A collection of VoIP audio codecs implemented for Rust. This crate provides a unified interface for encoding and decoding various audio formats commonly used in SIP, VoIP, and WebRTC applications.
| Codec | Implementation | Feature | no_std |
|---|---|---|---|
| G.711 (PCMA/PCMU) | Pure Rust | Built-in | yes (heap-free) |
| G.722 | Pure Rust | Built-in | yes (heap-free) |
| G.729 | Pure Rust (g729-sys) |
Built-in | yes (heap-free) |
| Opus | Pure Rust (opus-rs) |
opus (on by default) |
yes (needs alloc) |
| Telephone Event | RFC 4733 | Built-in | yes (heap-free) |
| Resampler | Polyphase FIR | Built-in | yes (heap-free) |
- Unified API: Simple
EncoderandDecodertraits for all codecs. no_stdsupport: every codec runs withoutstd. G.711/G.722/G.729, the resampler and telephone-event are fully heap-free (no allocator needed); Opus needs analloc(its internal working set is ~250 KB and is boxed).- Resampler: Built-in audio resampling utility.
Measured on Apple M2 Pro (processing 20ms audio frames):
| Codec | Encode (20ms) | Decode (20ms) | Rate |
|---|---|---|---|
| PCMU | ~50.09 ns | ~59.73 ns | 8kHz |
| PCMA | ~50.23 ns | ~59.63 ns | 8kHz |
| G.722 | ~5.02 µs | ~3.82 µs | 16kHz |
| G.729 | ~20.50 µs | ~6.16 µs | 8kHz |
| Opus | ~52.34 µs | ~23.19 µs | 48kHz |
Note: Benchmarks were run with cargo bench --bench codec_bench (Criterion). create_encoder(CodecType::Opus) currently uses the default Opus profile: 48kHz, stereo, Application::Audio, bitrate 64kbps, complexity 5.
Add this to your Cargo.toml:
[dependencies]
audio-codec = "0.4" # Opus is enabled by defaultTo opt out of Opus (smaller build, no alloc needed):
[dependencies]
audio-codec = { version = "0.4", default-features = false, features = ["std"] }use audio_codec::{create_decoder, CodecType};
fn main() {
let mut decoder = create_decoder(CodecType::PCMA);
let encoded_data: Vec<u8> = vec![/* ... */];
let pcm_samples = decoder.decode(&encoded_data);
println!("Decoded {} samples", pcm_samples.len());
}use audio_codec::{create_encoder, CodecType, Encoder};
fn main() {
let mut encoder = create_encoder(CodecType::G722);
let pcm_samples: Vec<i16> = vec![0; 320]; // 20ms @ 16kHz mono
let encoded_data = encoder.encode(&pcm_samples);
println!("Encoded into {} bytes", encoded_data.len());
}use audio_codec::{
create_opus_decoder, create_opus_encoder, Decoder, Encoder,
opus::OpusApplication,
};
fn main() {
// Explicit Opus encoder/decoder creation
// 48kHz stereo, Audio mode
let mut encoder = create_opus_encoder(48_000, 2, OpusApplication::Audio);
let mut decoder = create_opus_decoder(48_000, 2);
let pcm_samples: Vec<i16> = vec![0; 960 * 2]; // 20ms @ 48kHz, interleaved stereo
let encoded_data = encoder.encode(&pcm_samples);
let decoded_pcm = decoder.decode(&encoded_data);
println!("Encoded into {} bytes", encoded_data.len());
println!("Decoded {} samples", decoded_pcm.len());
}The crate works on bare-metal targets with no std. G.711/G.722/G.729, the
resampler and telephone-event are additionally heap-free (no allocator);
Opus needs an alloc because its encoder/decoder working set (~250 KB) is
boxed. Disable the default std feature and use the slice-based *_into API:
[dependencies.audio-codec]
version = "0.4"
default-features = falseuse audio_codec::{Encoder, Decoder, CodecError, g722};
fn g722_roundtrip(
encoder: &mut g722::G722Encoder,
decoder: &mut g722::G722Decoder,
pcm: &[i16],
) -> Result<usize, CodecError> {
// Caller-provided scratch buffers — no allocation.
let mut enc_buf = [0u8; 160]; // 20ms @ 16kHz = 160 bytes
let mut dec_buf = [0i16; 320];
let n = encoder.encode_into(pcm, &mut enc_buf)?;
let m = decoder.decode_into(&enc_buf[..n], &mut dec_buf)?;
Ok(m)
}Use max_encode_bytes / max_decode_samples to size the buffers correctly:
let needed_bytes = encoder.max_encode_bytes(pcm.len());
let needed_samples = decoder.max_decode_samples(n_bytes);Resampler borrows a caller-provided coefficient buffer (~24 KB):
use audio_codec::resampler::{Resampler, COEFFS_LEN};
let mut coeffs: [f32; COEFFS_LEN] = [0.0; COEFFS_LEN];
let mut r = Resampler::new(8000, 16000, &mut coeffs)?;
let mut out = [0i16; 160];
let n = r.resample_into(&input[..80], &mut out)?;Opus also works without std (it needs an alloc — the encoder/decoder are
boxed). Use the slice-based *_into API exactly like the other codecs:
use audio_codec::{Encoder, Decoder, opus::{OpusEncoder, OpusDecoder}};
// 48 kHz, mono. The structs are boxed internally, so this only needs `alloc`.
let mut encoder = OpusEncoder::new(48_000, 1);
let mut decoder = OpusDecoder::new(48_000, 1);
let pcm: &[i16] = /* 20 ms @ 48 kHz = 960 mono samples */;
let mut packet = [0u8; 1275];
let n = encoder.encode_into(pcm, &mut packet)?;
let mut out = [0i16; 960];
let m = decoder.decode_into(&packet[..n], &mut out)?;| Codec | Status |
|---|---|
| G.711 (PCMA/PCMU), G.722, G.729, Telephone Event | ✅ fully heap-free (no allocator) |
| Resampler | ✅ fully heap-free (borrowed coeffs buffer) |
| Opus | ✅ supported (needs alloc; encoder/decoder are boxed) |
The crate is verified to compile on thumbv7em-none-eabi and other bare-metal
targets. Float math (resampler + Opus) goes through libm when std is off.
This project is licensed under the MIT License.