@e9g/buffered-audio-nodes 0.2.0

package/README.md

@@ -0,0 +1,532 @@
+# buffered-audio-nodes
+
+A streaming audio processing protocol. Chainable modules that read, transform, and write audio — an open, scriptable, extensible alternative to GUI-bound audio engineering tools.
+
+## Install
+
+```bash
+npm install @e9g/buffered-audio-nodes
+```
+
+## Usage
+
+Three module types — sources produce audio, transforms process it, targets consume it. The `chain()` function wires them into a pipeline.
+
+```ts
+import { chain, read, normalize, write } from "@e9g/buffered-audio-nodes";
+
+const source = chain(
+  read("input.wav"),
+  normalize({ ceiling: 0.95 }),
+  write("output.wav", { bitDepth: "24" })
+);
+
+await source.render();
+```
+
+`render()` streams audio through the chain. Backpressure, buffering, and lifecycle are handled by the framework.
+
+### Fan
+
+Split a stream into parallel branches with `fan()`:
+
+```ts
+import { chain, read, fan, normalize, trim, write } from "@e9g/buffered-audio-nodes";
+
+const source = chain(
+  read("input.wav"),
+  fan(
+    chain(normalize(), write("normalized.wav")),
+    chain(trim(), write("trimmed.wav"))
+  )
+);
+
+await source.render();
+```
+
+## CLI
+
+Run pipelines from TypeScript files. The file's default export must be a `SourceModule`.
+
+```bash
+npx @e9g/buffered-audio-nodes process --pipeline pipeline.ts
+```
+
+```ts
+// pipeline.ts
+import { chain, read, normalize, trim, write } from "@e9g/buffered-audio-nodes";
+
+export default chain(
+  read("input.wav"),
+  normalize(),
+  trim({ threshold: -60 }),
+  write("output.wav")
+);
+```
+
+| Flag | Description |
+|------|-------------|
+| `--chunk-size <samples>` | Chunk size in samples |
+| `--high-water-mark <count>` | Stream backpressure high water mark |
+
+## Modules
+
+### Sources
+
+#### `read(path, options?)`
+
+Read audio from a file. WAV files are read natively. Other formats are transcoded via FFmpeg.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `path` | `string` | | File path |
+| `channels` | `number[]` | all | Channel indices to extract |
+| `ffmpegPath` | `string` | | Path to [ffmpeg](https://ffmpeg.org/download.html). Required for non-WAV files. |
+| `ffprobePath` | `string` | | Path to [ffprobe](https://ffmpeg.org/download.html). Required for non-WAV files. |
+
+### Targets
+
+#### `write(path, options?)`
+
+Write audio to a file. Writes WAV natively. Other formats are encoded via FFmpeg.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `path` | `string` | | Output file path |
+| `bitDepth` | `"16" \| "24" \| "32" \| "32f"` | `"16"` | WAV bit depth |
+| `encoding` | `EncodingOptions` | | Non-WAV encoding (format, bitrate, vbr) |
+| `ffmpegPath` | `string` | | Path to [ffmpeg](https://ffmpeg.org/download.html). Required for non-WAV formats. |
+
+### Composites
+
+#### `chain(...modules)`
+
+Wire modules into a linear pipeline. Returns the source module.
+
+#### `fan(...branches)`
+
+Split a stream into parallel transform branches.
+
+### Transforms — Basic
+
+#### `normalize(options?)`
+
+Adjust peak level to a target ceiling.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ceiling` | `number` | `1.0` | Target peak level (0–1) |
+
+#### `trim(options?)`
+
+Remove silence from start and end.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `threshold` | `number` | `0.001` | Silence threshold (0–1) |
+| `margin` | `number` | `0.01` | Margin to keep around content (0–1) |
+| `start` | `boolean` | `true` | Trim start |
+| `end` | `boolean` | `true` | Trim end |
+
+#### `cut(options?)`
+
+Remove regions from audio.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `regions` | `CutRegion[]` | | Regions to remove (`{ start, end }` in samples) |
+
+#### `pad(options?)`
+
+Add silence to start or end.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `before` | `number` | `0` | Samples of silence before audio |
+| `after` | `number` | `0` | Samples of silence after audio |
+
+#### `dither(options?)`
+
+Add dither noise before bit-depth reduction.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `bitDepth` | `"16" \| "24"` | `"16"` | Target bit depth |
+| `noiseShaping` | `boolean` | `false` | Apply noise shaping |
+
+#### `phase(options?)` / `invert()`
+
+Invert polarity or shift phase.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `invert` | `boolean` | `true` | Invert polarity |
+| `angle` | `number` | `0` | Phase angle (-180 to 180) |
+
+#### `reverse()`
+
+Reverse audio. No parameters.
