.mime;\n }\n }\n return input.type;\n}\n\n/** SideCodec (non-pcm) → codec-webcodecs AudioCodecName. */\nfunction asAudioCodec(codec: Exclude<SideCodec, 'pcm'>): AudioCodecName {\n return codec;\n}\n\n/** Core's `HardwareAcceleration` hint → the WebCodecs `hardwareAcceleration` enum. */\nfunction mapHwAccel(\n pref: NonNullable<ConvertOptions['hardwareAcceleration']>,\n): HardwareAcceleration {\n switch (pref) {\n case 'no':\n return 'prefer-software';\n case 'preferred':\n case 'required':\n return 'prefer-hardware';\n default:\n return 'no-preference';\n }\n}\n"]}

package/dist/index.d.cts ADDED Viewed

@@ -0,0 +1,172 @@
+import { Backend, FormatDescriptor, ConvertOptions, ConvertResult, BackendRegistry, WebcvtError } from '@catlabtech/webcvt-core';
+/**
+ * TranscodeBackend — the WebCodecs-first audio transcode backend.
+ *
+ * `canHandle` is two-stage + cached: (1) the static {@link TRANSCODE_MATRIX}
+ * gate rejects off-matrix pairs in O(1) with no probing; (2) both-sided
+ * concrete codec probes (`probeAudioDecoder` for the input, `probeAudioCodec`
+ * for the output) confirm the runtime actually supports the pair — this is what
+ * makes Safari-audio and no-WebCodecs runtimes fall through to ffmpeg-wasm.
+ * Probe results are memoised per session.
+ *
+ * `convert` runs the four-phase pipeline demux → decode → encode → mux, enforces
+ * the input cap first, honours `options.signal`, and closes codecs on exit.
+ */
+declare class TranscodeBackend implements Backend {
+    #private;
+    readonly name = "webcodecs-transcode";
+    readonly priority = 0;
+    canHandle(input: FormatDescriptor, output: FormatDescriptor): Promise<boolean>;
+    convert(input: Blob, output: FormatDescriptor, options: ConvertOptions): Promise<ConvertResult>;
+}
+/**
+ * Construct a TranscodeBackend and register it with the given registry (or
+ * core's `defaultRegistry` when omitted). Returns the backend so the caller can
+ * later `registry.unregister('webcodecs-transcode')`. Nothing registers on
+ * import — call this explicitly.
+ */
+declare function registerTranscodeBackend(registry?: BackendRegistry): TranscodeBackend;
+/**
+ * Typed error classes for @catlabtech/webcvt-transcode.
+ *
+ * All extend core's {@link WebcvtError} with a `TRANSCODE_`-prefixed code, so
+ * callers can catch by `instanceof` or match on `.code`. Never throw a bare
+ * `Error` from this package.
+ */
+/**
+ * Thrown when the requested conversion is not in the transcode backend's
+ * capability set (off-matrix pair, or a codec whose runtime probe failed).
+ * `convert()` should never be reached for an unsupported pair — `canHandle`
+ * gates it — but this guards the defensive path.
+ */
+declare class TranscodeUnsupportedError extends WebcvtError {
+    constructor(from: string, to: string);
+}
+/**
+ * Thrown when the input Blob exceeds the buffer-all input cap. The whole
+ * pipeline is buffer-all today (every serializer takes a complete in-memory
+ * model), so a hard cap protects against OOM.
+ */
+declare class TranscodeInputTooLargeError extends WebcvtError {
+    constructor(size: number, max: number);
+}
+/** Thrown when demuxing the input container fails or yields no audio. */
+declare class TranscodeDemuxError extends WebcvtError {
+    constructor(message: string, options?: ErrorOptions);
+}
+/** Wraps a decode/encode failure surfaced by codec-webcodecs. */
+declare class TranscodeCodecError extends WebcvtError {
+    constructor(message: string, options?: ErrorOptions);
+}
+/** Thrown when muxing the encoded output container fails. */
+declare class TranscodeMuxError extends WebcvtError {
+    constructor(message: string, options?: ErrorOptions);
+}
+/**
+ * Static v1 capability matrix + codec/bitrate resolution for the audio
+ * transcode paths.
+ *
+ * `canHandle` is two-stage: this frozen matrix is the cheap O(1) first gate
+ * (rejects off-matrix pairs like `→ mp3`/`→ mp4` and every video pair before
+ * any `isConfigSupported` round-trip), and the concrete both-sided codec probe
+ * is the second gate (see backend.ts). See docs/design-notes/transcode.md §D.
+ */
+/** Codec token for a demux/mux side. `'pcm'` = raw PCM (wav), no WebCodecs. */
+type SideCodec = 'pcm' | 'opus' | 'aac' | 'flac' | 'mp3';
+/** Output container family we can mux from scratch. */
+type OutputContainer = 'wav' | 'ogg' | 'webm' | 'aac' | 'flac';
+interface OutputTarget {
+    readonly container: OutputContainer;
+    /** Codec the WebCodecs encoder must produce (`'pcm'` = interleave, no encode). */
+    readonly codec: SideCodec;
+}
+/**
+ * MIME → input codec for the containers this stage can demux. `audio/ogg` is
+ * assumed to carry Opus (Vorbis-in-Ogg decode needs a multi-packet setup
+ * description and is deferred). mp4/m4a/webm/mkv audio-track extraction is
+ * deferred to the video stage (shared demuxers).
+ */
+declare const INPUT_CODECS: Readonly<Record<string, SideCodec>>;
+/**
+ * MIME → output target. `→ mp3` and `→ ogg(vorbis)` are intentionally absent
+ * (no WebCodecs encoder); `→ mp4/m4a` is absent (no from-scratch mp4 muxer yet).
