audio_decode 0.2.0

Decode MP3 and Ogg Vorbis to raw PCM in Dart, over FFI. Compiles stb_vorbis and minimp3 from source: self-contained, pure-Dart friendly, no platform plugins or prebuilt binaries.

audio_decode

Native Ogg Vorbis and MP3 decoding to raw PCM for Dart, over FFI. The C decoders are compiled from source by a Dart build hook, so the package is self-contained: no platform plugins, no bundled binary, and no system library to install beyond a C toolchain.

The same code path runs in pure Dart (command-line tools, servers, tests) and in Flutter, and it decodes the same bytes to the same samples on every platform. That makes it a good fit for waveform rendering, audio analysis, resampling, machine-learning preprocessing, servers and games.

It is built on two well-known public-domain single-file libraries:

• Ogg Vorbis: stb_vorbis by Sean Barrett.
• MP3: minimp3 by lieff.

What this is not

This is a decoder, not a player. It turns encoded bytes into PCM samples; it does not open audio devices or handle playback, streaming or seeking. For playback, use a player package such as just_audio or flutter_sound.

Scope:

• Decodes Ogg Vorbis and MP3 to interleaved signed 16-bit PCM.
• Encodes PCM back to a 16-bit WAV file (encodeWav), so decoded audio can be saved or handed to other tools.
• No encoding to Vorbis or MP3, and no other container or codec.

Quick start

import 'dart:io';
import 'package:audio_decode/audio_decode.dart';

void main() {
  final bytes = File('clip.mp3').readAsBytesSync();

  // Auto-detects Ogg Vorbis vs MP3 from the bytes.
  final pcm = decodeAudio(bytes);
  print('${pcm.sampleRate} Hz, ${pcm.channels} ch, ${pcm.duration}');
  print('${pcm.frameCount} frames, ${pcm.samples.length} interleaved samples');

  // Save the decoded audio as a WAV.
  File('clip.wav').writeAsBytesSync(encodeWav(pcm));
}

Decode a specific format directly when you already know it:

final ogg = decodeOgg(await File('clip.ogg').readAsBytes());
final mp3 = decodeMp3(await File('clip.mp3').readAsBytes());

API

• PcmAudio holds the result: sampleRate, channels, and samples (an Int16List of samples interleaved by channel). It exposes frameCount and duration.
• decodeAudio(Uint8List) sniffs the format and dispatches.
• decodeOgg(Uint8List) and decodeMp3(Uint8List) decode a known format.
• detectFormat(Uint8List) returns AudioFormat.ogg, AudioFormat.mp3 or AudioFormat.unknown.
• encodeWav(PcmAudio) returns a canonical 16-bit PCM WAV as Uint8List.
• audioInfo(Uint8List) returns an AudioInfo with sampleRate, channels, frameCount and duration without decoding to PCM; oggInfo and mp3Info do the same for a known format. See below.

Empty input throws ArgumentError. Bytes that are not decodable audio throw AudioDecodeException.

Samples are copied out of native memory before each call returns, so there is no native buffer for the caller to manage.

Duration without decoding

Showing track lengths in a playlist, validating an upload, or picking which files to process only needs a file's shape, not its samples. Decoding for that is expensive: four minutes of 44.1 kHz stereo is 40 MB of PCM, and you throw all of it away to read one number.

audioInfo answers from the stream itself and allocates no PCM at all:

final info = audioInfo(await File('track.mp3').readAsBytes());
print('${info.duration} at ${info.sampleRate} Hz, ${info.channels} ch');

It reports exactly what a full decode would, which the tests check against decodeAudio for every fixture. Measured on a one-second stereo fixture, warmed up and averaged (Apple M-series):

Format	audioInfo	decodeAudio
Ogg Vorbis	107 µs	511 µs
MP3	0.9 µs	217 µs

The two formats differ because of what each has to do. Vorbis stores its length in the container, so stb_vorbis opens the stream and reads it. MP3 has no total-length field, so the frame headers still have to be walked; what is skipped is the decoding and the PCM buffer, which is where the time goes.

A note on MP3 length

MP3 is a lossy frame format with built-in encoder and decoder delay, so a decoded MP3 is usually a little longer than the original audio, on the order of a thousand samples per channel. Ogg Vorbis decodes to a length very close to the source. If you need exact-length output, trim to the duration you expect.

Performance

Decoding runs entirely in native code. On an Apple M-series laptop, a 30-second stereo 44100 Hz Ogg Vorbis tone (2.6M interleaved samples) decodes in about 12 ms median, roughly 215 million samples per second. Run dart run bench/bench.dart to measure on your machine; it generates the clip with ffmpeg if ffmpeg is installed and otherwise decodes the committed one-second fixture. These are synthetic-tone numbers; a dense music track decodes more slowly.

Platforms

The build hook compiles the vendored C with the toolchain that package:native_toolchain_c drives (clang, gcc or MSVC). It targets the platforms Dart's native build hooks support: Linux, macOS and Windows on the Dart VM today, and Flutter as build-hook support there stabilises. Dart 3.9 or newer is required.

Credits and license

• stb_vorbis by Sean Barrett, public domain.
• minimp3 by lieff, public domain (CC0).

This package's own code is under the license in LICENSE, which also reproduces the upstream dedications for the vendored decoders.