Support loopback recording on macOS

.gitignore

@@ -4,3 +4,4 @@
 .DS_Store
 recorded.wav
 rls*.log
+/.direnv

CHANGELOG.md

@@ -10,6 +10,7 @@
 - CoreAudio: Change `Device::supported_configs` to return a single element containing the available sample rate range when all elements have the same `mMinimum` and `mMaximum` values.
 - CoreAudio: Change default audio device detection to be lazy when building a stream, instead of during device enumeration.
 - CoreAudio: Add `i8`, `i32` and `I24` sample format support (24-bit samples stored in 4 bytes).
+- CoreAudio: Add support for loopback recording (recording system audio output) on macOS.
 - iOS: Fix example by properly activating audio session.
 - WASAPI: Expose `IMMDevice` from WASAPI host Device.
 - WASAPI: Add `I24` and `U24` sample format support (24-bit samples stored in 4 bytes).

Cargo.toml

@@ -10,7 +10,10 @@ edition = "2021"
 rust-version = "1.70"
 
 [features]
-asio = ["asio-sys", "num-traits"] # Only available on Windows. See README for setup instructions.
+asio = [
+    "asio-sys",
+    "num-traits",
+] # Only available on Windows. See README for setup instructions.
 
 # Deprecated, the `oboe` backend has been removed
 oboe-shared-stdcxx = []
@@ -36,8 +39,8 @@ windows = { version = "0.54.0", features = [
     "Win32_System_SystemServices",
     "Win32_System_Variant",
     "Win32_Media_Multimedia",
-    "Win32_UI_Shell_PropertiesSystem"
-]}
+    "Win32_UI_Shell_PropertiesSystem",
+] }
 audio_thread_priority = { version = "0.33.0", optional = true }
 asio-sys = { version = "0.2", path = "asio-sys", optional = true }
 num-traits = { version = "0.2.6", optional = true }
@@ -60,6 +63,8 @@ objc2-core-audio = { version = "0.3.1", default-features = false, features = [
     "std",
     "AudioHardware",
     "AudioHardwareDeprecated",
+    "objc2",
+    "objc2-foundation",
 ] }
 objc2-audio-toolbox = { version = "0.3.1", default-features = false, features = [
     "std",
@@ -70,20 +75,44 @@ objc2-core-audio-types = { version = "0.3.1", default-features = false, features
     "std",
     "CoreAudioBaseTypes",
 ] }
+objc2-core-foundation = { version = "0.3.1" }
+objc2-foundation = { version = "0.3.1" }
+objc2 = { version = "0.6.2" }
 
 [target.'cfg(target_os = "emscripten")'.dependencies]
 wasm-bindgen = { version = "0.2.89" }
 wasm-bindgen-futures = "0.4.33"
 js-sys = { version = "0.3.35" }
-web-sys = { version = "0.3.35", features = [ "AudioContext", "AudioContextOptions", "AudioBuffer", "AudioBufferSourceNode", "AudioNode",  "AudioDestinationNode", "Window", "AudioContextState"] }
+web-sys = { version = "0.3.35", features = [
+    "AudioContext",
+    "AudioContextOptions",
+    "AudioBuffer",
+    "AudioBufferSourceNode",
+    "AudioNode",
+    "AudioDestinationNode",
+    "Window",
+    "AudioContextState",
+] }
 
 [target.'cfg(all(target_arch = "wasm32", target_os = "unknown"))'.dependencies]
 wasm-bindgen = { version = "0.2.58", optional = true }
 js-sys = { version = "0.3.35" }
-web-sys = { version = "0.3.35", features = [ "AudioContext", "AudioContextOptions", "AudioBuffer", "AudioBufferSourceNode", "AudioNode",  "AudioDestinationNode", "Window", "AudioContextState"] }
+web-sys = { version = "0.3.35", features = [
+    "AudioContext",
+    "AudioContextOptions",
+    "AudioBuffer",
+    "AudioBufferSourceNode",
+    "AudioNode",
+    "AudioDestinationNode",
+    "Window",
+    "AudioContextState",
+] }
 
 [target.'cfg(target_os = "android")'.dependencies]
-ndk = { version = "0.9", default-features = false, features = ["audio", "api-level-26"]}
+ndk = { version = "0.9", default-features = false, features = [
+    "audio",
+    "api-level-26",
+] }
 ndk-context = "0.1"
 jni = "0.21"
 num-derive = "0.4"

examples/record_wav.rs

@@ -12,10 +12,16 @@ use std::sync::{Arc, Mutex};
 #[derive(Parser, Debug)]
 #[command(version, about = "CPAL record_wav example", long_about = None)]
 struct Opt {
-    /// The audio device to use
+    /// The audio device to use.
+    /// For the default microphone, use "default".
+    /// For recording system output, use "default-output".
     #[arg(short, long, default_value_t = String::from("default"))]
     device: String,
 
+    /// How long to record, in seconds
+    #[arg(long, default_value_t = 3)]
+    duration: u64,
+
     /// Use the JACK host
     #[cfg(all(
         any(
@@ -69,20 +75,24 @@ fn main() -> Result<(), anyhow::Error> {
     let host = cpal::default_host();
 
     // Set up the input device and stream with the default input config.
-    let device = if opt.device == "default" {
-        host.default_input_device()
-    } else {
-        host.input_devices()?
-            .find(|x| x.name().map(|y| y == opt.device).unwrap_or(false))
+    let device = match opt.device.as_str() {
+        "default" => host.default_input_device(),
+        "default-output" => host.default_output_device(),
+        name => host
+            .input_devices()?
+            .find(|x| x.name().map(|y| y == name).unwrap_or(false)),
     }
     .expect("failed to find input device");
 
     println!("Input device: {}", device.name()?);
 
-    let config = device
-        .default_input_config()
-        .expect("Failed to get default input config");
-    println!("Default input config: {config:?}");
+    let config = if device.supports_input() {
+        device.default_input_config()
+    } else {
+        device.default_output_config()
+    }
+    .expect("Failed to get default input/output config");
+    println!("Default input/output config: {config:?}");
 
     // The WAV file we're recording to.
     const PATH: &str = concat!(env!("CARGO_MANIFEST_DIR"), "/recorded.wav");
@@ -135,7 +145,7 @@ fn main() -> Result<(), anyhow::Error> {
     stream.play()?;
 
     // Let recording go for roughly three seconds.
-    std::thread::sleep(std::time::Duration::from_secs(3));
+    std::thread::sleep(std::time::Duration::from_secs(opt.duration));
     drop(stream);
     writer.lock().unwrap().take().unwrap().finalize()?;
     println!("Recording {PATH} complete!");