+
+#### `splice(options?)`
+
+Insert audio at a position.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `insertPath` | `string` | | WAV file to insert |
+| `insertAt` | `number` | `0` | Insert position in samples |
+
+#### `waveform(options?)`
+
+Extract waveform data to a file.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `outputPath` | `string` | | Output file path |
+| `resolution` | `number` | `1000` | Number of waveform points (100–10000) |
+
+### Transforms — FFmpeg
+
+These transforms require an [ffmpeg](https://ffmpeg.org/download.html) binary.
+
+#### `ffmpeg(options)`
+
+Run arbitrary FFmpeg filters.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `args` | `string[] \| (ctx) => string[]` | `[]` | FFmpeg filter arguments |
+
+#### `resample(ffmpegPath, sampleRate, options?)`
+
+Change sample rate.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `sampleRate` | `number` | `44100` | Target sample rate (8000–192000) |
+| `dither` | `"triangular" \| "lipshitz" \| "none"` | `"triangular"` | Dither method |
+
+#### `loudness(ffmpegPath, options?)`
+
+Adjust loudness to EBU R128 target.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `target` | `number` | `-14` | Target LUFS (-50 to 0) |
+| `truePeak` | `number` | `-1` | True peak limit dBTP (-10 to 0) |
+| `lra` | `number` | `0` | Loudness range target (0–20) |
+
+#### `loudnessStats(options?)`
+
+Measure loudness statistics. Access results via `.stats` after render. No parameters.
+
+#### `spectrogram(outputPath, options?)`
+
+Generate spectrogram data file.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `outputPath` | `string` | | Output file path |
+| `fftSize` | `number` | `2048` | FFT size (256–8192) |
+| `hopSize` | `number` | `512` | Hop size (64–4096) |
+| `frequencyScale` | `"linear" \| "log"` | `"log"` | Frequency axis scale |
+| `numBands` | `number` | `512` | Number of frequency bands (log scale) |
+| `minFrequency` | `number` | `20` | Minimum frequency in Hz |
+| `maxFrequency` | `number` | `sr/2` | Maximum frequency in Hz |
+
+### Transforms — Audio Engineering
+
+#### `breathControl(options?)`
+
+Reduce or remove breath sounds.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.5` | Detection sensitivity (0–1) |
+| `reduction` | `number` | `-12` | Reduction in dB (-60 to 0) |
+| `mode` | `"remove" \| "attenuate"` | `"attenuate"` | Processing mode |
+
+#### `deBleed(referencePath, options?)`
+
+Remove microphone bleed using a reference signal.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `referencePath` | `string` | | Path to reference WAV |
+| `filterLength` | `number` | `1024` | Adaptive filter length (64–8192) |
+| `stepSize` | `number` | `0.1` | Adaptation step size (0.001–1) |
+
+#### `deClick(options?)`
+
+Remove clicks and pops.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.5` | Detection sensitivity (0–1) |
+| `maxClickDuration` | `number` | `200` | Max click duration in samples (1–1000) |
+
+#### `deCrackle(options?)`
+
+Remove crackle noise.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.5` | Detection sensitivity (0–1) |
+
+#### `mouthDeClick(options?)`
+
+Remove mouth clicks.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.7` | Detection sensitivity (0–1) |
+
+#### `deClip(options?)`
+
+Repair clipped audio.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `threshold` | `number` | `0.99` | Clipping detection threshold (0–1) |
+| `method` | `"ar" \| "sparse"` | `"ar"` | Repair method |
+
+#### `dePlosive(options?)`
+
+Reduce plosive sounds.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.5` | Detection sensitivity (0–1) |
+| `frequency` | `number` | `200` | Plosive frequency cutoff in Hz (50–500) |
+
+#### `deReverb(options?)`
+
+Reduce reverb and room tone using WPE dereverberation.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `sensitivity` | `number` | `0.5` | Controls prediction delay, filter length, and iterations (0–1) |
+| `predictionDelay` | `number` | | Prediction delay in frames (1–10) |
+| `filterLength` | `number` | | Filter length in frames (5–30) |
+| `iterations` | `number` | | WPE iterations (1–10) |
+| `vkfftAddonPath` | `string` | | Path to [vkfft-addon](https://github.com/visionsofparadise/vkfft-addon) (GPU FFT) |
