mirror of
https://github.com/jamiepine/voicebox.git
synced 2026-09-17 05:40:42 -07:00
Enhance audio capture functionality and update dependencies
- Added support for capturing system audio on Windows using WASAPI with improved error handling and thread safety. - Introduced the 'scopeguard' crate for better resource management during audio capture. - Updated Cargo.toml to include 'scopeguard' and modified Windows-specific dependencies for enhanced functionality. - Added a new test for validating audio capture output, ensuring the captured audio data is valid and non-empty.
This commit is contained in:
@@ -4,15 +4,16 @@ from PyInstaller.utils.hooks import collect_submodules
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from PyInstaller.utils.hooks import copy_metadata
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datas = []
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hiddenimports = ['backend', 'backend.main', 'backend.config', 'backend.database', 'backend.models', 'backend.profiles', 'backend.history', 'backend.tts', 'backend.transcribe', 'backend.utils.audio', 'backend.utils.cache', 'backend.utils.progress', 'backend.utils.hf_progress', 'backend.utils.validation', 'torch', 'transformers', 'fastapi', 'uvicorn', 'sqlalchemy', 'librosa', 'soundfile', 'qwen_tts', 'qwen_tts.inference', 'qwen_tts.inference.qwen3_tts_model', 'qwen_tts.inference.qwen3_tts_tokenizer', 'qwen_tts.core', 'qwen_tts.cli']
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hiddenimports = ['backend', 'backend.main', 'backend.config', 'backend.database', 'backend.models', 'backend.profiles', 'backend.history', 'backend.tts', 'backend.transcribe', 'backend.utils.audio', 'backend.utils.cache', 'backend.utils.progress', 'backend.utils.hf_progress', 'backend.utils.validation', 'torch', 'transformers', 'fastapi', 'uvicorn', 'sqlalchemy', 'librosa', 'soundfile', 'qwen_tts', 'qwen_tts.inference', 'qwen_tts.inference.qwen3_tts_model', 'qwen_tts.inference.qwen3_tts_tokenizer', 'qwen_tts.core', 'qwen_tts.cli', 'pkg_resources.extern']
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datas += collect_data_files('qwen_tts')
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datas += copy_metadata('qwen-tts')
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hiddenimports += collect_submodules('qwen_tts')
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hiddenimports += collect_submodules('jaraco')
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a = Analysis(
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['server.py'],
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pathex=['/Users/jamespine/Projects/voice/Qwen3-TTS'],
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pathex=['C:\\Users\\ijame\\Projects\\voice\\Qwen3-TTS'],
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binaries=[],
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datas=datas,
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hiddenimports=hiddenimports,
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Generated
+1
@@ -4490,6 +4490,7 @@ dependencies = [
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"coreaudio-sys",
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"hound",
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"objc",
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"scopeguard",
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"screencapturekit",
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"serde",
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"serde_json",
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@@ -22,6 +22,7 @@ serde_json = "1.0"
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tokio = { version = "1", features = ["full"] }
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hound = "3.5"
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base64 = "0.22"
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scopeguard = "1.2.0"
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[target.'cfg(target_os = "macos")'.dependencies]
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screencapturekit = { version = "1", features = ["async"] }
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@@ -31,7 +32,7 @@ core-foundation-sys = "0.8"
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[target.'cfg(target_os = "windows")'.dependencies]
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wasapi = "0.22"
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windows = { version = "0.62", features = ["Win32_Foundation", "Win32_UI_WindowsAndMessaging"] }
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windows = { version = "0.62", features = ["Win32_Foundation", "Win32_UI_WindowsAndMessaging", "Win32_System_Com"] }
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[target.'cfg(not(any(target_os = "android", target_os = "ios")))'.dependencies]
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tauri-plugin-updater = "2.0"
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@@ -5,8 +5,8 @@ use std::io::Cursor;
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use std::sync::Arc;
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use std::sync::atomic::{AtomicBool, Ordering};
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use std::thread;
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use std::time::Duration;
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use wasapi::*;
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use windows::Win32::System::Com::{CoInitializeEx, CoUninitialize, COINIT_MULTITHREADED};
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pub async fn start_capture(
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state: &AudioCaptureState,
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@@ -37,6 +37,20 @@ pub async fn start_capture(
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// Spawn capture task on a dedicated thread (WASAPI COM objects are not Send)
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// All WASAPI objects must be created and used on the same thread
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thread::spawn(move || {
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// Initialize COM for this thread
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unsafe {
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let hr = CoInitializeEx(None, COINIT_MULTITHREADED);
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if hr.is_err() {
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eprintln!("Failed to initialize COM: {:?}", hr);
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return;
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}
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}
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// Ensure COM is uninitialized when thread exits
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let _com_guard = scopeguard::guard((), |_| unsafe {
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CoUninitialize();
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});
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// Initialize WASAPI on this thread
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let device = match DeviceEnumerator::new()
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.and_then(|enumerator| enumerator.get_default_device(&Direction::Render))
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@@ -76,10 +90,21 @@ pub async fn start_capture(
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*sample_rate_arc.lock().unwrap() = mix_format.get_samplespersec();
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*channels_arc.lock().unwrap() = mix_format.get_nchannels();
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// Get device period
