win_audio: Upstream Windows audio backend code

The most notable changes are:

* Playback Async support
* Audio Client flags to hide app from SndVol when audio session has
  expired
* Updates to audio_streams and cros_async crates for playback async
  support

BUG=b:256655413
TEST=Verified to work downstream

Change-Id: Ifbe9a15791feaa41d6e1d5eaf2c5824b2c7c25b8
Reviewed-on: https://chromium-review.googlesource.com/c/crosvm/crosvm/+/3994882
Commit-Queue: Richard Zhang <rizhang@google.com>
Reviewed-by: Noah Gold <nkgold@google.com>

Cargo.lock

@@ -2191,8 +2191,10 @@ name = "win_audio"
 version = "0.1.0"
 dependencies = [
  "anyhow",
+ "async-trait",
  "audio_streams",
  "base",
+ "cros_async",
  "libc",
  "metrics",
  "once_cell",

common/audio_streams/src/async_api.rs

@@ -15,6 +15,8 @@
 use std::io::Result;
 #[cfg(unix)]
 use std::os::unix::net::UnixStream;
+#[cfg(windows)]
+use std::os::windows::io::RawHandle;
 use std::time::Duration;
 
 use async_trait::async_trait;
@@ -45,6 +47,13 @@ pub trait WriteAsync {
     ) -> Result<(usize, Vec<u8>)>;
 }
 
+/// Trait to wrap around EventAsync, because audio_streams can't depend on anything in `base` or
+/// `cros_async`.
+#[async_trait(?Send)]
+pub trait EventAsyncWrapper {
+    async fn wait(&self) -> Result<u64>;
+}
+
 pub trait ReadWriteAsync: ReadAsync + WriteAsync {}
 
 pub type AsyncStream = Box<dyn ReadWriteAsync + Send>;
@@ -56,6 +65,15 @@ pub trait AudioStreamsExecutor {
     #[cfg(unix)]
     fn async_unix_stream(&self, f: UnixStream) -> Result<AsyncStream>;
 
+    /// Wraps an event that will be triggered when the audio backend is ready to read more audio
+    /// data.
+    ///
+    /// # Safety
+    /// Callers needs to make sure the `RawHandle` is an Event, or at least a valid Handle for
+    /// their use case.
+    #[cfg(windows)]
+    unsafe fn async_event(&self, event: RawHandle) -> Result<Box<dyn EventAsyncWrapper>>;
+
     /// Returns a future that resolves after the specified time.
     async fn delay(&self, dur: Duration) -> Result<()>;
 }
@@ -79,5 +97,9 @@ pub mod test {
             std::thread::sleep(dur);
             return Ok(());
         }
+        #[cfg(windows)]
+        unsafe fn async_event(&self, _event: RawHandle) -> Result<Box<dyn EventAsyncWrapper>> {
+            unimplemented!("async_event is not yet implemented on windows");
+        }
     }
 }

cros_async/src/audio_streams_async.rs

@@ -9,18 +9,24 @@
 use std::io::Result;
 #[cfg(unix)]
 use std::os::unix::net::UnixStream;
+#[cfg(windows)]
+use std::os::windows::io::RawHandle;
 use std::time::Duration;
 
 use async_trait::async_trait;
 #[cfg(unix)]
 use audio_streams::async_api::AsyncStream;
 use audio_streams::async_api::AudioStreamsExecutor;
+use audio_streams::async_api::EventAsyncWrapper;
 use audio_streams::async_api::ReadAsync;
 use audio_streams::async_api::ReadWriteAsync;
 use audio_streams::async_api::WriteAsync;
+#[cfg(windows)]
+use base::{Event, FromRawDescriptor};
 
 #[cfg(unix)]
 use super::AsyncWrapper;
+use crate::EventAsync;
 use crate::IntoAsync;
 use crate::IoSourceExt;
 use crate::TimerAsync;
@@ -61,6 +67,13 @@ impl<T: IntoAsync + Send> WriteAsync for IoSourceWrapper<T> {
 #[async_trait(?Send)]
 impl<T: IntoAsync + Send> ReadWriteAsync for IoSourceWrapper<T> {}
 