+ */
+declare const OUTPUT_TARGETS: Readonly<Record<string, OutputTarget>>;
+/**
+ * Frozen set of `"${inputMime}|${outputMime}"` keys — the full product of every
+ * decodable input against every muxable output. This is the static gate.
+ */
+declare const TRANSCODE_MATRIX: ReadonlySet<string>;
+/** Build the matrix key for a pair. */
+declare function matrixKey(inputMime: string, outputMime: string): string;
+/** Lookup the input codec for a MIME, or `undefined` if not decodable here. */
+declare function inputCodecFor(mime: string): SideCodec | undefined;
+/** Lookup the output target for a MIME, or `undefined` if not muxable here. */
+declare function outputTargetFor(mime: string): OutputTarget | undefined;
+/** Demuxable container family (its sample/block iterators feed the pipeline). */
+type ContainerFamily = 'mp4' | 'webm' | 'mkv';
+/** WebCodecs video codec token for a demux/mux side. */
+type VideoSideCodec = 'h264' | 'vp9' | 'vp8' | 'av1' | 'hevc';
+/**
+ * Container input MIME → family. Covers the video containers (video/mp4,
+ * video/webm, video/x-matroska) and the audio-only projections that share a
+ * demuxer (audio/mp4 = m4a, audio/webm). A container input can be transcoded
+ * to a video target (its video track) OR have its audio track routed into the
+ * existing audio matrix. Kept out of INPUT_CODECS so `inputCodecFor('video/mp4')`
+ * stays `undefined` (a container is not one audio codec).
+ */
+declare const CONTAINER_INPUTS: Readonly<Record<string, ContainerFamily>>;
+/** Output container family for a video target (VP9 default, VP8 fallback). */
+interface VideoTarget {
+    readonly container: 'webm' | 'mkv';
+}
+/**
+ * Video output MIME → container. `→ mp4` is intentionally absent (no from-scratch
+ * mp4 muxer yet — deferred to a later stage). The encoder codec (VP9 preferred,
+ * VP8 fallback) is resolved at convert time by probing, not fixed here.
+ */
+declare const VIDEO_TARGETS: Readonly<Record<string, VideoTarget>>;
+/**
+ * Assumed audio-track codec per container family, used only by `canHandle`
+ * probing (the true codec is known after demux). mp4 audio is ~always AAC and
+ * webm/mkv audio is ~always Opus. A mismatch (e.g. Vorbis-in-webm) is rejected
+ * at demux — the same optimistic-probe contract the ogg(opus) input already uses.
+ */
+declare const CONTAINER_AUDIO_CODEC: Readonly<Record<ContainerFamily, Exclude<SideCodec, 'pcm'>>>;
+/** Assumed video-track codec per family, for `canHandle` decoder probing. */
+declare const CONTAINER_VIDEO_CODEC: Readonly<Record<ContainerFamily, VideoSideCodec>>;
+/** Lookup the container family for an input MIME, or `undefined`. */
+declare function containerFamilyFor(mime: string): ContainerFamily | undefined;
+/** Lookup the video output target for a MIME, or `undefined`. */
+declare function videoTargetFor(mime: string): VideoTarget | undefined;
+/** Default quality when `ConvertOptions.quality` is omitted (mid-ladder). */
+declare const DEFAULT_QUALITY = 0.7;
+/**
+ * Buffer-all input cap. Every serializer takes a complete in-memory model, so a
+ * hard cap protects against OOM. 256 MiB per the design note's conservative
+ * default; revisit when a streaming muxer exists (decoded PCM expands well
+ * beyond the compressed input size).
+ */
+declare const MAX_INPUT_BYTES: number;
+/**
+ * Map `quality` (0–1) to a codec bitrate in bits/s. The ladder is piecewise so
+ * the default `quality = 0.7` lands exactly on the mid-ladder figure from the
+ * design note (Opus 128 kbps, AAC 128 kbps stereo), with q=0 and q=1 at the
+ * ladder ends. Mono is scaled to ~0.6×. Lossless FLAC returns `undefined`
+ * (bitrate is not applicable).
+ */
+declare function resolveBitrate(codec: SideCodec, quality: number, channels: number): number | undefined;
+/**
+ * Map video geometry + `quality` to a VP9/VP8 bitrate in bits/s. Resolution
+ * drives the base (design note ladder: 720p ≈ 3 Mbps for VP9), scaled ±40% by
+ * quality around the default `0.7`, with VP8 at ≈1.3× VP9. Clamped to a sane
+ * floor so tiny test clips still get a positive bitrate.
+ */
+declare function resolveVideoBitrate(width: number, height: number, quality: number, codec: 'vp9' | 'vp8'): number;
+export { CONTAINER_AUDIO_CODEC, CONTAINER_INPUTS, CONTAINER_VIDEO_CODEC, type ContainerFamily, DEFAULT_QUALITY, INPUT_CODECS, MAX_INPUT_BYTES, OUTPUT_TARGETS, type OutputContainer, type OutputTarget, type SideCodec, TRANSCODE_MATRIX, TranscodeBackend, TranscodeCodecError, TranscodeDemuxError, TranscodeInputTooLargeError, TranscodeMuxError, TranscodeUnsupportedError, VIDEO_TARGETS, type VideoSideCodec, type VideoTarget, containerFamilyFor, inputCodecFor, matrixKey, outputTargetFor, registerTranscodeBackend, resolveBitrate, resolveVideoBitrate, videoTargetFor };