pmomusic/pmoaudio/src/nodes/resampling_node.rs

//! 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 = match chunk {
            AudioChunk::I16(_) => BitDepth::B16,
            AudioChunk::I24(_) => BitDepth::B24,
            AudioChunk::I32(_) => BitDepth::B32,
            AudioChunk::F32(_) => BitDepth::B32, // Traiter comme 32-bit
            AudioChunk::F64(_) => BitDepth::B32, // Traiter comme 32-bit
        };

        // 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 frames = data.get_frames();
            let left = frames.iter().map(|frame| frame[0] as i32).collect();
            let right = frames.iter().map(|frame| frame[1] as i32).collect();
            Ok((left, right))
        }
        AudioChunk::I24(data) => {
            let frames = data.get_frames();
            let left = frames.iter().map(|frame| frame[0].as_i32()).collect();
            let right = frames.iter().map(|frame| frame[1].as_i32()).collect();
            Ok((left, right))
        }
        AudioChunk::I32(data) => {
            let frames = data.get_frames();
            let left = frames.iter().map(|frame| frame[0]).collect();
            let right = frames.iter().map(|frame| frame[1]).collect();
            Ok((left, right))
        }
        AudioChunk::F32(data) => {
            let frames = data.get_frames();
            // Convertir f32 → i32 (dénormaliser)
            let left = frames
                .iter()
                .map(|frame| (frame[0] * i32::MAX as f32) as i32)
                .collect();
            let right = frames
                .iter()
                .map(|frame| (frame[1] * i32::MAX as f32) as i32)
                .collect();
            Ok((left, right))
        }
        AudioChunk::F64(data) => {
            let frames = data.get_frames();
            // Convertir f64 → i32 (dénormaliser)
            let left = frames
                .iter()
                .map(|frame| (frame[0] * i32::MAX as f64) as i32)
                .collect();
            let right = frames
                .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())
    }
}

#[cfg(test)]
mod tests {
    use super::*;
    use crate::{AudioChunk, AudioChunkData, SyncMarker};

    #[test]
    fn test_extract_channels_i16() {
        let chunk = AudioChunk::I16(AudioChunkData::new(
            vec![[100, 200], [300, 400]],
            48000,
            0.0,
        ));

        let (left, right) = extract_channels_i32(&chunk).unwrap();

        assert_eq!(left, vec![100i32, 300i32]);
        assert_eq!(right, vec![200i32, 400i32]);
    }

    #[test]
    fn test_extract_channels_i24() {
        let chunk = AudioChunk::I24(AudioChunkData::new(
            vec![
                [I24::new(1_000_000).unwrap(), I24::new(2_000_000).unwrap()],
                [I24::new(3_000_000).unwrap(), I24::new(4_000_000).unwrap()],
            ],
            48000,
            0.0,
        ));

        let (left, right) = extract_channels_i32(&chunk).unwrap();

        assert_eq!(left, vec![1_000_000i32, 3_000_000i32]);
        assert_eq!(right, vec![2_000_000i32, 4_000_000i32]);
    }

    #[test]
    fn test_reconstruct_chunk_i16() {
        let original = AudioChunk::I16(AudioChunkData::new(
            vec![[100, 200]],
            44100,
            0.0,
        ));

        let left = vec![100i32, 300i32];
        let right = vec![200i32, 400i32];

        let result = reconstruct_chunk(&original, left, right, 48000).unwrap();

        if let AudioChunk::I16(data) = result {
            assert_eq!(data.get_sample_rate(), 48000);
            let frames = data.get_frames();
            assert_eq!(frames.len(), 2);
            assert_eq!(frames[0], [100i16, 200i16]);
            assert_eq!(frames[1], [300i16, 400i16]);
        } else {
            panic!("Expected I16 chunk");
        }
    }