+#[async_trait(?Send)]
+impl EventAsyncWrapper for EventAsync {
+    async fn wait(&self) -> Result<u64> {
+        self.next_val().await.map_err(Into::into)
+    }
+}
+
 #[async_trait(?Send)]
 impl AudioStreamsExecutor for super::Executor {
     #[cfg(unix)]
@@ -70,6 +83,16 @@ impl AudioStreamsExecutor for super::Executor {
         }));
     }
 
+    /// # Safety
+    /// This is only safe if `event` is a handle to a Windows Event.
+    #[cfg(windows)]
+    unsafe fn async_event(&self, event: RawHandle) -> Result<Box<dyn EventAsyncWrapper>> {
+        Ok(Box::new(
+            EventAsync::new(Event::from_raw_descriptor(event), self)
+                .map_err(std::io::Error::from)?,
+        ))
+    }
+
     async fn delay(&self, dur: Duration) -> Result<()> {
         TimerAsync::sleep(self, dur).await.map_err(Into::into)
     }

win_audio/Cargo.toml

@@ -5,8 +5,10 @@ authors = ["The Chromium OS Authors"]
 edition = "2021"
 
 [target.'cfg(windows)'.dependencies]
+async-trait = "0.1.36"
 audio_streams = { path = "../common/audio_streams"}
 base = { path = "../base" }
+cros_async = { path = "../cros_async" }
 libc = "*"
 win_util = { path = "../win_util" }
 winapi = "*"

win_audio/src/lib.rs

@@ -2,7 +2,6 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
-#![cfg(windows)]
 #![allow(non_upper_case_globals)]
 include!(concat!(
     env!("CARGO_MANIFEST_DIR"),
@@ -25,18 +24,37 @@ mod win_audio_impl;
 use std::error;
 use std::sync::Arc;
 
+use audio_streams::AsyncPlaybackBufferStream;
 use audio_streams::NoopStream;
 use audio_streams::NoopStreamSource;
+use audio_streams::NoopStreamSourceGenerator;
 use audio_streams::PlaybackBufferStream;
 use audio_streams::SampleFormat;
 use audio_streams::StreamSource;
 use base::info;
 use base::warn;
 use sync::Mutex;
-use win_audio_impl::*;
+use win_audio_impl::async_stream::WinAudioStreamSourceGenerator;
+pub use win_audio_impl::*;
 
 pub type BoxError = Box<dyn error::Error + Send + Sync>;
 
+pub trait WinStreamSourceGenerator: Send + Sync {
+    fn generate(&self) -> Result<Box<dyn WinAudioServer>, BoxError>;
+}
+
+impl WinStreamSourceGenerator for WinAudioStreamSourceGenerator {
+    fn generate(&self) -> std::result::Result<Box<dyn WinAudioServer>, BoxError> {
+        Ok(Box::new(WinAudio::new()?))
+    }
+}
+
+impl WinStreamSourceGenerator for NoopStreamSourceGenerator {
+    fn generate(&self) -> Result<Box<dyn WinAudioServer>, BoxError> {
+        Ok(Box::new(NoopStreamSource))
+    }
+}
+
 /// Contains information about the audio engine's properties, such as its audio sample format
 /// and its period in frames.
 ///
@@ -66,9 +84,22 @@ pub trait WinAudioServer: StreamSource {
         buffer_size: usize,
     ) -> Result<(Arc<Mutex<Box<dyn PlaybackBufferStream>>>, AudioSharedFormat), BoxError>;
 
+    fn new_async_playback_stream_and_get_shared_format(
+        &mut self,
+        _num_channels: usize,
+        _format: SampleFormat,
+        _frame_rate: usize,
+        _buffer_size: usize,
+        _ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<(Box<dyn AsyncPlaybackBufferStream>, AudioSharedFormat), BoxError> {
+        unimplemented!()
+    }
+
     /// Evict the playback stream cache so that the audio device can be released, thus allowing
     /// for machines to go to sleep.
-    fn evict_playback_stream_cache(&mut self);
+    fn evict_playback_stream_cache(&mut self) {
+        unimplemented!()
+    }
 
