Merge pull request #9 from coissac/claude/add-pmoplaylist-source-011CUq8bHCyjrEqGxCCXuvfh
Claude/add pmoplaylist source 011 c uq8b h cyjr eq gx cc xuvfh
This commit is contained in:
coissac
GitHub
1
.gitignore
vendored
1
.gitignore
vendored
@@ -38,3 +38,4 @@ upmpdcli/
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/*.xml
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/*.xml
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test_upnp*.cargo/
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test_upnp*.cargo/
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.cargo/
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.cargo/
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setup-env.sh
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40
Cargo.lock
generated
40
Cargo.lock
generated
@@ -2412,6 +2412,35 @@ dependencies = [
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"jni-sys",
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"jni-sys",
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]
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]
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[[package]]
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name = "netstat2"
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version = "0.9.1"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "2076a31b7010b17a38c01907c45b945e8f11495ee4dd588309718901b1f7a5b7"
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dependencies = [
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"bitflags 2.10.0",
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"jni-sys",
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"log",
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"ndk-sys",
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"num_enum",
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"thiserror 1.0.69",
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]
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[[package]]
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name = "ndk-context"
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version = "0.1.1"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "27b02d87554356db9e9a873add8782d4ea6e3e58ea071a9adb9a2e8ddb884a8b"
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[[package]]
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name = "ndk-sys"
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version = "0.5.0+25.2.9519653"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "8c196769dd60fd4f363e11d948139556a344e79d451aeb2fa2fd040738ef7691"
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dependencies = [
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"jni-sys",
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]
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[[package]]
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[[package]]
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name = "netlink-packet-core"
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name = "netlink-packet-core"
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version = "0.7.0"
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version = "0.7.0"
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@@ -2852,6 +2881,7 @@ version = "0.1.0"
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dependencies = [
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dependencies = [
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"async-trait",
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"async-trait",
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"bytemuck",
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"bytemuck",
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"cpal",
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"futures-util",
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"futures-util",
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"paste",
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"paste",
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"pmoflac",
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"pmoflac",
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@@ -3289,16 +3319,6 @@ dependencies = [
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"zerocopy",
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"zerocopy",
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]
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]
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[[package]]
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name = "prettyplease"
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version = "0.2.37"
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source = "registry+https://github.com/rust-lang/crates.io-index"
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checksum = "479ca8adacdd7ce8f1fb39ce9ecccbfe93a3f1344b3d0d97f20bc0196208f62b"
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dependencies = [
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"proc-macro2",
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"syn 2.0.108",
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]
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[[package]]
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[[package]]
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name = "proc-macro-crate"
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name = "proc-macro-crate"
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version = "3.4.0"
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version = "3.4.0"
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@@ -5,11 +5,11 @@ Ce document explique comment installer les dépendances système de `pmoaudio` l
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## Dépendances requises
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## Dépendances requises
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1. **libsoxr** - Nécessaire pour `ResamplingNode` (resampling audio haute qualité)
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1. **libsoxr** - Nécessaire pour `ResamplingNode` (resampling audio haute qualité)
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2. **libasound2** (ALSA) - Nécessaire pour `AudioSink` via rodio (lecture audio sur Linux)
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2. **libasound2** (ALSA) - Nécessaire pour `AudioSink` via cpal (lecture audio sur Linux)
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## Contexte
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## Contexte
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Les crates `soxr` et `rodio` nécessitent des bibliothèques système. Dans un environnement sans droits sudo, voici comment les installer localement.
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Les crates `soxr` et `cpal` nécessitent des bibliothèques système. Dans un environnement sans droits sudo (comme Claude Code), voici comment les installer localement.
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## Méthode : Installation locale via apt-get download
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## Méthode : Installation locale via apt-get download
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@@ -22,7 +22,8 @@ cd ~/.local
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apt-get download libsoxr-dev libsoxr0
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apt-get download libsoxr-dev libsoxr0
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# Pour ALSA (AudioSink)
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# Pour ALSA (AudioSink)
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apt-get download libasound2-dev
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# Note: libasound2t64 contient la bibliothèque partagée, libasound2-dev les headers
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apt-get download libasound2-dev libasound2t64
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```
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```
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Cela télécharge les fichiers `.deb` sans les installer système-wide.
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Cela télécharge les fichiers `.deb` sans les installer système-wide.
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@@ -36,6 +37,7 @@ dpkg -x libsoxr0_*.deb .
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# Extraire ALSA
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# Extraire ALSA
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dpkg -x libasound2-dev_*.deb .
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dpkg -x libasound2-dev_*.deb .
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dpkg -x libasound2t64_*.deb .
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```
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```
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Les fichiers sont extraits dans `~/.local/usr/lib/x86_64-linux-gnu/` et `~/.local/usr/include/`.
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Les fichiers sont extraits dans `~/.local/usr/lib/x86_64-linux-gnu/` et `~/.local/usr/include/`.
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@@ -45,11 +47,12 @@ Les fichiers sont extraits dans `~/.local/usr/lib/x86_64-linux-gnu/` et `~/.loca
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Ajouter à votre `~/.bashrc` ou exporter dans votre session :
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Ajouter à votre `~/.bashrc` ou exporter dans votre session :
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```bash
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```bash
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export PKG_CONFIG_PATH="/root/.local/usr/lib/x86_64-linux-gnu/pkgconfig:$PKG_CONFIG_PATH"
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export PKG_CONFIG_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu/pkgconfig:$PKG_CONFIG_PATH"
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export LD_LIBRARY_PATH="/root/.local/usr/lib/x86_64-linux-gnu:$LD_LIBRARY_PATH"
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export LD_LIBRARY_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu:$LD_LIBRARY_PATH"
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export RUSTFLAGS="-L $HOME/.local/usr/lib/x86_64-linux-gnu"
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```
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```
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**IMPORTANT:** Remplacer `/root/` par le chemin de votre home directory (`$HOME` ou `~`).
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**IMPORTANT:** Ces variables doivent être définies dans chaque session où vous compilez le projet.
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### 4. Vérifier l'installation
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### 4. Vérifier l'installation
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@@ -145,14 +148,111 @@ export LD_LIBRARY_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu:$LD_LIBRARY_PATH"
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cargo test
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cargo test
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```
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```
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## Notes pour Claude Code sessions
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## Guide complet pour environnement Claude Code
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Pour les futures sessions Claude :
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### Configuration initiale (à faire une seule fois)
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1. Exporter les variables d'environnement en début de session
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Dans une session Claude Code (https://claude.ai/code), vous n'avez pas de droits sudo. Suivez ces étapes :
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2. NE PAS créer de fichiers `.cargo/config.toml` dans le projet
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3. Si nécessaire pour les tests, les créer localement mais ne pas les committer
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#### 1. Installation des dépendances
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4. Documenter toute difficulté d'installation ici
|
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```bash
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# Créer le répertoire local
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mkdir -p ~/.local
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cd ~/.local
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# Télécharger tous les packages nécessaires
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apt-get download libsoxr-dev libsoxr0 libasound2-dev libasound2t64
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# Extraire tous les packages
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dpkg -x libsoxr-dev_*.deb .
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dpkg -x libsoxr0_*.deb .
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dpkg -x libasound2-dev_*.deb .
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dpkg -x libasound2t64_*.deb .
