pre-passage f32

src-tauri/Cargo.toml

@@ -35,4 +35,4 @@ moka = {version = "0.12", features = ["future"] }
 arc-swap = "1.7"
 crossbeam-channel = "0.5"
 kanal = "0.1"
-rubato = "0.16.2"
+rubato = "0.16"

src-tauri/src/core/capture.rs

@@ -4,7 +4,7 @@ use std::thread;
 use std::thread::JoinHandle;
 use cpal::{default_host, BufferSize, Device, SampleRate, Stream, StreamConfig, SupportedStreamConfig};
 use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
-use rubato::{FftFixedInOut, Resampler};
+// ✅ Supprimé rubato complètement !
 use crate::core::opus::AudioOpus;
 use crate::domain::event::{Event, EventBus};
 use crate::utils::ringbuf::RingBuffer;
@@ -18,7 +18,7 @@ pub struct AudioCapture {
     event_bus: EventBus,
     microphone: Microphone,
     running: Arc<AtomicBool>,
-    ring_buffer: RingBuffer<f32>,
+    ring_buffer: RingBuffer<i16>,
     steam: Option<Stream>,
     worker: Option<JoinHandle<()>>,
 }
@@ -26,9 +26,7 @@ pub struct AudioCapture {
 impl Microphone {
     pub fn new(device: Device) -> Self {
         println!("Initializing microphone with device: {}", device.name().unwrap_or_else(|_| "Unknown".to_string()));
-        Self {
-            device
-        }
+        Self { device }
     }
 
     pub fn default() -> Self {
@@ -54,17 +52,16 @@ impl Microphone {
 
         StreamConfig {
             channels,
-            sample_rate: SampleRate(48_000),
+            sample_rate: SampleRate(48_000), // ✅ Force 48kHz
             buffer_size: BufferSize::Default,
         }
     }
 
     pub fn build_stream<F>(&self, callback: F) -> Stream
     where
-        F: FnMut(&[f32], &cpal::InputCallbackInfo) + Send + 'static,
+        F: FnMut(&[i16], &cpal::InputCallbackInfo) + Send + 'static,
     {
         let config = self.get_stream_config();
-
         self.device.build_input_stream(
             &config,
             callback,
@@ -73,7 +70,6 @@ impl Microphone {
         ).unwrap()
     }
 
-    // helpers
     pub fn get_supported_channels(&self) -> Vec<u16> {
         self.device.supported_input_configs()
             .map(|configs| configs.map(|c| c.channels()).collect())
@@ -108,9 +104,7 @@ impl AudioCapture {
         let writer = self.ring_buffer.writer();
         let stream_running = self.running.clone();
         let stream = self.microphone.build_stream(move |data, _| {
-            if !stream_running.load(Ordering::Relaxed){
-                return;
-            }
+            if !stream_running.load(Ordering::Relaxed) { return; }
             writer.push_slice_overwrite(data);
         });
         stream.play().unwrap();
@@ -126,74 +120,57 @@ impl AudioCapture {
     pub async fn stop(&mut self) {
         println!("Stopping audio capture");
         self.running.store(false, Ordering::Relaxed);
-        println!("Releasing audio stream");
         self.steam = None;
         self.ring_buffer.force_wake_up();
 
-        // code possiblement bloquant, wrap vers un thread tokio bloquant
         if let Some(worker) = self.worker.take() {
             println!("Waiting for audio processing worker to finish");
             tokio::task::spawn_blocking(move || {
                 worker.join().unwrap();
             }).await.unwrap();
         }
-        println!("Clearing ring buffer");
         self.ring_buffer.clear();
         println!("Audio capture stopped");
     }
 
-    fn run_processing_worker(&mut self){
+    fn run_processing_worker(&mut self) {
         println!("Configuring audio processing worker");
         let worker_running = self.running.clone();
         let event_bus = self.event_bus.clone();
-        let input_config = self.microphone.get_input_config();
-        println!("Audio input config: sample rate: {}, channels: {}",
-                 input_config.sample_rate().0, input_config.channels());
+        let stream_config = self.microphone.get_stream_config();
 