     /// Returns true if audio server is a noop stream. This determine if evicting a cache is worth
     /// doing
@@ -140,6 +171,57 @@ impl WinAudioServer for WinAudio {
         Ok((playback_buffer_stream, audio_shared_format))
     }
 
+    fn new_async_playback_stream_and_get_shared_format(
+        &mut self,
+        num_channels: usize,
+        _format: SampleFormat,
+        frame_rate: usize,
+        buffer_size: usize,
+        ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<(Box<dyn AsyncPlaybackBufferStream>, AudioSharedFormat), BoxError> {
+        let hr = WinAudio::co_init_once_per_thread();
+        let _ = check_hresult!(hr, RenderError::from(hr), "Co Initialized failed");
+
+        let (async_playback_buffer_stream, audio_shared_format): (
+            Box<dyn AsyncPlaybackBufferStream>,
+            AudioSharedFormat,
+        ) = match win_audio_impl::WinAudioRenderer::new_async(
+            num_channels,
+            frame_rate as u32,
+            buffer_size,
+            ex,
+        ) {
+            Ok(renderer) => {
+                let audio_shared_format = renderer.device.audio_shared_format;
+                let renderer_box = Box::new(renderer) as Box<dyn AsyncPlaybackBufferStream>;
+                (renderer_box, audio_shared_format)
+            }
+            Err(e) => {
+                warn!(
+                    "Failed to create WinAudioRenderer. Fallback to NoopStream with error: {}",
+                    e
+                );
+                (
+                    Box::new(NoopStream::new(
+                        num_channels,
+                        SampleFormat::S16LE,
+                        frame_rate as u32,
+                        buffer_size,
+                    )),
+                    AudioSharedFormat {
+                        bit_depth: 16,
+                        frame_rate,
+                        channels: 2,
+                        shared_audio_engine_period_in_frames: frame_rate / 100,
+                        channel_mask: None,
+                    },
+                )
+            }
+        };
+
+        Ok((async_playback_buffer_stream, audio_shared_format))
+    }
+
     fn evict_playback_stream_cache(&mut self) {
         self.cached_playback_buffer_stream = None;
     }
@@ -172,10 +254,6 @@ impl WinAudioServer for NoopStreamSource {
         ))
     }
 
-    fn evict_playback_stream_cache(&mut self) {
-        unimplemented!()
-    }
-
     fn is_noop_stream(&self) -> bool {
         true
     }