    #[test]
    fn test_reconstruct_chunk_i24() {
        let original = AudioChunk::I24(AudioChunkData::new(
            vec![[I24::new(1_000_000).unwrap(), I24::new(2_000_000).unwrap()]],
            44100,
            0.0,
        ));

        let left = vec![1_000_000i32, 3_000_000i32];
        let right = vec![2_000_000i32, 4_000_000i32];

        let result = reconstruct_chunk(&original, left, right, 48000).unwrap();

        if let AudioChunk::I24(data) = result {
            assert_eq!(data.get_sample_rate(), 48000);
            let frames = data.get_frames();
            assert_eq!(frames.len(), 2);
            assert_eq!(frames[0][0].as_i32(), 1_000_000);
            assert_eq!(frames[0][1].as_i32(), 2_000_000);
        } else {
            panic!("Expected I24 chunk");
        }
    }

    #[test]
    fn test_resample_chunk_no_change_if_same_rate() {
        let mut logic = ResamplingLogic::new(48000);

        let chunk = AudioChunk::I16(AudioChunkData::new(
            vec![[100, 200], [300, 400]],
            48000, // Déjà à 48kHz
            0.0,
        ));

        let result = logic.resample_chunk(&chunk).unwrap();

        // Doit retourner le même chunk sans resampling
        if let AudioChunk::I16(data) = result {
            assert_eq!(data.get_sample_rate(), 48000);
            assert_eq!(data.get_frames().len(), 2);
        } else {
            panic!("Expected I16 chunk");
        }
    }

    #[tokio::test]
    async fn test_resampling_logic_passes_sync_markers() {
        let mut logic = ResamplingLogic::new(48000);

        let (input_tx, input_rx) = mpsc::channel(10);
        let (output_tx, mut output_rx) = mpsc::channel(10);
        let stop_token = CancellationToken::new();

        // Créer un TrackBoundary
        let metadata = Arc::new(tokio::sync::RwLock::new(
            pmometadata::MemoryTrackMetadata::new()
        ));
        let boundary = AudioSegment::new_track_boundary(0, 0.0, metadata);

        // Envoyer le boundary
        input_tx.send(boundary.clone()).await.unwrap();
        drop(input_tx);

        // Lancer le traitement
        tokio::spawn(async move {
            logic
                .process(Some(input_rx), vec![output_tx], stop_token)
                .await
                .unwrap();
        });

        // Vérifier que le boundary passe tel quel
        let result = output_rx.recv().await.unwrap();
        assert!(result.as_sync_marker().is_some());

        if let Some(marker) = result.as_sync_marker() {
            assert!(matches!(**marker, SyncMarker::TrackBoundary { .. }));
        }
    }

    #[tokio::test]
    async fn test_resampling_node_integration() {
        // Test d'intégration complet avec ResamplingNode
        let (input_tx, input_rx) = mpsc::channel(10);
        let (output_tx, mut output_rx) = mpsc::channel(10);
        let stop_token = CancellationToken::new();

        let mut logic = ResamplingLogic::new(48000);

        // Créer un chunk à 44.1kHz
        let chunk_44k = AudioChunk::I16(AudioChunkData::new(
            vec![[1000, 2000]; 100], // 100 frames
            44100,
            0.0,
        ));

        let segment = Arc::new(AudioSegment {
            order: 0,
            timestamp_sec: 0.0,
            segment: crate::_AudioSegment::Chunk(Arc::new(chunk_44k)),
        });

        input_tx.send(segment).await.unwrap();
        drop(input_tx);

        // Lancer le traitement
        tokio::spawn(async move {
            logic
                .process(Some(input_rx), vec![output_tx], stop_token)
                .await
                .unwrap();
        });

        // Vérifier le résultat
        let result = output_rx.recv().await.unwrap();
        assert!(result.is_audio_chunk());

        if let Some(chunk) = result.as_chunk() {
            // Le chunk doit être I16 (même type)
            assert!(matches!(chunk.as_ref(), AudioChunk::I16(_)));

            // Le sample rate doit être 48000
            assert_eq!(chunk.sample_rate(), 48000);

            // Le nombre de frames doit avoir changé (ratio ~1.088)
            // 100 frames @ 44.1kHz ≈ 109 frames @ 48kHz
            if let AudioChunk::I16(data) = chunk.as_ref() {
                let frames = data.get_frames().len();
                assert!(frames >= 105 && frames <= 115, "Expected ~109 frames, got {}", frames);
            }
        } else {
            panic!("Expected audio chunk");
        }
    }
}