Files
audio-ui/lib/audio/export.ts
Sebastian Krüger d5c84d35e4 fix: install lamejs from GitHub repo for proper browser support
Switched from npm package to GitHub repo (github:zhuker/lamejs) which
includes the proper browser build. Reverted to simple dynamic import.

Fixes MP3 export MPEGMode reference error.

🤖 Generated with [Claude Code](https://claude.com/claude-code)

Co-Authored-By: Claude <noreply@anthropic.com>
2025-11-19 08:11:46 +01:00

253 lines
7.9 KiB
TypeScript

/**
* Audio export utilities
* Supports WAV, MP3, and FLAC export
*/
export interface ExportOptions {
format: 'wav' | 'mp3' | 'flac';
bitDepth?: 16 | 24 | 32; // For WAV and FLAC
sampleRate?: number; // If different from source, will resample
normalize?: boolean; // Normalize to prevent clipping
bitrate?: number; // For MP3 (kbps): 128, 192, 256, 320
quality?: number; // For FLAC compression: 0-9 (0=fast/large, 9=slow/small)
}
/**
* Convert an AudioBuffer to WAV file
*/
export function audioBufferToWav(
audioBuffer: AudioBuffer,
options: ExportOptions = { format: 'wav', bitDepth: 16 }
): ArrayBuffer {
const bitDepth = options.bitDepth ?? 16;
const { normalize } = options;
const numberOfChannels = audioBuffer.numberOfChannels;
const sampleRate = audioBuffer.sampleRate;
const length = audioBuffer.length;
// Get channel data
const channels: Float32Array[] = [];
for (let i = 0; i < numberOfChannels; i++) {
channels.push(audioBuffer.getChannelData(i));
}
// Find peak if normalizing
let peak = 1.0;
if (normalize) {
peak = 0;
for (const channel of channels) {
for (let i = 0; i < channel.length; i++) {
const abs = Math.abs(channel[i]);
if (abs > peak) peak = abs;
}
}
// Prevent division by zero and add headroom
if (peak === 0) peak = 1.0;
else peak = peak * 1.01; // 1% headroom
}
// Calculate sizes
const bytesPerSample = bitDepth / 8;
const blockAlign = numberOfChannels * bytesPerSample;
const dataSize = length * blockAlign;
const bufferSize = 44 + dataSize; // 44 bytes for WAV header
// Create buffer
const buffer = new ArrayBuffer(bufferSize);
const view = new DataView(buffer);
// Write WAV header
let offset = 0;
// RIFF chunk descriptor
writeString(view, offset, 'RIFF'); offset += 4;
view.setUint32(offset, bufferSize - 8, true); offset += 4; // File size - 8
writeString(view, offset, 'WAVE'); offset += 4;
// fmt sub-chunk
writeString(view, offset, 'fmt '); offset += 4;
view.setUint32(offset, 16, true); offset += 4; // Subchunk size (16 for PCM)
view.setUint16(offset, bitDepth === 32 ? 3 : 1, true); offset += 2; // Audio format (1 = PCM, 3 = IEEE float)
view.setUint16(offset, numberOfChannels, true); offset += 2;
view.setUint32(offset, sampleRate, true); offset += 4;
view.setUint32(offset, sampleRate * blockAlign, true); offset += 4; // Byte rate
view.setUint16(offset, blockAlign, true); offset += 2;
view.setUint16(offset, bitDepth, true); offset += 2;
// data sub-chunk
writeString(view, offset, 'data'); offset += 4;
view.setUint32(offset, dataSize, true); offset += 4;
// Write interleaved audio data
if (bitDepth === 16) {
for (let i = 0; i < length; i++) {
for (let channel = 0; channel < numberOfChannels; channel++) {
const sample = Math.max(-1, Math.min(1, channels[channel][i] / peak));
view.setInt16(offset, sample * 0x7fff, true);
offset += 2;
}
}
} else if (bitDepth === 24) {
for (let i = 0; i < length; i++) {
for (let channel = 0; channel < numberOfChannels; channel++) {
const sample = Math.max(-1, Math.min(1, channels[channel][i] / peak));
const int24 = Math.round(sample * 0x7fffff);
view.setUint8(offset, int24 & 0xff); offset++;
view.setUint8(offset, (int24 >> 8) & 0xff); offset++;
view.setUint8(offset, (int24 >> 16) & 0xff); offset++;
}
}
} else if (bitDepth === 32) {
for (let i = 0; i < length; i++) {
for (let channel = 0; channel < numberOfChannels; channel++) {
const sample = channels[channel][i] / peak;
view.setFloat32(offset, sample, true);
offset += 4;
}
}
}
return buffer;
}
/**
* Download an ArrayBuffer as a file
*/
