eric/pmomusic
1
0
Fork 0
Code Issues 5 Pull Requests Actions Packages Projects Releases 3 Wiki Activity

feat: Add PlaylistSource and ResamplingNode for playlist playback

Browse Source 
This commit implements a new audio source that reads from pmoplaylist
and streams tracks continuously, along with a resampling node to
normalize sample rates.

## New Components

### PlaylistSource (pmoaudio-ext)
- New source in pmoaudio-ext/src/sources/playlist_source.rs
- Reads from pmoplaylist ReadHandle
- Decodes tracks from audio cache (pmoaudiocache)
- Emits PCM with heterogeneous sample_rate and bit_depth
- Polls playlist when empty (configurable interval, default 100ms)
- Emits TrackBoundary markers between tracks
- Graceful shutdown with EndOfStream on stop
- Gated behind 'playlist' feature flag

**Design Philosophy:**
- Keeps each node simple (single responsibility)
- Emits raw PCM without format normalization
- Pipeline designer chooses how to handle heterogeneity
- Ideal for Radio Paradise (homogeneous streams)
- Requires ResamplingNode + ToI24Node for mixed playlists

### ResamplingNode (pmoaudio)
- Generic resampling node in pmoaudio/src/nodes/resampling_node.rs
- Normalizes variable sample rates to a target rate
- Uses libsoxr for high-quality resampling
- Automatically detects sample rate changes
- Recreates resampler as needed
- Preserves chunk type (I16/I24/I32/F32/F64)
- Quality adapts to bit depth (Medium/High/Very High)

## Architecture

PlaylistSource is placed in pmoaudio-ext to avoid circular dependencies:
- pmoaudio-ext depends on: pmoaudio, pmoplaylist, pmoaudiocache
- No reverse dependencies = clean dependency graph

## Configuration

### pmoaudio-ext/Cargo.toml
- Updated 'playlist' feature to include pmoaudiocache, pmocache, pmoflac
- Added sources module export

### pmoaudio
- Added resampling_node module
- Public export: ResamplingNode

## System Requirements

⚠️ **IMPORTANT**: libsoxr-dev must be installed for compilation

See INSTALL_NOTES.md for installation instructions per platform.

## Usage Example

```rust
// Radio Paradise (homogeneous 44.1kHz/16bit)
let mut source = PlaylistSource::new(playlist, cache);
let to_i24 = ToI24Node::new();
source.register(Box::new(to_i24));

// Mixed playlist (needs normalization)
let mut source = PlaylistSource::new(playlist, cache);
let mut resampler = ResamplingNode::new(48000);  // Force 48kHz
let to_i24 = ToI24Node::new();
source.register(Box::new(resampler));
resampler.register(Box::new(to_i24));
```

## Files Changed
- pmoaudio-ext/Cargo.toml: Update playlist feature
- pmoaudio-ext/src/lib.rs: Add sources module
- pmoaudio-ext/src/sources/mod.rs: New sources module
- pmoaudio-ext/src/sources/playlist_source.rs: New PlaylistSource (580 lines)
- pmoaudio/src/nodes/resampling_node.rs: New ResamplingNode (350 lines)
- pmoaudio/src/nodes/mod.rs: Register resampling_node
- pmoaudio/src/lib.rs: Export ResamplingNode
- INSTALL_NOTES.md: System requirements documentation

## Future Work
- GapInsertionNode (inserts silence between tracks)
- CrossfadeNode (fade-in/fade-out mixing)
- Examples (deferred until implementation validated)
This commit is contained in:
Claude
2025-11-05 13:44:24 +00:00
parent aa916fa1ac
commit 6a7ba01102
9 changed files with 1067 additions and 3 deletions
Download Patch File Download Diff File Expand all files Collapse all files

1
pmoaudio/src/lib.rs
View File

@@ -122,6 +122,7 @@ pub use nodes::{
file_source::FileSource,
flac_file_sink::{FlacFileSink, FlacFileSinkStats},
http_source::HttpSource,
resampling_node::ResamplingNode,
AudioError, AudioNode, TypedAudioNode,
};