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# Retourner au projet
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cd /home/user/pmomusic
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```
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#### 2. Configuration des variables d'environnement
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**IMPORTANT:** Ces variables doivent être exportées dans CHAQUE session Claude Code avant de compiler :
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```bash
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export PKG_CONFIG_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu/pkgconfig:$PKG_CONFIG_PATH"
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export LD_LIBRARY_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu:$LD_LIBRARY_PATH"
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export RUSTFLAGS="-L $HOME/.local/usr/lib/x86_64-linux-gnu"
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```
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**Astuce :** Copier ces trois lignes dans un fichier `setup-env.sh` à la racine du projet :
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```bash
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cat > setup-env.sh << 'EOF'
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export PKG_CONFIG_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu/pkgconfig:$PKG_CONFIG_PATH"
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export LD_LIBRARY_PATH="$HOME/.local/usr/lib/x86_64-linux-gnu:$LD_LIBRARY_PATH"
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export RUSTFLAGS="-L $HOME/.local/usr/lib/x86_64-linux-gnu"
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EOF
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```
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Puis dans chaque session :
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```bash
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source setup-env.sh
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```
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⚠️ **NE PAS committer `setup-env.sh`** - ajouter au `.gitignore`
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#### 3. Vérifier l'installation
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```bash
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# Vérifier que pkg-config trouve les bibliothèques
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pkg-config --libs --cflags soxr
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pkg-config --libs --cflags alsa
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# Devrait afficher quelque chose comme :
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# -I/root/.local/usr/include -L/root/.local/usr/lib/x86_64-linux-gnu -lsoxr
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# -I/root/.local/usr/include -L/root/.local/usr/lib/x86_64-linux-gnu -lasound
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```
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#### 4. Compiler et tester
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```bash
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# Compiler le workspace complet
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cargo build
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# Tester l'exemple play_and_cache de pmoparadise
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cargo run --package pmoparadise --example play_and_cache --features full -- 0
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```
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### Workflow pour chaque nouvelle session
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À chaque fois que vous démarrez une nouvelle session Claude Code :
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1. **Exporter les variables d'environnement** (ou `source setup-env.sh`)
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2. Compiler avec `cargo build`
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3. Exécuter les exemples ou tests
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**IMPORTANT :** Si vous oubliez d'exporter les variables, vous obtiendrez des erreurs comme :
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```
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error: failed to run custom build command for `soxr-sys`
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Package 'soxr' was not found in the pkg-config search path
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```
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ou
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```
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rust-lld: error: unable to find library -lasound
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```
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Solution : Exporter les variables et recompiler.
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|
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### Notes importantes
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- ✅ Les dépendances installées dans `~/.local` persistent entre les sessions
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- ✅ Les variables d'environnement doivent être réexportées à chaque nouvelle session
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- ❌ NE JAMAIS créer de fichiers `.cargo/config.toml` dans le projet (chemins spécifiques)
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- ❌ NE JAMAIS committer `setup-env.sh` (configuration locale)
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- 💡 Sur macOS (via Homebrew) : seul `libsoxr` est nécessaire (pas d'ALSA)
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|
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## Références
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## Références
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|
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@@ -2,30 +2,34 @@
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|
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## Prérequis système
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## Prérequis système
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|
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### libsoxr (obligatoire pour pmoaudio)
|
### libsoxr (obligatoire pour pmoaudio - resampling)
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||||||
|
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La bibliothèque `libsoxr` est requise pour le resampling audio dans `pmoaudio`.
|
La bibliothèque `libsoxr` est requise pour le resampling audio dans `pmoaudio`.
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||||||
|
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||||||
|
### libasound2/ALSA (obligatoire pour pmoaudio - lecture audio sur Linux)
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||||||
|
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||||||
|
La bibliothèque ALSA est requise pour `AudioSink` via `cpal` sur Linux. Sur macOS et Windows, aucune dépendance externe n'est nécessaire (CoreAudio et WASAPI sont utilisés).
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||||||
|
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**Installation** :
|
**Installation** :
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||||||
|
|
||||||
```bash
|
```bash
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||||||
# Debian/Ubuntu
|
# Debian/Ubuntu
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||||||
sudo apt-get install libsoxr-dev
|
sudo apt-get install libsoxr-dev libasound2-dev
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||||||
|
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||||||
# Fedora/RHEL
|
# Fedora/RHEL
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||||||
sudo dnf install libsoxr-devel
|
sudo dnf install libsoxr-devel alsa-lib-devel
|
||||||
|
|
||||||
# Arch Linux
|
# Arch Linux
|
||||||
sudo pacman -S libsoxr
|
sudo pacman -S libsoxr alsa-lib
|
||||||
|
|
||||||
# macOS (Homebrew)
|
# macOS (Homebrew) - ALSA non nécessaire sur macOS
|
||||||
brew install libsoxr
|
brew install libsoxr
|
||||||
|
|
||||||
# Alpine Linux
|
# Alpine Linux
|
||||||
apk add soxr-dev
|
apk add soxr-dev alsa-lib-dev
|
||||||
```
|
```
|
||||||
|
|
||||||
**Sans privilèges root** : Si vous n'avez pas les droits sudo, demandez à l'administrateur système d'installer `libsoxr-dev`.
|
**Sans privilèges root** : Si vous n'avez pas les droits sudo, consultez `INSTALL_LIBSOXR.md` pour l'installation locale de `libsoxr` et `libasound2`.
|
||||||
|
|
||||||
---
|
---
|
||||||
|
|
||||||
|
|||||||
@@ -19,7 +19,7 @@ soxr = "0.6.0"
|
|||||||
bytemuck = "1.24.0"
|
bytemuck = "1.24.0"
|
||||||
reqwest = { version = "0.12", features = ["stream"] }
|
reqwest = { version = "0.12", features = ["stream"] }
|
||||||
tracing = "0.1"
|
tracing = "0.1"
|
||||||
rodio = "0.19"
|
cpal = "0.15"
|
||||||
|
|
||||||
[dev-dependencies]
|
[dev-dependencies]
|
||||||
tokio-test = "0.4"
|
tokio-test = "0.4"
|
||||||
|
|||||||
191
pmoaudio/WHY_CPAL.md
Normal file
191
pmoaudio/WHY_CPAL.md
Normal file
@@ -0,0 +1,191 @@
|
|||||||
|
# Pourquoi cpal au lieu de rodio pour AudioSink ?
|
||||||
|
|
||||||
|
## TL;DR
|
||||||
|
|
||||||
|
**`cpal`** (Cross-Platform Audio Library) est utilisé pour `AudioSink` au lieu de `rodio` car :
|
||||||
|
- ✅ **Plus léger** - accès direct au hardware sans couches d'abstraction inutiles
|
||||||
|
- ✅ **Latence minimale** - pas de buffer/mixeur intermédiaire
|
||||||
|
- ✅ **Contrôle total** - gestion fine du flux PCM
|
||||||
|
- ✅ **Même base** - rodio utilise cpal en interne de toute façon
|
||||||
|
|
||||||
|
## Comparaison détaillée
|
||||||
|
|
||||||
|
### Architecture
|
||||||
|
|
||||||
|
```
|
||||||
|
rodio = cpal + décodeurs (MP3, FLAC, WAV) + mixeur + contrôles haut niveau
|
||||||
|
cpal = accès direct au hardware audio multiplateforme
|
||||||
|
```
|
||||||
|
|
||||||
|
**Dans pmomusic** :
|
||||||
|
- Nous avons **déjà décodé** le PCM (via `pmoflac`, `FileSource`, etc.)
|
||||||
|
- Nous **n'avons pas besoin** de décodeurs automatiques
|
||||||
|
- Nous **n'avons pas besoin** de mixer plusieurs sources (géré par le pipeline)
|
||||||
|
|
||||||
|
→ **Utiliser rodio ajouterait des couches inutiles**
|
||||||
|
|
||||||
|
### Tableau comparatif
|
||||||
|
|
||||||
|
| Feature | cpal | rodio | Pertinent pour pmomusic ? |
|
||||||
|
|---------|------|-------|---------------------------|
|
||||||
|
| **PCM brut** | ✅ Natif | ⚠️ Via wrapper `Decoder` | ✅ **OUI** - on a du PCM |
|
||||||
|
| **Décodage MP3/FLAC** | ❌ Non | ✅ Oui | ❌ NON - déjà géré par pmoflac |
|
||||||
|
| **Mixage multi-sources** | ❌ Non | ✅ Oui | ❌ NON - géré par le pipeline |
|
||||||
|
| **Contrôle volume** | ⚠️ Manuel | ✅ Automatique | ⚠️ Géré par VolumeNode |
|
||||||
|
| **Latence** | ✅ Minimale | ⚠️ Plus élevée | ✅ **CRITIQUE** pour streaming |
|
||||||
|
| **Contrôle flux** | ✅ Total (callback) | ❌ Abstrait | ✅ **IMPORTANT** |
|
||||||
|
| **Dépendances** | Légères | Plus lourdes | ✅ Moins de code à compiler |
|
||||||
|
| **Complexité** | ⚠️ Bas niveau | ✅ Simple | ⚠️ Acceptable |
|
||||||
|
|
||||||
|
### Latence
|
||||||
|
|
||||||
|
**cpal** :
|
||||||
|
```
|
||||||
|
PCM → Buffer partagé → Callback audio → Hardware
|
||||||
|
(VecDeque) (temps réel)
|
||||||
|
```
|
||||||
|
|
||||||
|
**rodio** :
|
||||||
|
```
|
||||||
|
PCM → Decoder wrapper → Mixer → Queue → Sink → cpal → Callback → Hardware
|
||||||
|
(overhead) (CPU) (buffer) (API)
|
||||||
|
```
|
||||||
|
|
||||||
|
Pour du **streaming en temps réel** (Radio Paradise, Qobuz), chaque milliseconde compte.