-        // création du worker opus
+        println!("Audio config: {} channels @ {}Hz",
+                 stream_config.channels, stream_config.sample_rate.0);
+
+        // ✅ Simple : on assume 48kHz partout !
         let opus = AudioOpus::new(48_000, 1, "voip");
         let mut encoder = opus.create_encoder().unwrap();
-
-        // création du worker resampler
-        let source_rate = input_config.sample_rate().0 as usize;
-        let mut resampler = if source_rate != 48_000 {
-            Some(Self::create_resampler(source_rate))  // ✅ Corrigé
-        } else {
-            None
-        };
-
-        // création du ringbuffer
         let reader = self.ring_buffer.reader();
 
-        // Démarrage du worker
         println!("Spawning audio processing thread");
-        let stream_config = self.microphone.get_stream_config(); // ✅ Clone la config
         self.worker = Some(thread::spawn(move || {
             println!("Audio processing thread started");
 
-            let source_rate = stream_config.sample_rate.0 as usize;  // ✅ Utilise stream_config
-            let frame_size = source_rate * 10 / 1000; // 10ms en échantillons
-            let mut raw_buffer = vec![0.0f32; frame_size];
+            let frame_size = 48_000 * 10 / 1000; // ✅ 10ms = 480 samples @ 48kHz
+            let mut raw_buffer = vec![0i16; frame_size];
 
             while worker_running.load(Ordering::Relaxed) {
-                let read_count = reader.pop_slice_blocking(&mut raw_buffer);
-                if !worker_running.load(Ordering::Relaxed){
+                let _read_count = reader.pop_slice_blocking(&mut raw_buffer);
+                if !worker_running.load(Ordering::Relaxed) {
                     println!("Audio processing thread stopping");
                     break;
                 }
 
-                // Resampling : 441→480, 48000→480, 96000→480, etc.
+                // ✅ Processing ultra-simple
                 let processed_audio = Self::process_audio_frame(
-                    &stream_config,  // ✅ Utilise stream_config
-                    resampler.as_mut(),
+                    &stream_config,
                     &raw_buffer
                 );
-                // processed_audio est TOUJOURS 480 f32 (mono/48kHz)
 
                 // Events
                 event_bus.emit_sync(Event::AudioIn(processed_audio.clone()));
 
-                match encoder.encode(&processed_audio){
+                match encoder.encode(&processed_audio) {
                     Ok(encoded_data) => {
                         event_bus.emit_sync(Event::AudioEncoded(encoded_data))
                     }
@@ -205,49 +182,22 @@ impl AudioCapture {
         }));
     }
 
-    /// Standardise un buffer en mono float32 échantillonné à 48kHz.
-    fn process_audio_frame(
-        config: &StreamConfig,
-        resampler: Option<&mut FftFixedInOut<f32>>,
-        samples: &[f32],
-    ) -> Vec<f32> {
-        // 1. Conversion mono d'abord
-        let mono = Self::sample_to_mono(config.channels as usize, samples);
-
-        // 2. Resampling si nécessaire
-        if let Some(resampler) = resampler {
-            let input: Vec<&[f32]> = vec![&mono];
-            match resampler.process(&input, None) {
-                Ok(mut output) => output.remove(0),
-                Err(e) => {
-                    println!("Erreur de resampling: {e}");
-                    mono // Fallback
-                }
-            }
-        } else {
-            mono
-        }
+    // ✅ Super simple : juste conversion mono
+    fn process_audio_frame(config: &StreamConfig, samples: &[i16]) -> Vec<i16> {
+        Self::sample_to_mono(config.channels as usize, samples)
     }
 
-    fn sample_to_mono(input_channels: usize, samples: &[f32]) -> Vec<f32> {
+    fn sample_to_mono(input_channels: usize, samples: &[i16]) -> Vec<i16> {
         if input_channels == 1 {
             samples.to_vec()
-        }else{
+        } else {
             samples
                 .chunks_exact(input_channels)
-                .map(|frame| frame.iter().copied().sum::<f32>() / input_channels as f32)
+                .map(|frame| {
+                    let sum: i32 = frame.iter().map(|&s| s as i32).sum();
+                    (sum / input_channels as i32) as i16
+                })
                 .collect()
         }
     }
-
-    // ✅ Méthode corrigée avec le bon nom
-    fn create_resampler(source_rate: usize) -> FftFixedInOut<f32> {
-        println!("Creating resampler: {}Hz -> 48kHz", source_rate);
-        FftFixedInOut::<f32>::new(
-            source_rate,
-            48_000,
-            512, // chunk size
-            1,   // mono après conversion
-        ).unwrap()
-    }
 }

src-tauri/src/core/opus.rs

@@ -65,8 +65,7 @@ impl AudioOpusEncoder {
         Ok(Self{audio_opus, encoder})
     }
 