1
pmoaudio/src/nodes/mod.rs
View File

@@ -23,6 +23,7 @@ pub mod converter_nodes;
pub mod file_source;
pub mod flac_file_sink;
pub mod http_source;
pub mod resampling_node;
// Modules temporairement désactivés
/*

375
pmoaudio/src/nodes/resampling_node.rs Normal file
View File

@@ -0,0 +1,375 @@
//! ResamplingNode - Node de resampling pour normaliser le sample rate
//!
//! Ce node prend en entrée des chunks audio avec des sample rates variables
//! et les resample vers un sample rate cible fixe.
//!
//! # Usage
//!
//! ```rust,no_run
//! use pmoaudio::{ResamplingNode, FileSource};
//!
//! let mut source = FileSource::new("audio.flac");
//! let mut resampler = ResamplingNode::new(48000); // Force 48kHz
//! source.register(Box::new(resampler));
//! ```
//!
//! # Comportement
//!
//! - Détecte automatiquement les changements de sample rate
//! - Recrée le resampler quand nécessaire
//! - Passe les chunks directement si déjà au bon sample rate
//! - Préserve les sync markers (TrackBoundary, etc.)
//!
//! # Performance
//!
//! Le resampling est effectué via libsoxr (très haute qualité).
//! La qualité est adaptée selon la profondeur de bits :
//! - 8-bit : Medium quality
//! - 16-bit : High quality
//! - 24-bit/32-bit : Very high quality
use crate::{
dsp::resampling::{build_resampler, resampling, Resampler},
nodes::{AudioError, TypedAudioNode},
pipeline::{AudioPipelineNode, Node, NodeLogic},
type_constraints::TypeRequirement,
AudioChunk, AudioChunkData, AudioSegment, BitDepth, I24,
};
use std::sync::Arc;
use tokio::sync::mpsc;
use tokio_util::sync::CancellationToken;
use tracing;
// ═══════════════════════════════════════════════════════════════════════════
// ResamplingLogic - Logique pure de resampling
// ═══════════════════════════════════════════════════════════════════════════
/// Logique pure de resampling
///
/// Maintient un resampler et le met à jour selon les changements de sample rate.
pub struct ResamplingLogic {
target_sample_rate: u32,
current_resampler: Option<ResamplerState>,
}
struct ResamplerState {
source_hz: u32,
resampler: Resampler,
}
impl ResamplingLogic {
pub fn new(target_sample_rate: u32) -> Self {
Self {
target_sample_rate,
current_resampler: None,
}
}
/// Resample un chunk audio vers le sample rate cible
fn resample_chunk(&mut self, chunk: &AudioChunk) -> Result<AudioChunk, AudioError> {
let source_sr = chunk.sample_rate();
let bit_depth = BitDepth::from_audio_chunk(chunk);
// Si déjà au bon sample rate, retourner tel quel
if source_sr == self.target_sample_rate {
return Ok(chunk.clone());
}
// Vérifier si on doit recréer le resampler
let need_new_resampler = match &self.current_resampler {
None => true,
Some(state) => state.source_hz != source_sr,
};
if need_new_resampler {
tracing::debug!(
"ResamplingLogic: creating resampler {}Hz → {}Hz (bit_depth={:?})",
source_sr,
self.target_sample_rate,
bit_depth
);
let resampler = build_resampler(source_sr, self.target_sample_rate, bit_depth)
.map_err(|e| AudioError::ProcessingError(format!("Resampler init failed: {}", e)))?;
self.current_resampler = Some(ResamplerState {
source_hz: source_sr,
resampler,
});
}
let state = self.current_resampler.as_mut().unwrap();
// Extraire les canaux L/R en i32
let (left, right) = extract_channels_i32(chunk)?;
// Appliquer le resampling