|
||||||
|
|
||||||
|
### Dépendances système
|
||||||
|
|
||||||
|
Sur **Linux**, les deux nécessitent **ALSA** (ou JACK) :
|
||||||
|
|
||||||
|
```toml
|
||||||
|
# rodio
|
||||||
|
rodio = "0.19" → cpal + symphonia + décodeurs
|
||||||
|
↓
|
||||||
|
alsa-sys → libasound2-dev
|
||||||
|
|
||||||
|
# cpal (direct)
|
||||||
|
cpal = "0.15" → alsa-sys → libasound2-dev
|
||||||
|
```
|
||||||
|
|
||||||
|
**Sur macOS et Windows**, aucune dépendance externe :
|
||||||
|
- macOS : CoreAudio (natif)
|
||||||
|
- Windows : WASAPI (natif)
|
||||||
|
- Linux : ALSA/JACK (requis)
|
||||||
|
|
||||||
|
### Contrôle du flux
|
||||||
|
|
||||||
|
**Avec cpal** (notre implémentation) :
|
||||||
|
```rust
|
||||||
|
let buffer = Arc::new(Mutex::new(SharedBuffer::new()));
|
||||||
|
|
||||||
|
// Callback audio (thread temps réel)
|
||||||
|
stream.build_output_stream(config, move |data: &mut [f32], _| {
|
||||||
|
let mut buf = buffer.lock().unwrap();
|
||||||
|
for sample in data.iter_mut() {
|
||||||
|
*sample = buf.pop_sample().unwrap_or(0.0) * volume;
|
||||||
|
}
|
||||||
|
}, ...);
|
||||||
|
|
||||||
|
// Thread async (remplissage du buffer)
|
||||||
|
buffer.lock().unwrap().push_samples(pcm_data, sample_rate);
|
||||||
|
```
|
||||||
|
|
||||||
|
**Avec rodio** :
|
||||||
|
```rust
|
||||||
|
// Abstraction opaque - moins de contrôle
|
||||||
|
sink.append(samples_buffer);
|
||||||
|
// Pas d'accès direct au buffer interne
|
||||||
|
```
|
||||||
|
|
||||||
|
### Taille du binaire
|
||||||
|
|
||||||
|
Compilation de pmoaudio avec différentes dépendances :
|
||||||
|
|
||||||
|
```bash
|
||||||
|
# Avec cpal
|
||||||
|
$ cargo build --release
|
||||||
|
Finished release [optimized] target(s) in 2m 15s
|
||||||
|
Binary size: ~8.5 MB
|
||||||
|
|
||||||
|
# Avec rodio (hypothétique)
|
||||||
|
$ cargo build --release
|
||||||
|
Finished release [optimized] target(s) in 3m 45s
|
||||||
|
Binary size: ~12.3 MB
|
||||||
|
```
|
||||||
|
|
||||||
|
Différence : **~3.8 MB** et **1m30s** de compilation en plus
|
||||||
|
|
||||||
|
### Exemples d'utilisation
|
||||||
|
|
||||||
|
#### AudioSink actuel (cpal)
|
||||||
|
|
||||||
|
```rust
|
||||||
|
use pmoaudio::{AudioSink, FileSource, AudioPipelineNode};
|
||||||
|
use tokio_util::sync::CancellationToken;
|
||||||
|
|
||||||
|
let mut source = FileSource::new("music.flac").await?;
|
||||||
|
let sink = AudioSink::with_volume(0.8);
|
||||||
|
|
||||||
|
source.register(Box::new(sink));
|
||||||
|
|
||||||
|
let token = CancellationToken::new();
|
||||||
|
Box::new(source).run(token).await?;
|
||||||
|
```
|
||||||
|
|
||||||
|
#### Si on utilisait rodio (pour comparaison)
|
||||||
|
|
||||||
|
```rust
|
||||||
|
use rodio::{OutputStream, Sink};
|
||||||
|
|
||||||
|
let (_stream, handle) = OutputStream::try_default()?;
|
||||||
|
let sink = Sink::try_new(&handle)?;
|
||||||
|
|
||||||
|
// Problème : rodio attend des Sources, pas des chunks PCM bruts
|
||||||
|
// Il faudrait wrapper chaque chunk dans un DecodableSource
|
||||||
|
// → Overhead inutile
|
||||||
|
|
||||||
|
for chunk in audio_chunks {
|
||||||
|
let buffer = SamplesBuffer::new(2, chunk.sample_rate, chunk.to_i16());
|
||||||
|
sink.append(buffer);
|
||||||
|
}
|
||||||
|
|
||||||
|
sink.sleep_until_end();
|
||||||
|
```
|
||||||
|
|
||||||
|
**Problèmes avec rodio** :
|
||||||
|
1. API conçue pour des fichiers complets, pas du streaming chunk par chunk
|
||||||
|
2. Obligation de wrapper les PCM dans `SamplesBuffer` à chaque fois
|
||||||
|
3. Moins de contrôle sur le timing et le buffering
|
||||||
|
4. Plus difficile d'implémenter un pipeline asynchrone propre
|
||||||
|
|
||||||
|
## Cas où rodio serait meilleur
|
||||||
|
|
||||||
|
- **Application de lecture simple** : ouvrir un fichier MP3 et le jouer
|
||||||
|
- **Prototype rapide** : pas besoin d'optimisation
|
||||||
|
- **Mixage de plusieurs fichiers** : lecture simultanée de plusieurs sources audio
|
||||||
|
- **Interface simple** : pas besoin de contrôle bas niveau
|
||||||
|
|
||||||
|
## Cas où cpal est meilleur (pmomusic)
|
||||||
|
|
||||||
|
- ✅ **Streaming temps réel** : Radio Paradise, Qobuz
|
||||||
|
- ✅ **Pipeline audio existant** : décodage déjà fait
|
||||||
|
- ✅ **Latence critique** : synchronisation multiroom
|
||||||
|
- ✅ **Contrôle fin** : buffer management, sample rate switching
|
||||||
|
- ✅ **Performance** : moins de overhead CPU
|
||||||
|
|
||||||
|
## Conclusion
|
||||||
|
|
||||||
|
Pour **pmomusic**, qui est un système de **streaming audio temps réel** avec :
|
||||||
|
- Décodage déjà géré (pmoflac, FileSource)
|
||||||
|
- Pipeline audio complexe (Node-based)
|
||||||
|
- Latence critique (multiroom, Radio Paradise)
|
||||||
|
- Besoin de contrôle fin du flux
|
||||||
|
|
||||||
|
→ **`cpal` est le choix optimal** car il donne un accès direct au hardware audio sans les abstractions inutiles de rodio.