win_audio/src/win_audio_impl/async_stream.rs

@@ -0,0 +1,254 @@
+// Copyright 2022 The ChromiumOS Authors
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+
+use std::mem::MaybeUninit;
+
+use async_trait::async_trait;
+use audio_streams::AsyncBufferCommit;
+use audio_streams::AsyncPlaybackBuffer;
+use audio_streams::AsyncPlaybackBufferStream;
+use audio_streams::BoxError;
+use audio_streams::NoopStream;
+use audio_streams::NoopStreamControl;
+use audio_streams::SampleFormat;
+use audio_streams::StreamControl;
+use audio_streams::StreamSource;
+use audio_streams::StreamSourceGenerator;
+use base::error;
+use base::warn;
+
+use crate::DeviceRenderer;
+use crate::RenderError;
+use crate::WinAudio;
+use crate::WinAudioRenderer;
+
+pub struct WinAudioStreamSourceGenerator {}
+
+impl StreamSourceGenerator for WinAudioStreamSourceGenerator {
+    fn generate(&self) -> std::result::Result<Box<dyn StreamSource>, BoxError> {
+        Ok(Box::new(WinAudio::new()?))
+    }
+}
+
+impl WinAudio {
+    pub(super) fn new_async_playback_stream_helper(
+        num_channels: usize,
+        _format: SampleFormat,
+        frame_rate: u32,
+        buffer_size: usize,
+        ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<
+        (
+            Box<dyn StreamControl>,
+            Box<dyn audio_streams::AsyncPlaybackBufferStream>,
+        ),
+        BoxError,
+    > {
+        let hr = WinAudio::co_init_once_per_thread();
+        let _ = check_hresult!(hr, RenderError::from(hr), "Co Initialized failed");
+
+        let playback_buffer_stream: Box<dyn AsyncPlaybackBufferStream> =
+            match WinAudioRenderer::new_async(num_channels, frame_rate, buffer_size, ex) {
+                Ok(renderer) => Box::new(renderer),
+                Err(e) => {
+                    warn!(
+                        "Failed to create WinAudioRenderer. Fallback to NoopStream with error: {}",
+                        e
+                    );
+                    Box::new(NoopStream::new(
+                        num_channels,
+                        SampleFormat::S16LE,
+                        frame_rate,
+                        buffer_size,
+                    ))
+                }
+            };
+
+        Ok((Box::new(NoopStreamControl::new()), playback_buffer_stream))
+    }
+}
+
+impl WinAudioRenderer {
+    /// Constructor to allow for async audio backend.
+    pub fn new_async(
+        num_channels: usize,
+        frame_rate: u32,
+        incoming_buffer_size_in_frames: usize,
+        ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<Self, RenderError> {
+        let device = Self::create_device_renderer_and_log_time(
+            num_channels,
+            frame_rate,
+            incoming_buffer_size_in_frames,
+            Some(ex),
+        )?;
+        Ok(Self {
+            device,
+            num_channels,
+            frame_rate,                     // guest frame rate
+            incoming_buffer_size_in_frames, // from the guest`
+        })
+    }
+}
+
+#[async_trait(?Send)]
+impl AsyncPlaybackBufferStream for WinAudioRenderer {
+    async fn next_playback_buffer<'a>(
+        &'a mut self,
+        _ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<audio_streams::AsyncPlaybackBuffer<'a>, BoxError> {
+        for _ in 0..Self::MAX_REATTACH_TRIES {
+            match self.device.async_next_win_buffer().await {
+                Ok(_) => {
+                    return self
+                        .device
+                        .async_playback_buffer()
+                        .map_err(|e| Box::new(e) as _)
+                }
+                // If the audio device was disconnected, set up whatever is now the default device
+                // and loop to try again.
+                Err(RenderError::DeviceInvalidated) => {
+                    warn!("Audio device disconnected, switching to new default device");
+                    self.reattach_device()?;
+                }
+                Err(e) => return Err(Box::new(e)),
+            }
+        }
+        error!("Unable to attach to a working audio device, giving up");
+        Err(Box::new(RenderError::DeviceInvalidated))
+    }
+}
+
+impl DeviceRenderer {
+    /// Similiar to `next_win_buffer`, this is the async version that will return a wraper around
+    /// the WASAPI buffer.
+    ///
+    /// Unlike `next_win_buffer`, there is no timeout if `async_ready_to_read_event` doesn't fire.
+    /// This should be fine, since the end result with or without the timeout will be no audio.
+    async fn async_next_win_buffer(&mut self) -> Result<(), RenderError> {
+        self.win_buffer = MaybeUninit::uninit().as_mut_ptr();
+