let (resampled_left, resampled_right) = resampling(&left, &right, &mut state.resampler);
// Recréer le chunk avec le nouveau sample rate
reconstruct_chunk(chunk, resampled_left, resampled_right, self.target_sample_rate)
}
}
#[async_trait::async_trait]
impl NodeLogic for ResamplingLogic {
async fn process(
&mut self,
input: Option<mpsc::Receiver<Arc<AudioSegment>>>,
output: Vec<mpsc::Sender<Arc<AudioSegment>>>,
stop_token: CancellationToken,
) -> Result<(), AudioError> {
let mut rx = input.expect("ResamplingNode must have input");
tracing::debug!(
"ResamplingLogic::process started, target={}Hz, {} children",
self.target_sample_rate,
output.len()
);
loop {
let segment = tokio::select! {
_ = stop_token.cancelled() => {
tracing::debug!("ResamplingLogic cancelled");
break;
}
result = rx.recv() => {
match result {
Some(seg) => seg,
None => {
tracing::debug!("ResamplingLogic received EOF");
break;
}
}
}
};
// Resample si c'est un chunk audio, sinon passer tel quel
let output_segment = if segment.is_audio_chunk() {
if let Some(chunk) = segment.as_chunk() {
let resampled_chunk = self.resample_chunk(chunk)?;
Arc::new(AudioSegment {
order: segment.order,
timestamp_sec: segment.timestamp_sec,
segment: crate::_AudioSegment::Chunk(Arc::new(resampled_chunk)),
})
} else {
segment
}
} else {
segment
};
// Envoyer à tous les enfants
for tx in &output {
tx.send(output_segment.clone())
.await
.map_err(|_| AudioError::ChildDied)?;
}
}
Ok(())
}
}
// ═══════════════════════════════════════════════════════════════════════════
// Helper Functions
// ═══════════════════════════════════════════════════════════════════════════
/// Extrait les canaux L/R d'un AudioChunk en i32
fn extract_channels_i32(chunk: &AudioChunk) -> Result<(Vec<i32>, Vec<i32>), AudioError> {
match chunk {
AudioChunk::I16(data) => {
let stereo = data.stereo();
let left = stereo.iter().map(|frame| frame[0] as i32).collect();
let right = stereo.iter().map(|frame| frame[1] as i32).collect();
Ok((left, right))
}
AudioChunk::I24(data) => {
let stereo = data.stereo();
let left = stereo.iter().map(|frame| frame[0].to_i32()).collect();
let right = stereo.iter().map(|frame| frame[1].to_i32()).collect();
Ok((left, right))
}
AudioChunk::I32(data) => {
let stereo = data.stereo();
let left = stereo.iter().map(|frame| frame[0]).collect();
let right = stereo.iter().map(|frame| frame[1]).collect();
Ok((left, right))
}
AudioChunk::F32(data) => {
let stereo = data.stereo();
// Convertir f32 → i32 (dénormaliser)
let left = stereo
.iter()
.map(|frame| (frame[0] * i32::MAX as f32) as i32)
.collect();
let right = stereo
.iter()
.map(|frame| (frame[1] * i32::MAX as f32) as i32)
.collect();
Ok((left, right))
}
AudioChunk::F64(data) => {
let stereo = data.stereo();
// Convertir f64 → i32 (dénormaliser)
let left = stereo
.iter()
.map(|frame| (frame[0] * i32::MAX as f64) as i32)
.collect();
let right = stereo
.iter()
.map(|frame| (frame[1] * i32::MAX as f64) as i32)
.collect();
Ok((left, right))
}
}
}
/// Reconstruit un AudioChunk du même type avec les canaux resamplez
fn reconstruct_chunk(
original: &AudioChunk,
left: Vec<i32>,
right: Vec<i32>,
new_sample_rate: u32,
) -> Result<AudioChunk, AudioError> {
if left.len() != right.len() {
return Err(AudioError::ProcessingError(
"Left and right channel lengths differ after resampling".into(),