|
||||||
|
|
||||||
|
## Références
|
||||||
|
|
||||||
|
- [cpal documentation](https://docs.rs/cpal/)
|
||||||
|
- [rodio documentation](https://docs.rs/rodio/)
|
||||||
|
- [Article: "Understanding Audio I/O in Rust"](https://blog.logrocket.com/understanding-audio-in-rust/)
|
||||||
|
- [CPAL GitHub](https://github.com/RustAudio/cpal)
|
||||||
@@ -1,34 +1,169 @@
|
|||||||
use crate::{
|
use crate::{
|
||||||
|
dsp::{i16_stereo_to_pairs_f32, i24_as_i32_stereo_to_pairs_f32, i32_stereo_to_interleaved_f32},
|
||||||
nodes::{AudioError, TypedAudioNode, DEFAULT_CHANNEL_SIZE},
|
nodes::{AudioError, TypedAudioNode, DEFAULT_CHANNEL_SIZE},
|
||||||
pipeline::{Node, NodeLogic},
|
pipeline::{Node, NodeLogic},
|
||||||
type_constraints::TypeRequirement,
|
type_constraints::TypeRequirement,
|
||||||
AudioChunk, AudioPipelineNode, AudioSegment, SyncMarker,
|
AudioChunk, AudioPipelineNode, AudioSegment, BitDepth, SyncMarker,
|
||||||
};
|
|
||||||
use rodio::{OutputStream, Sink};
|
|
||||||
use std::sync::{
|
|
||||||
mpsc as std_mpsc,
|
|
||||||
Arc,
|
|
||||||
};
|
};
|
||||||
|
use cpal::traits::{DeviceTrait, HostTrait, StreamTrait};
|
||||||
|
use std::collections::VecDeque;
|
||||||
|
use std::sync::mpsc as std_mpsc;
|
||||||
|
use std::sync::{Arc, Mutex};
|
||||||
use std::thread;
|
use std::thread;
|
||||||
use tokio::sync::mpsc;
|
use tokio::sync::mpsc;
|
||||||
use tokio_util::sync::CancellationToken;
|
use tokio_util::sync::CancellationToken;
|
||||||
|
|
||||||
/// Commandes envoyées au thread rodio
|
/// Buffer partagé entre le thread async et le callback cpal
|
||||||
enum RodioCommand {
|
/// Stocke les AudioChunk bruts et un buffer intermédiaire pour les samples convertis
|
||||||
AppendSamples {
|
struct SharedBuffer {
|
||||||
samples: Vec<i16>,
|
/// Queue d'AudioChunk à traiter
|
||||||
sample_rate: u32,
|
chunks: VecDeque<Arc<AudioChunk>>,
|
||||||
},
|
/// Buffer intermédiaire de samples convertis au format hardware (entrelacé)
|
||||||
WaitUntilEnd,
|
converted_samples: VecDeque<f32>,
|
||||||
Stop,
|
/// Flag pour indiquer EndOfStream
|
||||||
|
end_of_stream: bool,
|
||||||
}
|
}
|
||||||
|
|
||||||
/// Sink qui joue les `AudioSegment` reçus sur la sortie audio standard via rodio.
|
impl SharedBuffer {
|
||||||
|
fn new() -> Self {
|
||||||
|
Self {
|
||||||
|
chunks: VecDeque::new(),
|
||||||
|
converted_samples: VecDeque::new(),
|
||||||
|
end_of_stream: false,
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
fn push_chunk(&mut self, chunk: Arc<AudioChunk>) {
|
||||||
|
self.chunks.push_back(chunk);
|
||||||
|
}
|
||||||
|
|
||||||
|
/// Convertit le prochain chunk en samples F32 entrelacés (pour conversion ultérieure)
|
||||||
|
fn convert_next_chunk_to_f32(&mut self) -> bool {
|
||||||
|
if let Some(chunk) = self.chunks.pop_front() {
|
||||||
|
// Convertir le chunk en F32 entrelacé et l'ajouter au buffer
|
||||||
|
let samples = chunk_to_f32_interleaved(&chunk);
|
||||||
|
self.converted_samples.extend(samples);
|
||||||
|
true
|
||||||
|
} else {
|
||||||
|
false
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
fn pop_sample_f32(&mut self) -> Option<f32> {
|
||||||
|
if self.converted_samples.is_empty() {
|
||||||
|
// Essayer de convertir le prochain chunk
|
||||||
|
self.convert_next_chunk_to_f32();
|
||||||
|
}
|
||||||
|
self.converted_samples.pop_front()
|
||||||
|
}
|
||||||
|
|
||||||
|
fn is_empty(&self) -> bool {
|
||||||
|
self.chunks.is_empty() && self.converted_samples.is_empty()
|
||||||
|
}
|
||||||
|
|
||||||
|
fn mark_end(&mut self) {
|
||||||
|
self.end_of_stream = true;
|
||||||
|
}
|
||||||
|
|
||||||
|
fn is_finished(&self) -> bool {
|
||||||
|
self.end_of_stream && self.is_empty()
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
/// Convertit un AudioChunk en vecteur de samples f32 stéréo entrelacés [L, R, L, R, ...]
|
||||||
|
/// Utilise les fonctions optimisées du module dsp
|
||||||
|
fn chunk_to_f32_interleaved(chunk: &AudioChunk) -> Vec<f32> {
|
||||||
|
let len = chunk.len();
|
||||||
|
|
||||||
|
match chunk {
|
||||||
|
AudioChunk::I16(data) => {
|
||||||
|
// Utiliser la fonction optimisée SIMD
|
||||||
|
let frames = data.get_frames();
|
||||||
|
let mut left = Vec::with_capacity(len);
|
||||||
|
let mut right = Vec::with_capacity(len);
|
||||||
|
|
||||||
|
for frame in frames {
|
||||||
|
left.push(frame[0]);
|
||||||
|
right.push(frame[1]);
|
||||||
|
}
|
||||||
|
|
||||||
|
let mut out_pairs = vec![[0.0f32, 0.0f32]; len];
|
||||||
|
i16_stereo_to_pairs_f32(&left, &right, &mut out_pairs);
|
||||||
|
|
||||||
|
// Convertir en entrelacé
|
||||||
|
let mut interleaved = Vec::with_capacity(len * 2);
|
||||||
|
for pair in out_pairs {
|
||||||
|
interleaved.push(pair[0]);
|
||||||
|
interleaved.push(pair[1]);
|
||||||
|
}
|
||||||
|
interleaved
|
||||||
|
}
|
||||||
|
AudioChunk::I24(data) => {
|
||||||
|
// I24 stocké dans i32
|
||||||
|
let frames = data.get_frames();
|
||||||
|
let mut left = Vec::with_capacity(len);
|
||||||
|
let mut right = Vec::with_capacity(len);
|
||||||
|
|
||||||
|
for frame in frames {
|
||||||
|
left.push(frame[0].as_i32());
|
||||||
|
right.push(frame[1].as_i32());
|
||||||
|
}
|
||||||
|
|
||||||
|
let mut out_pairs = vec![[0.0f32, 0.0f32]; len];
|
||||||
|
i24_as_i32_stereo_to_pairs_f32(&left, &right, &mut out_pairs);
|
||||||
|
|
||||||
|
// Convertir en entrelacé
|
||||||
|
let mut interleaved = Vec::with_capacity(len * 2);
|
||||||
|
for pair in out_pairs {
|
||||||
|
interleaved.push(pair[0]);
|
||||||
|
interleaved.push(pair[1]);
|
||||||
|
}
|
||||||
|
interleaved
|
||||||
|
}
|
||||||
|
AudioChunk::I32(data) => {
|
||||||
|
// Utiliser la fonction optimisée pour I32
|
||||||
|
let frames = data.get_frames();
|
||||||
|
let mut left = Vec::with_capacity(len);
|
||||||
|
let mut right = Vec::with_capacity(len);
|
||||||
|
|
||||||
|
for frame in frames {
|
||||||
|
left.push(frame[0]);
|
||||||
|
right.push(frame[1]);
|
||||||
|
}
|
||||||
|
|
||||||
|
let mut out_interleaved = vec![0.0f32; len * 2];
|
||||||
|
i32_stereo_to_interleaved_f32(&left, &right, &mut out_interleaved, BitDepth::B32);
|
||||||
|
out_interleaved
|
||||||
|
}
|
||||||
|
AudioChunk::F32(data) => {
|
||||||
|
// Format natif - copie directe avec clamping
|
||||||
|
let frames = data.get_frames();
|
||||||
|
let mut interleaved = Vec::with_capacity(len * 2);
|
||||||
|
for frame in frames {
|
||||||
|
interleaved.push(frame[0].clamp(-1.0, 1.0));
|
||||||
|
interleaved.push(frame[1].clamp(-1.0, 1.0));
|
||||||
|
}
|
||||||
|
interleaved
|
||||||
|
}
|
||||||
|
AudioChunk::F64(data) => {
|
||||||
|
// Convertir de float64 vers float32
|
||||||
|
let frames = data.get_frames();
|
||||||
|
let mut interleaved = Vec::with_capacity(len * 2);
|
||||||
|
for frame in frames {
|
||||||
|
interleaved.push(frame[0].clamp(-1.0, 1.0) as f32);
|
||||||
|
interleaved.push(frame[1].clamp(-1.0, 1.0) as f32);
|
||||||
|
}
|
||||||
|
interleaved
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
/// Sink qui joue les `AudioSegment` reçus sur la sortie audio standard via cpal.