+| `fftwAddonPath` | `string` | | Path to [fftw-addon](https://github.com/visionsofparadise/fftw-addon) (CPU FFT) |
+
+#### `dialogueIsolate(options)`
+
+Isolate dialogue using MDX-Net vocal separation.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `modelPath` | `string` | | Path to [Kim_Vocal_2.onnx](https://huggingface.co/seanghay/uvr_models) |
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `onnxAddonPath` | `string` | | Path to [onnx-runtime-addon](https://github.com/visionsofparadise/onnx-runtime-addon) |
+| `highPass` | `number` | `80` | High pass filter in Hz (20–500) |
+| `lowPass` | `number` | `20000` | Low pass filter in Hz (1000–22050) |
+
+#### `eqMatch(referencePath, options?)`
+
+Match EQ profile to a reference recording.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `referencePath` | `string` | | Path to reference WAV |
+| `smoothing` | `number` | `0.33` | Spectral smoothing (0–1) |
+| `vkfftAddonPath` | `string` | | Path to [vkfft-addon](https://github.com/visionsofparadise/vkfft-addon) |
+| `fftwAddonPath` | `string` | | Path to [fftw-addon](https://github.com/visionsofparadise/fftw-addon) |
+
+#### `leveler(options?)`
+
+Automatic level control.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `target` | `number` | `-20` | Target level in dB (-60 to 0) |
+| `window` | `number` | `0.5` | Analysis window in seconds (0.01–5) |
+| `speed` | `number` | `0.1` | Adjustment speed (0.01–1) |
+| `maxGain` | `number` | `12` | Maximum gain in dB (0–40) |
+| `maxCut` | `number` | `12` | Maximum cut in dB (0–40) |
+
+#### `musicRebalance(modelPath, stems, options?)`
+
+Adjust stem levels using HTDemucs source separation.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `modelPath` | `string` | | Path to [htdemucs.onnx](https://github.com/facebookresearch/demucs) (requires .onnx.data file alongside) |
+| `stems` | `Partial<StemGains>` | | Gain per stem: `{ vocals, drums, bass, other }` |
+| `onnxAddonPath` | `string` | | Path to [onnx-runtime-addon](https://github.com/visionsofparadise/onnx-runtime-addon) |
+
+#### `pitchShift(ffmpegPath, semitones, options?)`
+
+Shift pitch without changing duration.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `semitones` | `number` | | Semitones to shift (-24 to 24) |
+| `cents` | `number` | `0` | Cents to shift (-100 to 100) |
+
+#### `spectralRepair(regions, options?)`
+
+Repair spectral regions.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `regions` | `SpectralRegion[]` | | Regions to repair (`{ startTime, endTime, startFreq, endFreq }`) |
+| `method` | `"ar" \| "nmf"` | `"ar"` | Repair method |
+| `vkfftAddonPath` | `string` | | Path to [vkfft-addon](https://github.com/visionsofparadise/vkfft-addon) |
+| `fftwAddonPath` | `string` | | Path to [fftw-addon](https://github.com/visionsofparadise/fftw-addon) |
+
+#### `timeStretch(ffmpegPath, rate, options?)`
+
+Change duration without affecting pitch.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `rate` | `number` | | Time stretch rate (0.25–4) |
+
+#### `voiceDenoise(options)`
+
+Remove background noise from speech using DTLN.
+
+| Parameter | Type | Default | Description |
+|-----------|------|---------|-------------|
+| `modelPath1` | `string` | | Path to [DTLN model_1.onnx](https://github.com/breizhn/DTLN) (magnitude mask estimation) |
+| `modelPath2` | `string` | | Path to [DTLN model_2.onnx](https://github.com/breizhn/DTLN) (signal reconstruction) |
+| `ffmpegPath` | `string` | | Path to ffmpeg |
+| `onnxAddonPath` | `string` | | Path to [onnx-runtime-addon](https://github.com/visionsofparadise/onnx-runtime-addon) |
+| `vkfftAddonPath` | `string` | | Path to [vkfft-addon](https://github.com/visionsofparadise/vkfft-addon) |
+| `fftwAddonPath` | `string` | | Path to [fftw-addon](https://github.com/visionsofparadise/fftw-addon) |
+
+## Creating Modules
+
+Extend `SourceModule`, `TransformModule`, or `TargetModule`.
+
+### Transform
+
+The most common module type. A transform's `bufferSize` controls how audio is collected before processing:
+
+- `0` — streaming. Each chunk passes through `_unbuffer` immediately.
+- `n` — block-based. Chunks accumulate in a `ChunkBuffer` until `n` frames are collected, then `_process` runs on the buffer and `_unbuffer` emits the result.