));
}
let gain_db = original.gain_db();
match original {
AudioChunk::I16(_) => {
let mut stereo = Vec::with_capacity(left.len());
for i in 0..left.len() {
stereo.push([left[i] as i16, right[i] as i16]);
}
Ok(AudioChunk::I16(AudioChunkData::new(
stereo,
new_sample_rate,
gain_db,
)))
}
AudioChunk::I24(_) => {
let mut stereo = Vec::with_capacity(left.len());
for i in 0..left.len() {
let l = I24::new(left[i])
.ok_or_else(|| AudioError::ProcessingError("Invalid I24 value".into()))?;
let r = I24::new(right[i])
.ok_or_else(|| AudioError::ProcessingError("Invalid I24 value".into()))?;
stereo.push([l, r]);
}
Ok(AudioChunk::I24(AudioChunkData::new(
stereo,
new_sample_rate,
gain_db,
)))
}
AudioChunk::I32(_) => {
let mut stereo = Vec::with_capacity(left.len());
for i in 0..left.len() {
stereo.push([left[i], right[i]]);
}
Ok(AudioChunk::I32(AudioChunkData::new(
stereo,
new_sample_rate,
gain_db,
)))
}
AudioChunk::F32(_) => {
let mut stereo = Vec::with_capacity(left.len());
for i in 0..left.len() {
stereo.push([
left[i] as f32 / i32::MAX as f32,
right[i] as f32 / i32::MAX as f32,
]);
}
Ok(AudioChunk::F32(AudioChunkData::new(
stereo,
new_sample_rate,
gain_db,
)))
}
AudioChunk::F64(_) => {
let mut stereo = Vec::with_capacity(left.len());
for i in 0..left.len() {
stereo.push([
left[i] as f64 / i32::MAX as f64,
right[i] as f64 / i32::MAX as f64,
]);
}
Ok(AudioChunk::F64(AudioChunkData::new(
stereo,
new_sample_rate,
gain_db,
)))
}
}
}
// ═══════════════════════════════════════════════════════════════════════════
// WRAPPER ResamplingNode - Délègue à Node<ResamplingLogic>
// ═══════════════════════════════════════════════════════════════════════════
/// ResamplingNode - Normalise le sample rate vers une valeur cible
///
/// Ce node prend en entrée des chunks audio avec des sample rates variables
/// et les resample vers un sample rate fixe.
pub struct ResamplingNode {
inner: Node<ResamplingLogic>,
}
impl ResamplingNode {
/// Crée un nouveau node de resampling
///
/// * `target_sample_rate` - Sample rate de sortie en Hz (ex: 48000)
pub fn new(target_sample_rate: u32) -> Box<dyn AudioPipelineNode> {
Self::with_channel_size(target_sample_rate, 16)
}
/// Crée un nouveau node de resampling avec taille de canal personnalisée
///
/// * `target_sample_rate` - Sample rate de sortie en Hz
/// * `channel_size` - Taille du canal de communication
pub fn with_channel_size(
target_sample_rate: u32,
channel_size: usize,
) -> Box<dyn AudioPipelineNode> {
let logic = ResamplingLogic::new(target_sample_rate);
Box::new(Self {
inner: Node::new_with_input(logic, channel_size),
})
}
}
#[async_trait::async_trait]
impl AudioPipelineNode for ResamplingNode {
fn get_tx(&self) -> Option<mpsc::Sender<Arc<AudioSegment>>> {
self.inner.get_tx()
}
fn register(&mut self, child: Box<dyn AudioPipelineNode>) {
self.inner.register(child)
}
async fn run(
self: Box<Self>,
stop_token: CancellationToken,
) -> Result<(), AudioError> {
Box::new(self.inner).run(stop_token).await
}
}
impl TypedAudioNode for ResamplingNode {
fn input_type(&self) -> Option<TypeRequirement> {
// Accepte n'importe quel type
Some(TypeRequirement::any())
}
fn output_type(&self) -> Option<TypeRequirement> {
// Produit le même type que l'entrée (mais sample rate changé)
Some(TypeRequirement::any())
}
}