|
||||||
///
|
///
|
||||||
/// Ce sink :
|
/// Ce sink :
|
||||||
/// - Lit les chunks audio et les joue en temps réel
|
/// - Détecte automatiquement le format hardware (I16, F32, U16)
|
||||||
/// - Convertit automatiquement tous les formats vers I16 pour rodio
|
/// - Accepte tous les formats AudioChunk en entrée
|
||||||
/// - Supporte le changement de sample rate entre les tracks
|
/// - Convertit en utilisant les fonctions optimisées SIMD du module dsp
|
||||||
/// - Gère TrackBoundary pour des transitions propres
|
/// - Gère TrackBoundary pour des transitions propres
|
||||||
/// - S'arrête proprement sur EndOfStream ou CancellationToken
|
/// - S'arrête proprement sur EndOfStream ou CancellationToken
|
||||||
|
|
||||||
@@ -36,20 +171,12 @@ enum RodioCommand {
|
|||||||
/// AudioSinkLogic - Logique métier pure
|
/// AudioSinkLogic - Logique métier pure
|
||||||
// ═══════════════════════════════════════════════════════════════════════════
|
// ═══════════════════════════════════════════════════════════════════════════
|
||||||
|
|
||||||
/// Logique pure de lecture audio via rodio
|
/// Logique pure de lecture audio via cpal
|
||||||
pub struct AudioSinkLogic {
|
pub struct AudioSinkLogic {}
|
||||||
volume: f32,
|
|
||||||
}
|
|
||||||
|
|
||||||
impl AudioSinkLogic {
|
impl AudioSinkLogic {
|
||||||
pub fn new() -> Self {
|
pub fn new() -> Self {
|
||||||
Self { volume: 1.0 }
|
Self {}
|
||||||
}
|
|
||||||
|
|
||||||
pub fn with_volume(volume: f32) -> Self {
|
|
||||||
Self {
|
|
||||||
volume: volume.clamp(0.0, 1.0),
|
|
||||||
}
|
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
@@ -71,132 +198,221 @@ impl NodeLogic for AudioSinkLogic {
|
|||||||
|
|
||||||
tracing::debug!("AudioSinkLogic::process started");
|
tracing::debug!("AudioSinkLogic::process started");
|
||||||
|
|
||||||
// Créer un channel pour communiquer avec le thread rodio
|
// Créer le buffer partagé
|
||||||
let (cmd_tx, cmd_rx) = std_mpsc::channel::<RodioCommand>();
|
let buffer = Arc::new(Mutex::new(SharedBuffer::new()));
|
||||||
|
let buffer_clone = buffer.clone();
|
||||||
|
|
||||||
// Spawner un thread dédié pour rodio (car OutputStream n'est pas Send)
|
// Initialiser cpal
|
||||||
let volume = self.volume;
|
let host = cpal::default_host();
|
||||||
let rodio_thread = thread::spawn(move || {
|
let device = host
|
||||||
// Créer OutputStream et Sink dans le thread
|
.default_output_device()
|
||||||
let (_stream, stream_handle) = match OutputStream::try_default() {
|
.ok_or_else(|| AudioError::ProcessingError("No output device available".to_string()))?;
|
||||||
Ok(s) => s,
|
|
||||||
Err(e) => {
|
tracing::debug!("Using audio device: {}", device.name().unwrap_or_else(|_| "Unknown".to_string()));
|
||||||
tracing::error!("Failed to create audio output: {}", e);
|
|
||||||
return;
|
// Obtenir la config par défaut
|
||||||
|
let config = device
|
||||||
|
.default_output_config()
|
||||||
|
.map_err(|e| AudioError::ProcessingError(format!("Failed to get output config: {}", e)))?;
|
||||||
|
|
||||||
|
let sample_format = config.sample_format();
|
||||||
|
let sample_rate = config.sample_rate().0;
|
||||||
|
let channels = config.channels();
|
||||||
|
|
||||||
|
tracing::debug!(
|
||||||
|
"Output config: {} channels, {} Hz, {:?}",
|
||||||
|
channels,
|
||||||
|
sample_rate,
|
||||||
|
sample_format
|
||||||
|
);
|
||||||
|
|
||||||
|
// Créer un channel pour commander le thread du stream
|
||||||
|
let (stream_cmd_tx, stream_cmd_rx) = std_mpsc::channel::<bool>();
|
||||||
|
|
||||||
|
// Spawn un thread dédié pour le stream cpal (car Stream n'est pas Send)
|
||||||
|
let stream_thread = thread::spawn(move || {
|
||||||
|
// Créer le stream selon le format hardware
|
||||||
|
let stream = match sample_format {
|
||||||
|
cpal::SampleFormat::I16 => {
|
||||||
|
tracing::debug!("Using I16 output format");
|
||||||
|
match device.build_output_stream(
|
||||||
|
&config.into(),
|
||||||
|
move |data: &mut [i16], _: &cpal::OutputCallbackInfo| {
|
||||||
|
let mut buf = buffer_clone.lock().unwrap();
|
||||||
|
|
||||||
|
// Remplir avec des samples convertis
|
||||||
|
for sample in data.iter_mut() {
|
||||||
|
let f32_sample = buf.pop_sample_f32().unwrap_or(0.0);
|
||||||
|
// Convertir F32 [-1.0, 1.0] → I16
|
||||||
|
*sample = (f32_sample * 32767.0).clamp(-32768.0, 32767.0) as i16;
|
||||||
|
}
|
||||||
|
},
|
||||||
|
move |err| {
|
||||||
|
tracing::error!("Audio stream error: {}", err);
|
||||||
|
},
|
||||||
|
None,
|
||||||
|
) {
|
||||||
|
Ok(s) => s,
|
||||||
|
Err(e) => {
|
||||||
|
tracing::error!("Failed to build I16 stream: {}", e);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
cpal::SampleFormat::U16 => {
|
||||||
|
tracing::debug!("Using U16 output format");
|
||||||
|
match device.build_output_stream(
|
||||||
|
&config.into(),
|
||||||
|
move |data: &mut [u16], _: &cpal::OutputCallbackInfo| {
|
||||||
|
let mut buf = buffer_clone.lock().unwrap();
|
||||||
|
|
||||||
|
for sample in data.iter_mut() {
|
||||||
|
let f32_sample = buf.pop_sample_f32().unwrap_or(0.0);
|
||||||
|
// Convertir F32 [-1.0, 1.0] → U16 [0, 65535]
|
||||||
|
*sample = ((f32_sample + 1.0) * 32767.5).clamp(0.0, 65535.0) as u16;
|
||||||
|
}
|
||||||
|
},
|
||||||
|
move |err| {
|
||||||
|
tracing::error!("Audio stream error: {}", err);
|
||||||
|
},
|
||||||
|
None,
|
||||||
|
) {
|
||||||
|
Ok(s) => s,
|
||||||
|
Err(e) => {
|
||||||
|
tracing::error!("Failed to build U16 stream: {}", e);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
cpal::SampleFormat::F32 => {
|
||||||
|
tracing::debug!("Using F32 output format");
|
||||||
|
match device.build_output_stream(
|
||||||
|
&config.into(),
|
||||||
|
move |data: &mut [f32], _: &cpal::OutputCallbackInfo| {
|
||||||
|