-    // old version i16 en input, on garde au cas ou ...
-    pub fn encode_i16(&mut self, frames: &[i16]) -> Result<Vec<u8>, String> {
+    pub fn encode(&mut self, frames: &[i16]) -> Result<Vec<u8>, String> {
         let mut output = vec![0u8; 1276]; // 1276 octets (la vraie worst-case recommandée par Opus).
         let len = self.encoder.encode(frames, output.as_mut_slice())
             .map_err(|e| format!("Erreur encodage: {:?}", e))?;
@@ -74,21 +73,6 @@ impl AudioOpusEncoder {
         Ok(output)
     }
 
-    pub fn encode(&mut self, frames: &[f32]) -> Result<Vec<u8>, String> {
-        // Conversion f32 -> i16 seulement ici
-        let frames_i16: Vec<i16> = frames.iter()
-            .map(|&sample| (sample * i16::MAX as f32)
-                .clamp(i16::MIN as f32, i16::MAX as f32) as i16)
-            .collect();
-
-        let mut output = vec![0u8; 1276];
-        let len = self.encoder.encode(&frames_i16, output.as_mut_slice())
-            .map_err(|e| format!("Erreur encodage: {:?}", e))?;
-        output.truncate(len);
-        Ok(output)
-    }
-
-
     // 🔄 Approche avec buffer réutilisable (encore plus optimal)
     fn encode_reuse(&mut self, frames: &[i16], output: &mut Vec<u8>) -> Result<usize, String> {
         output.clear();
@@ -117,24 +101,10 @@ impl AudioOpusDecoder {
         Ok(Self{audio_opus, decoder})
     }
 
-    pub fn decode_i16(&mut self, frames: &[u8]) -> Result<Vec<i16>, String> {
+    pub fn decode(&mut self, frames: &[u8]) -> Result<Vec<i16>, String> {
         let mut output = vec![0i16; 5760];
         let len = self.decoder.decode(frames, output.as_mut_slice(), false).map_err(|e| format!("Erreur décodage: {:?}", e))?;
         output.truncate(len);
         Ok(output)
     }
-
-    pub fn decode(&mut self, frames: &[u8]) -> Result<Vec<f32>, String> {
-        let mut output_i16 = vec![0i16; 5760];
-        let len = self.decoder.decode(frames, &mut output_i16, false)
-            .map_err(|e| format!("Erreur décodage: {:?}", e))?;
-
-        // Conversion i16 -> f32
-        let output_f32: Vec<f32> = output_i16[..len].iter()
-            .map(|&sample| sample as f32 / i16::MAX as f32)
-            .collect();
-
-        Ok(output_f32)
-    }
-
 }