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let (_def_period, min_period) = match audio_client.get_device_period() {
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Ok(periods) => periods,
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Err(e) => {
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eprintln!("Failed to get device period: {}", e);
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return;
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}
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};
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// Initialize audio client for loopback with StreamMode
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// For loopback mode: get Render device, initialize with Capture direction
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// This triggers AUDCLNT_STREAMFLAGS_LOOPBACK in the wasapi crate
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let stream_mode = StreamMode::EventsShared {
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autoconvert: false,
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buffer_duration_hns: 0, // 0 = use default buffer size
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autoconvert: true, // Enable automatic format conversion
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buffer_duration_hns: min_period, // Use minimum period
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};
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if let Err(e) = audio_client.initialize_client(&mix_format, &Direction::Capture, &stream_mode) {
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@@ -89,6 +114,15 @@ pub async fn start_capture(
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return;
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}
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// Set up event handle for EventsShared mode
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let h_event = match audio_client.set_get_eventhandle() {
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Ok(event) => event,
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Err(e) => {
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eprintln!("Failed to set event handle: {}", e);
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return;
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}
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};
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let capture_client = match audio_client.get_audiocaptureclient() {
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Ok(client) => client,
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Err(e) => {
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@@ -158,8 +192,10 @@ pub async fn start_capture(
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}
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}
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// Sleep briefly to avoid busy-waiting
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thread::sleep(Duration::from_millis(10));
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// Wait for event signal (with timeout to allow checking stop flag)
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if h_event.wait_for_event(100).is_err() {
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// Timeout is expected - just continue to check stop flag
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}
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}
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// Stop the stream when done
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@@ -0,0 +1 @@
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pub mod audio_capture;
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@@ -0,0 +1,59 @@
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// NOTE: This test requires system audio to be playing during execution.
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// To run this test successfully:
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// 1. Start playing audio (music, video, etc.)
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// 2. Run: cargo test --test audio_capture_test -- --nocapture
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// 3. The test will capture audio for 5 seconds and verify the output
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use voicebox::audio_capture::{AudioCaptureState, start_capture, stop_capture};
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use base64::Engine;
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#[tokio::test]
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async fn test_system_audio_capture() {
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// Create AudioCaptureState
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let state = AudioCaptureState::new();
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println!("Starting system audio capture with 5 second max duration...");
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// Start capture with 5 second max duration
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let result = start_capture(&state, 5).await;
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if let Err(e) = result {
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panic!("Failed to start capture: {}", e);
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}
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println!("Capture started, waiting 5 seconds...");
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// Wait 5 seconds for capture to complete
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tokio::time::sleep(tokio::time::Duration::from_secs(5)).await;
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println!("Stopping capture...");
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// Stop capture and get the result
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let audio_data = stop_capture(&state).await;
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match audio_data {
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Ok(base64_wav) => {
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println!("Capture stopped successfully");
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// Validate the returned base64 WAV data
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println!("Validating base64 WAV data...");
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// Decode base64 to bytes
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let decoded_bytes = base64::engine::general_purpose::STANDARD
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.decode(&base64_wav)
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.expect("Failed to decode base64 data");
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// Verify bytes array is not empty
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assert!(!decoded_bytes.is_empty(), "Decoded bytes array is empty");
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// Confirm data has content (length > 0)
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println!("WAV data length: {} bytes", decoded_bytes.len());
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assert!(decoded_bytes.len() > 0, "WAV data has no content");
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println!("✓ Test passed: Audio capture produced valid WAV data");
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}
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Err(e) => {
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panic!("Failed to stop capture or get audio data: {}", e);
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}
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}
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}
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