+        // We will wait for windows to tell us when it is ready to take in the next set of
+        // audio samples from the guest
+        loop {
+            let async_ready_to_read_event = self
+                .async_ready_to_read_event
+                .as_ref()
+                .ok_or(RenderError::MissingEventAsync)?;
+            async_ready_to_read_event.wait().await.map_err(|e| {
+                RenderError::AsyncError(
+                    e,
+                    "Failed to wait for async event to get next playback buffer.".to_string(),
+                )
+            })?;
+
+            if self.enough_available_frames()? {
+                break;
+            }
+        }
+
+        self.get_buffer()?;
+
+        Ok(())
+    }
+
+    /// Similar to `playback_buffer`. This will return an `AsyncPlaybackBuffer` that allows for
+    /// async operations.
+    ///
+    /// Due to the way WASAPI is, `AsyncPlaybackBuffer` won't actually have any async operations
+    /// that will await.
+    fn async_playback_buffer(&mut self) -> Result<AsyncPlaybackBuffer, RenderError> {
+        // Safe because `win_buffer` is allocated and retrieved from WASAPI. The size requested,
+        // which we specified in `next_win_buffer` is exactly
+        // `shared_audio_engine_period_in_frames`, so the size parameter should be valid.
+        let (frame_size_bytes, buffer_slice) = unsafe {
+            Self::get_frame_size_and_buffer_slice(
+                self.audio_shared_format.bit_depth as usize,
+                self.audio_shared_format.channels as usize,
+                self.win_buffer,
+                self.audio_shared_format
+                    .shared_audio_engine_period_in_frames,
+            )?
+        };
+
+        AsyncPlaybackBuffer::new(frame_size_bytes, buffer_slice, self)
+            .map_err(RenderError::PlaybackBuffer)
+    }
+}
+
+#[async_trait(?Send)]
+impl AsyncBufferCommit for DeviceRenderer {
+    async fn commit(&mut self, nframes: usize) {
+        // Safe because `audio_render_client` is initialized and parameters passed
+        // into `ReleaseBuffer()` are valid
+        unsafe {
+            let hr = self.audio_render_client.ReleaseBuffer(nframes as u32, 0);
+            let _ = check_hresult!(
+                hr,
+                RenderError::from(hr),
+                "Audio Render Client ReleaseBuffer() failed"
+            );
+        }
+    }
+}
+
+#[cfg(test)]
+mod tests {
+    use super::*;
+
+    use cros_async::Executor;
+
+    // This test is meant to run through the normal audio playback procedure in order to make
+    // debugging easier. The test needs to be ran with a playback device format of 48KHz,
+    // stereo, 16bit. This test might not pass on AMD, since its period is 513 instead of 480.
+    #[ignore]
+    #[test]
+    fn test_async() {
+        async fn test(ex: &Executor) {
+            let stream_source_generator: Box<dyn StreamSourceGenerator> =
+                Box::new(WinAudioStreamSourceGenerator {});
+            let mut stream_source = stream_source_generator
+                .generate()
+                .expect("Failed to create stream source.");
+
+            let (_, mut async_pb_stream) = stream_source
+                .new_async_playback_stream(2, SampleFormat::S16LE, 48000, 480, ex)
+                .expect("Failed to create async playback stream.");
+
+            // The `ex` here won't be used, but it will satisfy the trait function requirement.
+            let mut async_pb_buffer = async_pb_stream
+                .next_playback_buffer(ex)
+                .await
+                .expect("Failed to get next playback buffer");
+
+            // The buffer size is calculated by "period * channels * bit depth". The actual buffer
+            // from `next_playback_buffer` may vary, depending on the device format and the user's
+            // machine.
+            let buffer = [1u8; 480 * 2 * 2];
+
+            async_pb_buffer
+                .copy_cb(buffer.len(), |out| out.copy_from_slice(&buffer))
+                .unwrap();
+
+            async_pb_buffer.commit().await;
+
+            let mut async_pb_buffer = async_pb_stream
+                .next_playback_buffer(ex)
+                .await
+                .expect("Failed to get next playback buffer");
+
+            let buffer = [1u8; 480 * 2 * 2];
+
+            async_pb_buffer
+                .copy_cb(buffer.len(), |out| out.copy_from_slice(&buffer))
+                .unwrap();
+
+            async_pb_buffer.commit().await;
+        }
+
+        let ex = Executor::new().expect("Failed to create executor.");
+
+        ex.run_until(test(&ex)).unwrap();
+    }
+}