+- `Infinity` — full-file. All audio is buffered before `_process` and `_unbuffer` run.
+
+The three hooks:
+
+- **`_buffer(chunk, buffer)`** — called for each incoming chunk. Override to inspect or modify data as it's buffered. Default appends to the buffer.
+- **`_process(buffer)`** — called once the buffer reaches `bufferSize`. Use this for analysis or in-place modification of the full buffer.
+- **`_unbuffer(chunk)`** — called for each chunk emitted from the buffer. Transform or replace the chunk here. Return `undefined` to drop it.
+
+```ts
+import {
+  TransformModule,
+  type TransformModuleProperties,
+  type AudioChunk,
+  type ChunkBuffer,
+} from "@e9g/buffered-audio-nodes";
+import { z } from "zod";
+
+const schema = z.object({
+  ceiling: z.number().min(0).max(1).default(1.0),
+});
+
+interface NormalizeProperties extends z.infer<typeof schema>, TransformModuleProperties {}
+
+class NormalizeModule extends TransformModule<NormalizeProperties> {
+  static override readonly moduleName = "Normalize";
+  static override readonly moduleDescription = "Adjust peak level to a target ceiling";
+  static override readonly schema = schema;
+
+  override readonly type = ["buffered-audio-node", "transform", "normalize"] as const;
+
+  // Buffer all audio before processing — we need to find the peak first
+  override readonly bufferSize = Infinity;
+  override readonly latency = Infinity;
+
+  private peak = 0;
+  private scale = 1;
+
+  // _buffer: called for each incoming chunk. Track the peak while buffering.
+  override async _buffer(chunk: AudioChunk, buffer: ChunkBuffer): Promise<void> {
+    await super._buffer(chunk, buffer);
+
+    for (const channel of chunk.samples) {
+      for (const sample of channel) {
+        const absolute = Math.abs(sample);
+        if (absolute > this.peak) this.peak = absolute;
+      }
+    }
+  }
+
+  // _process: called once all audio is buffered. Compute the gain scale.
+  override _process(_buffer: ChunkBuffer): void {
+    this.scale = this.peak === 0 ? 1 : this.properties.ceiling / this.peak;
+  }
+
+  // _unbuffer: called for each chunk emitted from the buffer. Apply the gain.
+  override _unbuffer(chunk: AudioChunk): AudioChunk {
+    if (this.scale === 1) return chunk;
+
+    const scaled = chunk.samples.map((channel) => {
+      const out = new Float32Array(channel.length);
+      for (let i = 0; i < channel.length; i++) {
+        out[i] = (channel[i] ?? 0) * this.scale;
+      }
+      return out;
+    });
+
+    return { samples: scaled, offset: chunk.offset, duration: chunk.duration };
+  }
+
+  clone(overrides?: Partial<NormalizeProperties>): NormalizeModule {
+    return new NormalizeModule({ ...this.properties, previousProperties: this.properties, ...overrides });
+  }
+}
+
+export function normalize(options?: { ceiling?: number }): NormalizeModule {
+  return new NormalizeModule({ ceiling: options?.ceiling ?? 1.0 });
+}
+```
+
+### Source
+
+Implement `_init` to return stream metadata (`sampleRate`, `channels`, `duration`), `_read` to produce chunks via the controller, and `_flush` for cleanup.
+
+### Target
+
+Implement `_write` to consume each chunk and `_close` to finalize (close file handles, flush buffers).
+
+### FFT Backends
+
+Transforms that use spectral processing (STFT/iSTFT) can use native FFT backends for performance. The framework selects a backend based on the stream's `executionProviders` preference:
+
+| Backend | Provider | Addon | Description |
+|---------|----------|-------|-------------|
+| VkFFT | `gpu` | [vkfft-addon](https://github.com/visionsofparadise/vkfft-addon) | GPU-accelerated FFT via Vulkan |
+| FFTW | `cpu-native` | [fftw-addon](https://github.com/visionsofparadise/fftw-addon) | Native CPU FFT |
+| JavaScript | `cpu` | Built-in | Pure JS fallback, no addon needed |
+
+Pass addon paths via module properties (`vkfftAddonPath`, `fftwAddonPath`). Falls back to the built-in JavaScript implementation when no native addon is available.
+
+### ONNX Models
+
+ML-based transforms use ONNX Runtime for inference via a native addon. Modules that use ONNX accept:
+
+- `onnxAddonPath` — path to the [onnx-runtime-addon](https://github.com/visionsofparadise/onnx-runtime-addon) native binary
+- `modelPath` — path to the `.onnx` model file
+
+Models are not bundled with the package. Each module's parameter table links to the expected model source.
