ox_speak_client/src-tauri/src/utils/audio_utils.rs

use rubato::{Resampler, SincFixedIn, SincInterpolationType, SincInterpolationParameters, WindowFunction};
use parking_lot::Mutex;
use std::sync::Arc;

/// Resampler audio optimisé avec rubato pour temps réel
#[derive(Clone)]
pub struct AudioResampler {
    // État du resampler (recréé quand les paramètres changent)
    state: Arc<Mutex<ResamplerState>>,
    // Buffer de conversion réutilisable (évite allocations)
    conversion_buffers: Arc<Mutex<ConversionBuffers>>,
}

struct ResamplerState {
    resampler: Option<SincFixedIn<f32>>,
    current_from_rate: u32,
    current_to_rate: u32,
    current_channels: usize,
}

struct ConversionBuffers {
    f32_buffer: Vec<f32>,
    planar_input: Vec<Vec<f32>>,
    output_i16: Vec<i16>,
}

impl AudioResampler {
    pub fn new() -> Self {
        Self {
            state: Arc::new(Mutex::new(ResamplerState {
                resampler: None,
                current_from_rate: 0,
                current_to_rate: 0,
                current_channels: 0,
            })),
            conversion_buffers: Arc::new(Mutex::new(ConversionBuffers {
                f32_buffer: Vec::with_capacity(8192),
                planar_input: Vec::new(),
                output_i16: Vec::with_capacity(8192),
            })),
        }
    }

    /// Resample audio en gardant la continuité entre les chunks
    pub fn resample(
        &self,
        input: &[i16],
        from_sample_rate: u32,
        to_sample_rate: u32,
        channels: usize,
    ) -> Vec<i16> {
        // ✅ Pas de conversion si même sample rate
        if from_sample_rate == to_sample_rate || input.is_empty() {
            return input.to_vec();
        }

        let mut state = self.state.lock();
        let mut buffers = self.conversion_buffers.lock();

        // 🔄 Recrée le resampler si configuration changée
        let need_new_resampler = state.resampler.is_none()
            || state.current_from_rate != from_sample_rate
            || state.current_to_rate != to_sample_rate
            || state.current_channels != channels;

        if state.resampler.is_none()
            || state.current_from_rate != from_sample_rate
            || state.current_to_rate != to_sample_rate
            || state.current_channels != channels {
            match Self::create_resampler(from_sample_rate, to_sample_rate, channels) {
                Ok(new_resampler) => {
                    state.resampler = Some(new_resampler);
                    state.current_from_rate = from_sample_rate;
                    state.current_to_rate = to_sample_rate;
                    state.current_channels = channels;
                    println!("🔧 Resampler reconfiguré: {}Hz → {}Hz, {} canaux",
                             from_sample_rate, to_sample_rate, channels);
                }
                Err(e) => {
                    eprintln!("❌ Erreur création resampler: {}", e);
                    return input.to_vec();
                }
            }
        }

        // 🚀 Processing avec le resampler
        if let Some(ref mut resampler) = state.resampler {
            match Self::process_with_resampler(resampler, input, channels, &mut buffers) {
                Ok(output) => output,
                Err(e) => {
                    eprintln!("❌ Erreur resampling: {}", e);
                    input.to_vec()
                }
            }
        } else {
            input.to_vec()
        }
    }

    /// Crée un resampler optimisé pour votre cas d'usage
    fn create_resampler(
        from_rate: u32,
        to_rate: u32,
        channels: usize,
    ) -> Result<SincFixedIn<f32>, Box<dyn std::error::Error>> {
        let ratio = to_rate as f64 / from_rate as f64;

        // 🎯 Paramètres optimisés pour audio temps réel de qualité
        let params = SincInterpolationParameters {
            sinc_len: 256,        // Bon compromis qualité/performance
            f_cutoff: 0.95,       // Anti-aliasing fort
            interpolation: SincInterpolationType::Linear, // Plus rapide que Cubic
            oversampling_factor: 256,
            window: WindowFunction::BlackmanHarris2,
        };

        // Chunk size optimisé pour vos frames audio
        let chunk_size = 1024; // Compatible avec vos frames de 960-1024 samples

        Ok(SincFixedIn::<f32>::new(
            ratio,
            2.0, // Max ratio change pour stabilité
            params,
            chunk_size,
            channels,
        )?)
    }

    /// Process audio avec buffers réutilisables
    fn process_with_resampler(
        resampler: &mut SincFixedIn<f32>,
        input: &[i16],
        channels: usize,
        buffers: &mut ConversionBuffers,
    ) -> Result<Vec<i16>, Box<dyn std::error::Error>> {
        let frames = input.len() / channels;

        // 🔄 1. Conversion i16 → f32 (réutilise buffer)
        buffers.f32_buffer.clear();
        buffers.f32_buffer.extend(input.iter().map(|&s| s as f32 / 32768.0));

        // 🔄 2. Conversion interleaved → planar (réutilise buffers)
        buffers.planar_input.clear();
        buffers.planar_input.resize(channels, Vec::with_capacity(frames));

        for ch in 0..channels {
            buffers.planar_input[ch].clear();
        }

        for (frame_idx, frame) in buffers.f32_buffer.chunks_exact(channels).enumerate() {
            for (ch, &sample) in frame.iter().enumerate() {
                buffers.planar_input[ch].push(sample);
            }
        }

        // 🎯 3. Resampling magique !
        let output_planar = resampler.process(&buffers.planar_input, None)?;

        // 🔄 4. Conversion planar → interleaved i16 (réutilise buffer)
        let output_frames = output_planar[0].len();
        buffers.output_i16.clear();
        buffers.output_i16.reserve(output_frames * channels);

        for frame_idx in 0..output_frames {
            for ch in 0..channels {
                let sample = (output_planar[ch][frame_idx] * 32767.0)
                    .round()
                    .clamp(-32768.0, 32767.0) as i16;
                buffers.output_i16.push(sample);
            }
        }

        Ok(buffers.output_i16.clone())
    }

    /// Réinitialise l'état interne (pour éviter glitches lors de changements)
    pub fn reset(&self) {
        let mut state = self.state.lock();
        if let Some(ref mut resampler) = state.resampler {
            let _ = resampler.reset();
        }
        println!("🔄 Resampler reset");
    }

    /// Version oneshot sans état (pour tests)
    pub fn resample_oneshot(
        input: &[i16],
        from_rate: u32,
        to_rate: u32,
        channels: usize,
    ) -> Result<Vec<i16>, Box<dyn std::error::Error>> {
        let resampler = AudioResampler::new();
        Ok(resampler.resample(input, from_rate, to_rate, channels))
    }
}