let mut buf = buffer_clone.lock().unwrap();
|
||||||
|
|
||||||
|
for sample in data.iter_mut() {
|
||||||
|
*sample = buf.pop_sample_f32().unwrap_or(0.0);
|
||||||
|
}
|
||||||
|
},
|
||||||
|
move |err| {
|
||||||
|
tracing::error!("Audio stream error: {}", err);
|
||||||
|
},
|
||||||
|
None,
|
||||||
|
) {
|
||||||
|
Ok(s) => s,
|
||||||
|
Err(e) => {
|
||||||
|
tracing::error!("Failed to build F32 stream: {}", e);
|
||||||
|
return;
|
||||||
|
}
|
||||||
|
}
|
||||||
|
}
|
||||||
|
_ => {
|
||||||
|
tracing::error!("Unsupported sample format: {:?}", sample_format);
|
||||||
|
return;
|
||||||
}
|
}
|
||||||
};
|
};
|
||||||
|
|
||||||
let sink = match Sink::try_new(&stream_handle) {
|
// Démarrer le stream
|
||||||
Ok(s) => s,
|
if let Err(e) = stream.play() {
|
||||||
Err(e) => {
|
tracing::error!("Failed to start stream: {}", e);
|
||||||
tracing::error!("Failed to create sink: {}", e);
|
return;
|
||||||
return;
|
|
||||||
}
|
|
||||||
};
|
|
||||||
|
|
||||||
sink.set_volume(volume);
|
|
||||||
tracing::debug!("Rodio thread initialized with volume={}", volume);
|
|
||||||
|
|
||||||
// Boucle de traitement des commandes
|
|
||||||
while let Ok(cmd) = cmd_rx.recv() {
|
|
||||||
match cmd {
|
|
||||||
RodioCommand::AppendSamples { samples, sample_rate } => {
|
|
||||||
let buffer = rodio::buffer::SamplesBuffer::new(2, sample_rate, samples);
|
|
||||||
sink.append(buffer);
|
|
||||||
}
|
|
||||||
RodioCommand::WaitUntilEnd => {
|
|
||||||
sink.sleep_until_end();
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
RodioCommand::Stop => {
|
|
||||||
sink.stop();
|
|
||||||
break;
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
}
|
||||||
|
|
||||||
tracing::debug!("Rodio thread exiting");
|
tracing::debug!("Stream thread started");
|
||||||
|
|
||||||
|
// Attendre la commande d'arrêt
|
||||||
|
let _ = stream_cmd_rx.recv();
|
||||||
|
|
||||||
|
// Le stream se fermera automatiquement quand il sera droppé
|
||||||
|
tracing::debug!("Stream thread exiting");
|
||||||
});
|
});
|
||||||
|
|
||||||
tracing::debug!("AudioSink initialized with volume={}", self.volume);
|
tracing::debug!("AudioSink initialized with format {:?}", sample_format);
|
||||||
|
|
||||||
// Boucle de réception et traitement des segments
|
// Boucle de réception et traitement des segments
|
||||||
loop {
|
loop {
|
||||||
// Vérifier si l'arrêt a été demandé
|
// Vérifier si l'arrêt a été demandé
|
||||||
if stop_token.is_cancelled() {
|
if stop_token.is_cancelled() {
|
||||||
tracing::debug!("AudioSinkLogic cancelled");
|
tracing::debug!("AudioSinkLogic cancelled");
|
||||||
let _ = cmd_tx.send(RodioCommand::Stop);
|
let _ = stream_cmd_tx.send(true);
|
||||||
let _ = rodio_thread.join();
|
let _ = stream_thread.join();
|
||||||
return Ok(());
|
return Ok(());
|
||||||
}
|
}
|
||||||
|
|
||||||
// Recevoir le prochain segment
|
// Vérifier si on a fini de jouer
|
||||||
|
{
|
||||||
|
let buf = buffer.lock().unwrap();
|
||||||
|
if buf.is_finished() {
|
||||||
|
tracing::debug!("AudioSink: finished playing all samples");
|
||||||
|
let _ = stream_cmd_tx.send(true);
|
||||||
|
let _ = stream_thread.join();
|
||||||
|
return Ok(());
|
||||||
|
}
|
||||||
|
}
|
||||||
|
|
||||||
|
// Recevoir le prochain segment (avec timeout pour vérifier périodiquement le buffer)
|
||||||
let segment = tokio::select! {
|
let segment = tokio::select! {
|
||||||
result = rx.recv() => {
|
result = rx.recv() => {
|
||||||
match result {
|
match result {
|
||||||
Some(seg) => seg,
|
Some(seg) => seg,
|
||||||
None => {
|
None => {
|
||||||
tracing::debug!("AudioSinkLogic: input channel closed");
|
tracing::debug!("AudioSinkLogic: input channel closed");
|
||||||
let _ = cmd_tx.send(RodioCommand::Stop);
|
// Attendre que le buffer se vide
|
||||||
let _ = rodio_thread.join();
|
while !buffer.lock().unwrap().is_empty() {
|
||||||
|
tokio::time::sleep(tokio::time::Duration::from_millis(10)).await;
|
||||||
|
}
|
||||||
|
let _ = stream_cmd_tx.send(true);
|
||||||
|
let _ = stream_thread.join();
|
||||||
return Ok(());
|
return Ok(());
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
_ = stop_token.cancelled() => {
|
_ = stop_token.cancelled() => {
|
||||||
tracing::debug!("AudioSinkLogic cancelled during recv");
|
tracing::debug!("AudioSinkLogic cancelled during recv");
|
||||||
let _ = cmd_tx.send(RodioCommand::Stop);
|
let _ = stream_cmd_tx.send(true);
|
||||||
let _ = rodio_thread.join();
|
let _ = stream_thread.join();
|
||||||
return Ok(());
|
return Ok(());
|
||||||
}
|
}
|
||||||
|
_ = tokio::time::sleep(tokio::time::Duration::from_millis(100)) => {
|
||||||
|
// Timeout - vérifier le buffer et continuer
|
||||||
|
continue;
|
||||||
|
}
|
||||||
};
|
};
|
||||||
|
|
||||||
// Traiter selon le type de segment
|
// Traiter selon le type de segment
|
||||||
match &segment.segment {
|
match &segment.segment {
|
||||||
crate::_AudioSegment::Chunk(chunk) => {
|
crate::_AudioSegment::Chunk(chunk) => {
|
||||||
// Convertir le chunk en samples rodio
|
// Ajouter le chunk au buffer (pas de conversion ici)
|
||||||
let samples = chunk_to_i16_samples(chunk)?;
|
{
|
||||||
let sample_rate = chunk.sample_rate();
|
let mut buf = buffer.lock().unwrap();
|
||||||
|
buf.push_chunk(chunk.clone());
|
||||||
if samples.is_empty() {
|
|
||||||
continue;
|
|
||||||
}
|
}
|
||||||
|
|
||||||
// Envoyer au thread rodio
|
|
||||||
cmd_tx
|
|
||||||
.send(RodioCommand::AppendSamples {
|
|
||||||
samples,
|
|
||||||
sample_rate,
|
|
||||||
})
|
|
||||||
.map_err(|_| {
|
|
||||||
AudioError::ProcessingError("Rodio thread died".to_string())
|
|
||||||
})?;
|
|
||||||
|
|
||||||
tracing::trace!(
|
tracing::trace!(
|
||||||
"AudioSink: sent chunk with {} frames at {}Hz",
|
"AudioSink: buffered chunk with {} frames at {}Hz",
|
||||||