+
+| Module | Model | Source |
+|--------|-------|--------|
+| DialogueIsolate | Kim_Vocal_2.onnx | [uvr_models](https://huggingface.co/seanghay/uvr_models) |
+| MusicRebalance | htdemucs.onnx + .onnx.data | [demucs](https://github.com/facebookresearch/demucs) |
+| VoiceDenoise | model_1.onnx, model_2.onnx | [DTLN](https://github.com/breizhn/DTLN) |
+
+## License
+
+ISC

package/dist/cli.js

@@ -0,0 +1,84 @@
+#!/usr/bin/env node
+import { Command } from 'commander';
+import { existsSync, readFileSync } from 'fs';
+import { resolve } from 'path';
+import { SourceNode, validateGraphDefinition, renderGraph } from 'buffered-audio-nodes-core';
+
+var program = new Command();
+program.name("buffered-audio-nodes").description("Process audio through buffered audio node pipelines");
+program.command("process").description("Run an async audio processing pipeline").requiredOption("--pipeline <file>", "TypeScript file with default SourceAsyncModule export").option("--chunk-size <samples>", "Chunk size in samples").option("--high-water-mark <count>", "Stream backpressure high water mark").action(async (options) => {
+  const pipelinePath = resolve(options.pipeline);
+  if (!existsSync(pipelinePath)) {
+    process.stderr.write(`Error: pipeline file not found: ${pipelinePath}
+`);
+    process.exit(1);
+  }
+  const { register } = await import('tsx/esm/api');
+  const unregister = register();
+  try {
+    const mod = await import(pipelinePath);
+    const source = mod.default;
+    if (!(source instanceof SourceNode)) {
+      process.stderr.write("Error: default export must be a SourceAsyncModule\n");
+      process.exit(1);
+    }
+    const chunkSize = options.chunkSize ? parseInt(options.chunkSize, 10) : void 0;
+    const highWaterMark = options.highWaterMark ? parseInt(options.highWaterMark, 10) : void 0;
+    if (chunkSize !== void 0 && (!Number.isFinite(chunkSize) || chunkSize <= 0)) {
+      process.stderr.write(`Error: --chunk-size must be a positive integer, got "${options.chunkSize}"
+`);
+      process.exit(1);
+    }
+    if (highWaterMark !== void 0 && (!Number.isFinite(highWaterMark) || highWaterMark <= 0)) {
+      process.stderr.write(`Error: --high-water-mark must be a positive integer, got "${options.highWaterMark}"
+`);
+      process.exit(1);
+    }
+    const renderOptions = {
+      chunkSize,
+      highWaterMark
+    };
+    process.stdout.write(`Processing pipeline: ${pipelinePath}
+`);
+    await source.render(renderOptions);
+    process.stdout.write("Done.\n");
+  } finally {
+    await unregister();
+  }
+});
+program.command("render").description("Render a .bag graph definition file").argument("<file>", "Path to .bag file (JSON)").option("--chunk-size <samples>", "Chunk size in samples").option("--high-water-mark <count>", "Stream backpressure high water mark").action(async (file, options) => {
+  const bagPath = resolve(file);
+  if (!existsSync(bagPath)) {
+    process.stderr.write(`Error: file not found: ${bagPath}
+`);
+    process.exit(1);
+  }
+  const json = JSON.parse(readFileSync(bagPath, "utf-8"));
+  const definition = validateGraphDefinition(json);
+  const { register } = await import('tsx/esm/api');
+  const unregister = register();
+  try {
+    const registry = /* @__PURE__ */ new Map();
+    for (const nodeDef of definition.nodes) {
+      if (!registry.has(nodeDef.package)) {
+        const mod = await import(nodeDef.package);
+        const packageMap = /* @__PURE__ */ new Map();
+        for (const [key, value] of Object.entries(mod)) {
+          if (typeof value === "function") {
+            packageMap.set(key, value);
+          }
+        }
+        registry.set(nodeDef.package, packageMap);
+      }
+    }
+    const chunkSize = options.chunkSize ? parseInt(options.chunkSize, 10) : void 0;
+    const highWaterMark = options.highWaterMark ? parseInt(options.highWaterMark, 10) : void 0;
+    process.stdout.write(`Rendering graph: ${definition.name}
+`);
+    await renderGraph(definition, registry, { chunkSize, highWaterMark });
+    process.stdout.write("Done.\n");
+  } finally {
+    await unregister();
+  }
+});
+program.parse();