win_audio/src/win_audio_impl/mod.rs

@@ -2,9 +2,12 @@
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+pub mod async_stream;
+mod completion_handler;
+mod wave_format;
+
 use std::convert::From;
 use std::fmt::Debug;
-use std::mem::MaybeUninit;
 use std::os::raw::c_void;
 use std::ptr::null_mut;
 use std::sync::Arc;
@@ -12,6 +15,8 @@ use std::sync::Once;
 use std::thread_local;
 use std::time::Duration;
 
+use audio_streams::async_api::EventAsyncWrapper;
+use audio_streams::AsyncPlaybackBufferStream;
 use audio_streams::BoxError;
 use audio_streams::BufferCommit;
 use audio_streams::NoopStream;
@@ -44,6 +49,8 @@ use winapi::shared::mmreg::WAVEFORMATEX;
 use winapi::shared::winerror::S_FALSE;
 use winapi::shared::winerror::S_OK;
 use winapi::um::audioclient::*;
+use winapi::um::audiosessiontypes::AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED;
+use winapi::um::audiosessiontypes::AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED;
 use winapi::um::audiosessiontypes::AUDCLNT_SHAREMODE_SHARED;
 use winapi::um::audiosessiontypes::AUDCLNT_STREAMFLAGS_EVENTCALLBACK;
 use winapi::um::combaseapi::*;
@@ -62,9 +69,6 @@ use wio::com::ComPtr;
 
 use crate::AudioSharedFormat;
 
-mod completion_handler;
-mod wave_format;
-
 const READY_TO_READ_TIMEOUT_MS: u32 = 2000;
 pub const STEREO_CHANNEL_COUNT: u16 = 2;
 pub const MONO_CHANNEL_COUNT: u16 = 1;
@@ -116,9 +120,9 @@ impl WinAudio {
 }
 
 impl StreamSource for WinAudio {
-    // Returns a stream control and a buffer generator object. The stream control object is not used.
-    // The buffer generator object is a wrapper around WASAPI's objects that will create a buffer for
-    // crosvm to copy audio bytes into.
+    /// Returns a stream control and a buffer generator object. The stream control object is not
+    /// used. The buffer generator object is a wrapper around WASAPI's objects that will create a
+    /// buffer for crosvm to copy audio bytes into.
     fn new_playback_stream(
         &mut self,
         num_channels: usize,
@@ -148,6 +152,25 @@ impl StreamSource for WinAudio {
 
         Ok((Box::new(NoopStreamControl::new()), playback_buffer_stream))
     }
+
+    /// Similar to `new_playback_stream, but will return an `AsyncPlaybackBufferStream` that can
+    /// run async operations.
+    fn new_async_playback_stream(
+        &mut self,
+        num_channels: usize,
+        format: SampleFormat,
+        frame_rate: u32,
+        buffer_size: usize,
+        ex: &dyn audio_streams::AudioStreamsExecutor,
+    ) -> Result<(Box<dyn StreamControl>, Box<dyn AsyncPlaybackBufferStream>), BoxError> {
+        WinAudio::new_async_playback_stream_helper(
+            num_channels,
+            format,
+            frame_rate,
+            buffer_size,
+            ex,
+        )
+    }
 }
 
 /// Proxy for a `DeviceRenderer` that handles device invalidated errors by switching to a new
@@ -160,20 +183,21 @@ pub(crate) struct WinAudioRenderer {
 }
 
 impl WinAudioRenderer {
+    const MAX_REATTACH_TRIES: usize = 50;
+
     // Initializes WASAPI objects needed for audio.
     pub fn new(
         num_channels: usize,
         frame_rate: u32,
         incoming_buffer_size_in_frames: usize,
     ) -> Result<Self, RenderError> {
-        let start = std::time::Instant::now();
-        let device = DeviceRenderer::new(num_channels, frame_rate, incoming_buffer_size_in_frames)?;
-        // This can give us insights to how other long other machines take to intialize audio.
-        // Eventually this should be a histogram metric.
-        info!(
-            "DeviceRenderer took {}ms to initialize audio.",
-            start.elapsed().as_millis()
-        );
+        let device = Self::create_device_renderer_and_log_time(
+            num_channels,
+            frame_rate,
+            incoming_buffer_size_in_frames,
+            None,
+        )?;
+
         Ok(Self {
             device,
             num_channels,
@@ -182,6 +206,24 @@ impl WinAudioRenderer {
         })
     }
 
