8270d839bf
Following a different approach to better keep main-custom up to date with main by resetting and merging pending upstream PRs on top. Change-Id: I26c7a2c63b166715d01b50da0bd47aaf99aa3c1a Reviewed-on: https://chromium-review.googlesource.com/c/external/github.com/WebKit/JetStream/+/6976606 Reviewed-by: Victor Gomes <victorgomes@chromium.org>
63 lines
2.7 KiB
JavaScript
63 lines
2.7 KiB
JavaScript
// Copyright 2025 the V8 project authors. All rights reserved.
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// Use of this source code is governed by a BSD-style license that can be
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// found in the LICENSE file.
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// Convert an WAV audio file to a format that Transformers.js Whisper model can
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// work with.
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// Write the output as raw float values, so we don't need any additional
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// WAV/RIFF parser but can directly use it.
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// See https://huggingface.co/docs/transformers.js/guides/node-audio-processing
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import wavefile from 'wavefile';
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import * as fs from 'fs';
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if (process.argv.length < 5) {
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throw new Error("Usage: node convert-audio.mjs input.wav output.raw <start_sample> <end_sample>");
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}
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const inputFilename = process.argv[2];
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const outputFilename = process.argv[3];
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const outputStartSample = parseInt(process.argv[4]);
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const outputEndSample = parseInt(process.argv[5]);
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const wav = new wavefile.WaveFile(fs.readFileSync(inputFilename));
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const inputChannelCount = wav.fmt.numChannels;
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const inputSampleRate = wav.fmt.sampleRate;
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const inputBitDepth = wav.fmt.bitsPerSample;
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// Pipeline expects input as a Float32Array.
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wav.toBitDepth('32f');
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// Whisper expects audio with a sampling rate of 16000.
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const outputSampleRate = 16000;
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wav.toSampleRate(outputSampleRate);
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// Explicitly set output type to `Float32Array` to make the raw output smaller.
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// (Otherwise, `Float64Array` is the default.)
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let audioData = wav.getSamples(/* interleaved */ false, Float32Array);
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// Downmix to mono, if necessary.
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const outputChannelCount = 1;
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if (Array.isArray(audioData)) {
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if (audioData.length > 1) {
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const SCALING_FACTOR = Math.sqrt(2);
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for (let i = 0; i < audioData[0].length; ++i) {
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audioData[0][i] = SCALING_FACTOR * (audioData[0][i] + audioData[1][i]) / 2;
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}
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}
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audioData = audioData[0];
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}
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const inputSampleCount = audioData.length;
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// Select samples as specified by outputStartSample and outputEndSample.
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console.assert(outputStartSample < outputEndSample, "Given start is smaller than end.");
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console.assert(outputEndSample <= audioData.length, "Given end is longer than input file.");
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audioData = audioData.slice(outputStartSample, outputEndSample);
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const outputSampleCount = audioData.length;
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const outputSizeBytes = audioData.byteLength;
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fs.writeFileSync(outputFilename, audioData);
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const durationSeconds = outputSampleCount / outputSampleRate;
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console.log(`Converted ${durationSeconds}s of audio`);
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console.log(` from '${inputFilename}', ${inputChannelCount} channel(s), ${inputSampleRate} Hz, ${inputBitDepth} bit, ${inputSampleCount} samples`);
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console.log(` to '${outputFilename}', ${outputChannelCount} channel(s), ${outputSampleRate} Hz, 32 bit float, ${outputSampleCount} samples, ${outputSizeBytes} bytes`);
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