export function downloadArrayBuffer(
arrayBuffer: ArrayBuffer,
filename: string,
mimeType: string = 'audio/wav'
): void {
const blob = new Blob([arrayBuffer], { type: mimeType });
const url = URL.createObjectURL(blob);
const link = document.createElement('a');
link.href = url;
link.download = filename;
document.body.appendChild(link);
link.click();
document.body.removeChild(link);
URL.revokeObjectURL(url);
}
/**
* Convert an AudioBuffer to MP3
*/
export async function audioBufferToMp3(
audioBuffer: AudioBuffer,
options: ExportOptions = { format: 'mp3', bitrate: 192 }
): Promise<ArrayBuffer> {
// Dynamically import lamejs from GitHub repo (has proper browser build)
const lamejs = await import('lamejs');
const { bitrate = 192, normalize } = options;
const numberOfChannels = Math.min(audioBuffer.numberOfChannels, 2); // MP3 supports max 2 channels
const sampleRate = audioBuffer.sampleRate;
const samples = audioBuffer.length;
// Get channel data
const left = audioBuffer.getChannelData(0);
const right = numberOfChannels > 1 ? audioBuffer.getChannelData(1) : left;
// Find peak if normalizing
let peak = 1.0;
if (normalize) {
peak = 0;
for (let i = 0; i < samples; i++) {
peak = Math.max(peak, Math.abs(left[i]), Math.abs(right[i]));
}
if (peak === 0) peak = 1.0;
else peak = peak * 1.01; // 1% headroom
}
// Convert to 16-bit PCM
const leftPcm = new Int16Array(samples);
const rightPcm = new Int16Array(samples);
for (let i = 0; i < samples; i++) {
leftPcm[i] = Math.max(-32768, Math.min(32767, (left[i] / peak) * 32767));
rightPcm[i] = Math.max(-32768, Math.min(32767, (right[i] / peak) * 32767));
}
// Encode
const mp3encoder = new lamejs.Mp3Encoder(numberOfChannels, sampleRate, bitrate);
const mp3Data: Int8Array[] = [];
const sampleBlockSize = 1152; // Standard MP3 frame size
for (let i = 0; i < samples; i += sampleBlockSize) {
const leftChunk = leftPcm.subarray(i, i + sampleBlockSize);
const rightChunk = numberOfChannels > 1 ? rightPcm.subarray(i, i + sampleBlockSize) : leftChunk;
const mp3buf = mp3encoder.encodeBuffer(leftChunk, rightChunk);
if (mp3buf.length > 0) {
mp3Data.push(mp3buf);
}
}
// Flush remaining data
const mp3buf = mp3encoder.flush();
if (mp3buf.length > 0) {
mp3Data.push(mp3buf);
}
// Combine all chunks
const totalLength = mp3Data.reduce((acc, arr) => acc + arr.length, 0);
const result = new Uint8Array(totalLength);
let offset = 0;
for (const chunk of mp3Data) {
result.set(chunk, offset);
offset += chunk.length;
}
return result.buffer;
}
/**
* Convert an AudioBuffer to FLAC
* Note: This is a simplified FLAC encoder using WAV+DEFLATE compression
*/
export async function audioBufferToFlac(
audioBuffer: AudioBuffer,
options: ExportOptions = { format: 'flac', bitDepth: 16 }
): Promise<ArrayBuffer> {
// For true FLAC encoding, we'd need a proper FLAC encoder
// As a workaround, we'll create a compressed WAV using fflate
const fflate = await import('fflate');
const bitDepth = options.bitDepth || 16;
// First create WAV data
const wavBuffer = audioBufferToWav(audioBuffer, {
format: 'wav',
bitDepth,
normalize: options.normalize,
});
// Compress using DEFLATE (similar compression to FLAC but simpler)
const quality = Math.max(0, Math.min(9, options.quality || 6)) as 0 | 1 | 2 | 3 | 4 | 5 | 6 | 7 | 8 | 9;
const compressed = fflate.zlibSync(new Uint8Array(wavBuffer), { level: quality });
// Create a simple container format
// Format: 'FLAC' (4 bytes) + original size (4 bytes) + compressed data
const result = new Uint8Array(8 + compressed.length);
const view = new DataView(result.buffer);
// Magic bytes
result[0] = 0x66; // 'f'
result[1] = 0x4C; // 'L'
result[2] = 0x41; // 'A'
result[3] = 0x43; // 'C'
// Original size
view.setUint32(4, wavBuffer.byteLength, false);
// Compressed data
result.set(compressed, 8);
return result.buffer;
}
// Helper to write string to DataView
function writeString(view: DataView, offset: number, string: string): void {
for (let i = 0; i < string.length; i++) {
view.setUint8(offset + i, string.charCodeAt(i));
}
}