+    fn create_device_renderer_and_log_time(
+        num_channels: usize,
+        frame_rate: u32,
+        incoming_buffer_size_in_frames: usize,
+        ex: Option<&dyn audio_streams::AudioStreamsExecutor>,
+    ) -> Result<DeviceRenderer, RenderError> {
+        let start = std::time::Instant::now();
+        let device =
+            DeviceRenderer::new(num_channels, frame_rate, incoming_buffer_size_in_frames, ex)?;
+        // This can give us insights to how other long other machines take to initialize audio.
+        // Eventually this should be a histogram metric.
+        info!(
+            "DeviceRenderer took {}ms to initialize audio.",
+            start.elapsed().as_millis()
+        );
+        Ok(device)
+    }
+
     // Drops the existing DeviceRenderer and initializes a new DeviceRenderer for the default
     // device.
     fn reattach_device(&mut self) -> Result<(), RenderError> {
@@ -189,6 +231,7 @@ impl WinAudioRenderer {
             self.num_channels,
             self.frame_rate,
             self.incoming_buffer_size_in_frames,
+            None,
         )?;
         Ok(())
     }
@@ -197,8 +240,7 @@ impl WinAudioRenderer {
 impl PlaybackBufferStream for WinAudioRenderer {
     /// Returns a wrapper around the WASAPI buffer.
     fn next_playback_buffer<'b, 's: 'b>(&'s mut self) -> Result<PlaybackBuffer<'b>, BoxError> {
-        const MAX_REATTACH_TRIES: usize = 50;
-        for _ in 0..MAX_REATTACH_TRIES {
+        for _ in 0..Self::MAX_REATTACH_TRIES {
             match self.device.next_win_buffer() {
                 Ok(_) => return self.device.playback_buffer().map_err(|e| Box::new(e) as _),
                 // If the audio device was disconnected, set up whatever is now the default device
@@ -220,10 +262,11 @@ impl PlaybackBufferStream for WinAudioRenderer {
 pub(crate) struct DeviceRenderer {
     audio_render_client: ComPtr<IAudioRenderClient>,
     audio_client: ComPtr<IAudioClient>,
-    win_buffer: *mut *mut u8,
+    win_buffer: *mut u8,
     pub audio_shared_format: AudioSharedFormat,
     audio_render_client_buffer_frame_count: u32,
     ready_to_read_event: Event,
+    async_ready_to_read_event: Option<Box<dyn EventAsyncWrapper>>,
 }
 
 impl DeviceRenderer {
@@ -232,6 +275,7 @@ impl DeviceRenderer {
         num_channels: usize,
         _guest_frame_rate: u32,
         incoming_buffer_size_in_frames: usize,
+        ex: Option<&dyn audio_streams::AudioStreamsExecutor>,
     ) -> Result<Self, RenderError> {
         if num_channels > 2 {
             return Err(RenderError::InvalidChannelCount(num_channels));
@@ -252,7 +296,9 @@ impl DeviceRenderer {
                 AUDCLNT_SHAREMODE_SHARED,
                 AUDCLNT_STREAMFLAGS_EVENTCALLBACK
                     | AUDCLNT_STREAMFLAGS_AUTOCONVERTPCM
-                    | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY,
+                    | AUDCLNT_STREAMFLAGS_SRC_DEFAULT_QUALITY
+                    | AUDCLNT_SESSIONFLAGS_DISPLAY_HIDEWHENEXPIRED
+                    | AUDCLNT_SESSIONFLAGS_EXPIREWHENUNOWNED,
                 0, /* hnsBufferDuration */
                 0, /* hnsPeriodicity */
                 format.as_ptr(),
@@ -271,7 +317,7 @@ impl DeviceRenderer {
         let hr = unsafe { audio_client.SetEventHandle(ready_to_read_event.as_raw_descriptor()) };
         check_hresult!(hr, RenderError::from(hr), "SetEventHandle() failed.")?;
 
-        let audio_render_client = DeviceRenderer::create_audio_render_client(&*audio_client)?;
+        let audio_render_client = DeviceRenderer::create_audio_render_client(&audio_client)?;
 
         let mut shared_default_size_in_100nanoseconds: i64 = 0;
         let mut exclusive_min: i64 = 0;
@@ -322,14 +368,29 @@ impl DeviceRenderer {
             "Audio Render Client Start() failed."
         )?;
 