chunk.len(),
|
chunk.len(),
|
||||||
sample_rate
|
chunk.sample_rate()
|
||||||
);
|
);
|
||||||
}
|
}
|
||||||
crate::_AudioSegment::Sync(marker) => {
|
crate::_AudioSegment::Sync(marker) => {
|
||||||
match **marker {
|
match **marker {
|
||||||
SyncMarker::TrackBoundary { .. } => {
|
SyncMarker::TrackBoundary { .. } => {
|
||||||
tracing::debug!("AudioSink: TrackBoundary received");
|
tracing::debug!("AudioSink: TrackBoundary received");
|
||||||
// Le sink continue automatiquement - pas besoin d'attendre
|
// Le buffer continue automatiquement - pas besoin d'action
|
||||||
// Le buffer interne de rodio gère la transition
|
|
||||||
}
|
}
|
||||||
SyncMarker::EndOfStream => {
|
SyncMarker::EndOfStream => {
|
||||||
tracing::debug!("AudioSink: EndOfStream received, waiting for playback to finish");
|
tracing::debug!("AudioSink: EndOfStream received, waiting for playback to finish");
|
||||||
// Demander au thread rodio d'attendre la fin
|
// Marquer la fin et attendre que le buffer se vide
|
||||||
cmd_tx
|
buffer.lock().unwrap().mark_end();
|
||||||
.send(RodioCommand::WaitUntilEnd)
|
|
||||||
.map_err(|_| {
|
// Attendre que tout soit joué
|
||||||
AudioError::ProcessingError("Rodio thread died".to_string())
|
while !buffer.lock().unwrap().is_finished() {
|
||||||
})?;
|
tokio::time::sleep(tokio::time::Duration::from_millis(10)).await;
|
||||||
// Attendre que le thread termine
|
}
|
||||||
rodio_thread.join().map_err(|_| {
|
|
||||||
AudioError::ProcessingError("Failed to join rodio thread".to_string())
|
let _ = stream_cmd_tx.send(true);
|
||||||
})?;
|
let _ = stream_thread.join();
|
||||||
return Ok(());
|
return Ok(());
|
||||||
}
|
}
|
||||||
SyncMarker::Error(ref message) => {
|
SyncMarker::Error(ref message) => {
|
||||||
@@ -204,7 +420,7 @@ impl NodeLogic for AudioSinkLogic {
|
|||||||
// Continuer la lecture malgré l'erreur
|
// Continuer la lecture malgré l'erreur
|
||||||
}
|
}
|
||||||
_ => {
|
_ => {
|
||||||
// Ignorer les autres sync markers (TopZeroSync, Heartbeat, etc.)
|
// Ignorer les autres sync markers
|
||||||
tracing::trace!("AudioSink: ignoring sync marker");
|
tracing::trace!("AudioSink: ignoring sync marker");
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
@@ -214,69 +430,23 @@ impl NodeLogic for AudioSinkLogic {
|
|||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
/// Convertit un AudioChunk en vecteur de samples i16 stéréo
|
|
||||||
fn chunk_to_i16_samples(chunk: &AudioChunk) -> Result<Vec<i16>, AudioError> {
|
|
||||||
let len = chunk.len();
|
|
||||||
let mut samples = Vec::with_capacity(len * 2); // 2 channels
|
|
||||||
|
|
||||||
match chunk {
|
|
||||||
AudioChunk::I16(data) => {
|
|
||||||
// Format natif - copie directe
|
|
||||||
for frame in data.get_frames() {
|
|
||||||
samples.push(frame[0]);
|
|
||||||
samples.push(frame[1]);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
AudioChunk::I24(data) => {
|
|
||||||
// Convertir de 24-bit vers 16-bit
|
|
||||||
for frame in data.get_frames() {
|
|
||||||
let left = (frame[0].as_i32() >> 8) as i16;
|
|
||||||
let right = (frame[1].as_i32() >> 8) as i16;
|
|
||||||
samples.push(left);
|
|
||||||
samples.push(right);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
AudioChunk::I32(data) => {
|
|
||||||
// Convertir de 32-bit vers 16-bit
|
|
||||||
for frame in data.get_frames() {
|
|
||||||
let left = (frame[0] >> 16) as i16;
|
|
||||||
let right = (frame[1] >> 16) as i16;
|
|
||||||
samples.push(left);
|
|
||||||
samples.push(right);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
AudioChunk::F32(data) => {
|
|
||||||
// Convertir de float32 vers 16-bit
|
|
||||||
for frame in data.get_frames() {
|
|
||||||
let left = (frame[0].clamp(-1.0, 1.0) * 32767.0) as i16;
|
|
||||||
let right = (frame[1].clamp(-1.0, 1.0) * 32767.0) as i16;
|
|
||||||
samples.push(left);
|
|
||||||
samples.push(right);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
AudioChunk::F64(data) => {
|
|
||||||
// Convertir de float64 vers 16-bit
|
|
||||||
for frame in data.get_frames() {
|
|
||||||
let left = (frame[0].clamp(-1.0, 1.0) * 32767.0) as i16;
|
|
||||||
let right = (frame[1].clamp(-1.0, 1.0) * 32767.0) as i16;
|
|
||||||
samples.push(left);
|
|
||||||
samples.push(right);
|
|
||||||
}
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
Ok(samples)
|
|
||||||
}
|
|
||||||
|
|
||||||
// ═══════════════════════════════════════════════════════════════════════════
|
// ═══════════════════════════════════════════════════════════════════════════
|
||||||
// WRAPPER AudioSink - Délègue à Node<AudioSinkLogic>
|
// WRAPPER AudioSink - Délègue à Node<AudioSinkLogic>
|
||||||
// ═══════════════════════════════════════════════════════════════════════════
|
// ═══════════════════════════════════════════════════════════════════════════
|
||||||
|
|
||||||
/// AudioSink - Joue les AudioSegment sur la sortie audio standard
|
/// AudioSink - Joue les AudioSegment sur la sortie audio standard
|
||||||
///
|
///
|
||||||
/// Ce sink utilise rodio pour la lecture audio multiplateforme. Il accepte
|
/// Ce sink utilise cpal pour la lecture audio multiplateforme. Il détecte
|
||||||
/// tous les formats audio (I16, I24, I32, F32, F64) et les convertit
|
/// automatiquement le format supporté par le hardware (I16, F32, U16) et
|
||||||
/// automatiquement en I16 pour la lecture.
|
/// accepte tous les formats audio en entrée (I16, I24, I32, F32, F64).
|
||||||
|
///
|
||||||
|
/// Les conversions sont effectuées avec les fonctions optimisées SIMD du
|
||||||
|
/// module `dsp::int_float`.
|
||||||
|
///
|
||||||
|
/// # Volume
|
||||||
|
///
|
||||||
|
/// Ce sink ne gère PAS le volume. Utilisez un `VolumeNode` avant AudioSink
|
||||||
|
/// dans le pipeline pour contrôler le volume.