src-tauri/src/core/playback.rs

@@ -40,13 +40,94 @@ impl Speaker {
     }
 
     pub fn get_stream_config(&self) -> StreamConfig {
-        let config = self.get_output_config();
-        let mut stream_config: StreamConfig = config.into();
-        stream_config.channels = 2;
-        stream_config.sample_rate = SampleRate(44100);
-        // stream_config.buffer_size = BufferSize::Fixed(960);
-        stream_config
+        // Lister toutes les configurations supportées
+        self.print_supported_configs();
+
+        StreamConfig {
+            channels: 2,
+            sample_rate: SampleRate(44100),
+            buffer_size: BufferSize::Default
         }
+    }
+
+    pub fn print_supported_configs(&self) {
+        println!("\n=== CONFIGURATIONS AUDIO DISPONIBLES ===");
+
+        // Configuration par défaut
+        match self.device.default_output_config() {
+            Ok(config) => {
+                println!("📌 Configuration par défaut:");
+                println!("   Canaux: {}", config.channels());
+                println!("   Sample Rate: {} Hz", config.sample_rate().0);
+                println!("   Format: {:?}", config.sample_format());
+                println!("   Buffer Size: {:?}", config.buffer_size());
+            },
+            Err(e) => println!("❌ Impossible d'obtenir la config par défaut: {}", e)
+        }
+
+        // Toutes les configurations supportées
+        println!("\n📋 Toutes les configurations supportées:");
+        match self.device.supported_output_configs() {
+            Ok(configs) => {
+                for (i, config_range) in configs.enumerate() {
+                    println!("\n   Config #{}", i + 1);
+                    println!("      Canaux: {}", config_range.channels());
+                    println!("      Sample Rate: {} - {} Hz",
+                             config_range.min_sample_rate().0,
+                             config_range.max_sample_rate().0);
+                    println!("      Format: {:?}", config_range.sample_format());
+                    println!("      Buffer Size: {:?}", config_range.buffer_size());
+
+                    // Suggestions de sample rates courants
+                    let common_rates = [8000, 11025, 16000, 22050, 44100, 48000, 88200, 96000];
+                    let mut supported_common = Vec::new();
+                    for rate in common_rates {
+                        if rate >= config_range.min_sample_rate().0 && rate <= config_range.max_sample_rate().0 {
+                            supported_common.push(rate);
+                        }
+                    }
+                    if !supported_common.is_empty() {
+                        println!("      Sample rates courants supportés: {:?}", supported_common);
+                    }
+                }
+            },
+            Err(e) => println!("❌ Impossible de lister les configs: {}", e)
+        }
+
+        // Informations sur le device
+        println!("\n🎧 Informations du device:");
+        if let Ok(name) = self.device.name() {
+            println!("   Nom: {}", name);
+        }
+
+        // Test de configurations spécifiques
+        println!("\n🧪 Test de configurations spécifiques:");
+        let test_configs = [
+            (44100, 1, "Mono 44.1kHz"),
+            (44100, 2, "Stéréo 44.1kHz"),
+            (48000, 1, "Mono 48kHz"),
+            (48000, 2, "Stéréo 48kHz"),
+            (22050, 2, "Stéréo 22.05kHz"),
+        ];
+
+        for (sample_rate, channels, description) in test_configs {
+            let test_config = StreamConfig {
+                channels,
+                sample_rate: SampleRate(sample_rate),
+                buffer_size: BufferSize::Default
+            };
+
+            // Test si cette config est supportée (tentative de création d'un stream fictif)
+            let dummy_callback = |_: &mut [f32], _: &cpal::OutputCallbackInfo| {};
+            match self.device.build_output_stream(&test_config, dummy_callback, |_| {}, None) {
+                Ok(_) => println!("   ✅ {} - SUPPORTÉ", description),
+                Err(_) => println!("   ❌ {} - NON SUPPORTÉ", description),
+            }
+        }
+
+        println!("\n===========================================\n");
+    }
+
 
     pub fn build_stream<F>(&self, callback: F) -> Stream
     where
@@ -104,7 +185,6 @@ impl AudioPlayback {
             if !stream_running.load(Ordering::Relaxed){
                 return;
             }
-            println!("Audio playback stream tick");
 
             let audio_mixer = mixer.read(data.len());
             data.copy_from_slice(&audio_mixer);

src-tauri/src/domain/event.rs

@@ -1,23 +1,20 @@
 use bytes::Bytes;
-// use tokio::sync::mpsc;
 use crate::network::protocol::{MessageClient, MessageServer};
 
 pub enum Event {
     AppStarted,
     AppStopped,
 
-    AudioIn(Vec<f32>),
+    AudioIn(Vec<i16>),
     AudioEncoded(Vec<u8>),
 
     PlaybackTick(usize),
-    // PlaybackRequest(Bytes),
 
     NetConnected,
     NetDisconnected,
     NetIn(MessageServer),
     NetOut(MessageClient),
 
-
     UiStarted,
     UiStopped,
 

src-tauri/src/utils/audio_utils.rs

@@ -0,0 +1,195 @@
+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))
+    }
+}

src-tauri/src/utils/mod.rs

@@ -1,3 +1,4 @@
 pub mod ringbuf;
 pub mod real_time_event;
 pub mod shared_store;
+pub mod audio_utils;