evanwashere/opus

fast opus bindings for node and browsers

fast opus bindings for node and browsers

bun add @evan/opus

npm install @evan/opus

import { Encoder, Decoder } from '@evan/opus';

const e = new Encoder({ channels: 2, sample_rate: 48_000 });
const d = new Decoder({ channels: 2, sample_rate: 48_000 });

d.decode(e.encode(pcm));

supported platforms

	node@10	node@12	node@14	node@16	node@18
wasm32	✕	✓	✓	✓	✓
macos x64	✓	✓	✓	✓	✓
macos arm64	✓	✓	✓	✓	✓
windows x64	✓	✓	✓	✓	✓
linux x64 gnu	✓	✓	✓	✓	✓
linux arm64 gnu	✓	✓	✓	✓	✓

you can force usage of wasm by setting OPUS_FORCE_WASM env variable

for deno and browsers use @evan/wasm/target/opus/deno.js from npm/cdn

benchmarks

License

MIT © Evan

opus's Issues

Async encode/decode, and encoding an AudioBuffer

Hello, thank you for creating this library. I had some questions about how to use it.

From what I understand in wasm/index.js, the encode/decode functions are synchronous and they block the main thread in browser/node. Is that true?

If so, is it possible to load this library from a worker, to make it async/non-blocking? I'll see if I can achieve it and report back here if successful.

Looking at other similar libraries, wasm-media-encoders, wasm-audio-decoders, and opusscript, I see their API is synchronous and blocking also. ffmpeg.wasm and lame-wasm are async.

That's interesting because I would have expected, for most purposes, async encode/decode is preferred for both browser and server side, so as not to block the main thread. For example, how AudioContext.decodeAudioData() works.

---

Currently, I'm encountering some difficulties trying to encode (and decode back) an instance of AudioBuffer.

First, I noticed that if the sample rate is 44100, the Encoder/Decoder constructor throws an error, opus: invalid argument.

const { encode } = new Encoder({
  channels: audioBuffer.numberOfChannels,
  sample_rate: audioBuffer.sampleRate
})

If I force it to 48000, it doesn't throw.

Next, I learned how to extract PCM data from an AudioBuffer using getChannelData().

const buffers = [] // Float32Array per channel

for (let i=0, len=audioBuffer.numberOfChannels; i < len; i++) {
  buffers.push(audioBuffer.getChannelData(i))
}

When I pass a channel's data to encode(), it throws RangeError: source array is too long.

The error is from the first line of RawEncoder.encode() in @evan/wasm/target/opus/deno.js.

encode(buffer) {
    bptrs.set(buffer = new Uint8Array(buffer.buffer, buffer.byteOffset, buffer.byteLength));
    return pptrs.slice(0, err(wasm.opus_encode(this.#ptr, bptr, buffer.length / 2 / this.channels, pptr, pptrl)));
}

..So I should convert my Float32Array to an Uint8Array before passing it to encode. Is that correct?

const channelData = buffers[0].buffer // Get ArrayBuffer from Float32Array
const pcm = new Uint8Array(channelData)
const opusData = encode(pcm)

The conversion seems to work, but still results in the same error, RangeError: source array is too long.

EDIT: I'm guessing it's because I set the wrong sample rate when instantiating the encoder. And probably it's impossible (or unwise, haha) to try to up-scale from 44.1 to 48 kHz.

---

Well, I'll try to figure out these questions on my own, but if you have any thoughts, I'd appreciate it.