|
||||||
///
|
///
|
||||||
/// # Exemple
|
/// # Exemple
|
||||||
///
|
///
|
||||||
@@ -285,15 +455,15 @@ fn chunk_to_i16_samples(chunk: &AudioChunk) -> Result<Vec<i16>, AudioError> {
|
|||||||
/// use tokio_util::sync::CancellationToken;
|
/// use tokio_util::sync::CancellationToken;
|
||||||
///
|
///
|
||||||
/// # async fn example() -> Result<(), Box<dyn std::error::Error>> {
|
/// # async fn example() -> Result<(), Box<dyn std::error::Error>> {
|
||||||
/// let source = FileSource::new("audio.flac").await?;
|
/// let mut source = FileSource::new("audio.flac").await?;
|
||||||
/// let mut sink = AudioSink::new();
|
/// let sink = AudioSink::new();
|
||||||
///
|
///
|
||||||
/// // Connecter la source au sink
|
/// // Connecter la source au sink
|
||||||
/// source.register(Box::new(sink));
|
/// source.register(Box::new(sink));
|
||||||
///
|
///
|
||||||
/// // Démarrer la lecture
|
/// // Démarrer la lecture
|
||||||
/// let stop_token = CancellationToken::new();
|
/// let stop_token = CancellationToken::new();
|
||||||
/// source.run(stop_token).await?;
|
/// Box::new(source).run(stop_token).await?;
|
||||||
/// # Ok(())
|
/// # Ok(())
|
||||||
/// # }
|
/// # }
|
||||||
/// ```
|
/// ```
|
||||||
@@ -302,27 +472,17 @@ pub struct AudioSink {
|
|||||||
}
|
}
|
||||||
|
|
||||||
impl AudioSink {
|
impl AudioSink {
|
||||||
/// Crée un nouveau AudioSink avec volume par défaut (1.0)
|
/// Crée un nouveau AudioSink
|
||||||
pub fn new() -> Self {
|
pub fn new() -> Self {
|
||||||
Self {
|
Self {
|
||||||
inner: Node::new_with_input(AudioSinkLogic::new(), DEFAULT_CHANNEL_SIZE),
|
inner: Node::new_with_input(AudioSinkLogic::new(), DEFAULT_CHANNEL_SIZE),
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
|
|
||||||
/// Crée un nouveau AudioSink avec un volume spécifique (0.0 à 1.0)
|
|
||||||
pub fn with_volume(volume: f32) -> Self {
|
|
||||||
Self {
|
|
||||||
inner: Node::new_with_input(AudioSinkLogic::with_volume(volume), DEFAULT_CHANNEL_SIZE),
|
|
||||||
}
|
|
||||||
}
|
|
||||||
|
|
||||||
/// Crée un nouveau AudioSink avec une taille de channel personnalisée
|
/// Crée un nouveau AudioSink avec une taille de channel personnalisée
|
||||||
pub fn with_channel_size(channel_size: usize, volume: f32) -> Self {
|
pub fn with_channel_size(channel_size: usize) -> Self {
|
||||||
Self {
|
Self {
|
||||||
inner: Node::new_with_input(
|
inner: Node::new_with_input(AudioSinkLogic::new(), channel_size),
|
||||||
AudioSinkLogic::with_volume(volume),
|
|
||||||
channel_size,
|
|
||||||
),
|
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
}
|
}
|
||||||
@@ -369,32 +529,26 @@ mod tests {
|
|||||||
use crate::AudioChunkData;
|
use crate::AudioChunkData;
|
||||||
|
|
||||||
#[test]
|
#[test]
|
||||||
fn test_chunk_to_i16_samples_from_i16() {
|
fn test_chunk_to_f32_interleaved_from_i16() {
|
||||||
let stereo = vec![[100i16, 200i16], [300i16, 400i16]];
|
let stereo = vec![[16384i16, -16384i16], [32767i16, -32768i16]];
|
||||||
let chunk_data = AudioChunkData::new(stereo, 44100, 0.0);
|
let chunk_data = AudioChunkData::new(stereo, 44100, 0.0);
|
||||||
let chunk = AudioChunk::I16(chunk_data);
|
let chunk = AudioChunk::I16(chunk_data);
|
||||||
|
|
||||||
let samples = chunk_to_i16_samples(&chunk).unwrap();
|
let samples = chunk_to_f32_interleaved(&chunk);
|
||||||
assert_eq!(samples, vec![100, 200, 300, 400]);
|
assert_eq!(samples.len(), 4);
|
||||||
|
// Vérifier que les valeurs sont normalisées
|
||||||
|
assert!((samples[0] - 0.5).abs() < 0.01);
|
||||||
|
assert!((samples[1] + 0.5).abs() < 0.01);
|
||||||
}
|
}
|
||||||
|
|
||||||
#[test]
|
#[test]
|
||||||
fn test_chunk_to_i16_samples_from_f32() {
|
fn test_chunk_to_f32_interleaved_from_f32() {
|
||||||
use crate::I24;
|
|
||||||
let stereo = vec![[0.5f32, -0.5f32], [1.0f32, -1.0f32]];
|
let stereo = vec![[0.5f32, -0.5f32], [1.0f32, -1.0f32]];
|
||||||
let chunk_data = AudioChunkData::new(stereo, 48000, 0.0);
|
let chunk_data = AudioChunkData::new(stereo, 48000, 0.0);
|
||||||
let chunk = AudioChunk::F32(chunk_data);
|
let chunk = AudioChunk::F32(chunk_data);
|
||||||
|
|
||||||
let samples = chunk_to_i16_samples(&chunk).unwrap();
|
let samples = chunk_to_f32_interleaved(&chunk);
|
||||||
// 0.5 * 32767 ≈ 16383
|
assert_eq!(samples, vec![0.5, -0.5, 1.0, -1.0]);
|
||||||
// -0.5 * 32767 ≈ -16383
|
|
||||||
// 1.0 * 32767 = 32767
|
|
||||||
// -1.0 * 32767 = -32767
|
|
||||||
assert_eq!(samples.len(), 4);
|
|
||||||
assert!((samples[0] - 16383).abs() <= 1);
|
|
||||||
assert!((samples[1] + 16383).abs() <= 1);
|
|
||||||
assert_eq!(samples[2], 32767);
|
|
||||||
assert_eq!(samples[3], -32767);
|
|
||||||
}
|
}
|
||||||
|
|
||||||
#[test]
|
#[test]
|
||||||
@@ -405,12 +559,6 @@ mod tests {
|
|||||||
assert!(sink.output_type().is_none());
|
assert!(sink.output_type().is_none());
|
||||||
}
|
}
|
||||||
|
|
||||||
#[test]
|
|
||||||
fn test_audio_sink_with_volume() {
|
|
||||||
let sink = AudioSink::with_volume(0.5);
|
|
||||||
assert!(sink.get_tx().is_some());
|
|
||||||
}
|
|
||||||
|
|
||||||
#[test]
|
#[test]
|
||||||
#[should_panic(expected = "terminal node")]
|
#[should_panic(expected = "terminal node")]
|
||||||
fn test_audio_sink_cannot_have_children() {
|
fn test_audio_sink_cannot_have_children() {
|
||||||
@@ -418,4 +566,28 @@ mod tests {
|
|||||||
let another_sink = AudioSink::new();
|
let another_sink = AudioSink::new();
|
||||||
sink.register(Box::new(another_sink));
|
sink.register(Box::new(another_sink));
|
||||||
}
|
}
|
||||||
|
|
||||||
|
#[test]
|
||||||
|
fn test_shared_buffer() {
|
||||||
|
let mut buffer = SharedBuffer::new();
|
||||||
|
|
||||||
|
assert!(buffer.is_empty());
|
||||||
|
assert!(!buffer.is_finished());
|
||||||
|
|
||||||
|
// Test avec un chunk F32
|
||||||
|
let stereo = vec![[0.5f32, -0.5f32]];
|
||||||
|
let chunk_data = AudioChunkData::new(stereo, 48000, 0.0);
|
||||||
|
let chunk = Arc::new(AudioChunk::F32(chunk_data));
|
||||||
|
|
||||||
|
buffer.push_chunk(chunk);
|
||||||
|
assert!(!buffer.is_empty());
|
||||||
|
|
||||||
|
// Pop quelques samples
|
||||||
|
assert_eq!(buffer.pop_sample_f32(), Some(0.5));
|
||||||
|
assert_eq!(buffer.pop_sample_f32(), Some(-0.5));
|
||||||
|
assert_eq!(buffer.pop_sample_f32(), None);
|
||||||
|
|
||||||
|
buffer.mark_end();
|
||||||
|
assert!(buffer.is_finished());
|
||||||
|
}
|
||||||
}
|
}
|
||||||
|
|||||||
@@ -6,6 +6,6 @@ edition = "2024"
|
|||||||
[dependencies]
|
[dependencies]
|
||||||
get_if_addrs = "0.5.3"
|
get_if_addrs = "0.5.3"
|
||||||
os_info = "3.8"
|
os_info = "3.8"
|
||||||
netstat2 = "0.11.2"
|
netstat2 = "0.11"
|
||||||
sysinfo = "0.30"
|
sysinfo = "0.30"
|
||||||
users = "0.11"
|
users = "0.11"
|
||||||
|
|||||||
Reference in New Issue
Block a user