+        let async_ready_to_read_event = if let Some(ex) = ex {
+            // Unsafe if `ready_to_read_event` and `async_ready_to_read_event` have different
+            // lifetimes because both can close the underlying `RawDescriptor`. However, both
+            // are stored in the `DeviceRenderer` fields, so this is safe.
+            Some(unsafe {
+                ex.async_event(ready_to_read_event.as_raw_descriptor())
+                    .map_err(|e| {
+                        RenderError::AsyncError(e, "Failed to create async event".to_string())
+                    })?
+            })
+        } else {
+            None
+        };
+
         Ok(Self {
             audio_render_client,
             audio_client,
-            win_buffer: MaybeUninit::uninit().as_mut_ptr(),
+            win_buffer: std::ptr::null_mut(),
             audio_shared_format: format
                 .create_audio_shared_format(shared_audio_engine_period_in_frames),
             audio_render_client_buffer_frame_count,
             ready_to_read_event,
+            async_ready_to_read_event,
         })
     }
 
@@ -366,7 +427,7 @@ impl DeviceRenderer {
         }
 
         info!("Audio Engine Mix Format Used: \n{:?}", format);
-        Self::check_format(&*audio_client, &format, wave_format_details, event_code)?;
+        Self::check_format(audio_client, &format, wave_format_details, event_code)?;
 
         Ok(format)
     }
@@ -506,7 +567,7 @@ impl DeviceRenderer {
 
         // Safe because `device` is guaranteed to be initialized
         let device = unsafe { ComPtr::from_raw(device) };
-        Self::print_device_info(&*device)?;
+        Self::print_device_info(&device)?;
 
         // Call Windows API functions to get the `async_op` which will be used to retrieve the
         // AudioClient. More details above function definition.
@@ -672,8 +733,6 @@ impl DeviceRenderer {
 
     // Returns a wraper around the WASAPI buffer
     fn next_win_buffer(&mut self) -> Result<(), RenderError> {
-        self.win_buffer = MaybeUninit::uninit().as_mut_ptr();
-
         // We will wait for windows to tell us when it is ready to take in the next set of
         // audio samples from the guest
         loop {
@@ -691,35 +750,53 @@ impl DeviceRenderer {
                     );
                     break;
                 }
-                let num_frames_padding: *mut u32 = MaybeUninit::uninit().as_mut_ptr();
-                let hr = self.audio_client.GetCurrentPadding(num_frames_padding);
-                check_hresult!(
-                    hr,
-                    RenderError::from(hr),
-                    "Audio Client GetCurrentPadding() failed."
-                )?;
-
-                // If the available free frames is less than the frames that are being sent over from the guest, then
-                // we want to only grab the number of frames available.
-                let num_frames_available =
-                    (self.audio_render_client_buffer_frame_count - *num_frames_padding) as usize;
-
-                if num_frames_available
-                    >= self
-                        .audio_shared_format
-                        .shared_audio_engine_period_in_frames
-                {
+                if self.enough_available_frames()? {
                     break;
                 }
             }
         }
 
+        self.get_buffer()?;
+
+        Ok(())
+    }
+
+    /// Returns true if the number of frames avaialble in the Windows playback buffer is at least
+    /// the size of one full period worth of audio samples.
+    fn enough_available_frames(&mut self) -> Result<bool, RenderError> {
+        // Safe because `num_frames_padding` is an u32 and `GetCurrentPadding` is a simple
+        // Windows function that shouldn't fail.
+        let mut num_frames_padding = 0u32;
+        let hr = unsafe { self.audio_client.GetCurrentPadding(&mut num_frames_padding) };
+        check_hresult!(
+            hr,
+            RenderError::from(hr),
+            "Audio Client GetCurrentPadding() failed."
+        )?;
+
+        // If the available free frames is less than the frames that are being sent over from the
+        // guest, then we want to only grab the number of frames available.
+        let num_frames_available =
+            (self.audio_render_client_buffer_frame_count - num_frames_padding) as usize;
+
+        Ok(num_frames_available
+            >= self
+                .audio_shared_format
+                .shared_audio_engine_period_in_frames)
+    }
+
+    fn get_buffer(&mut self) -> Result<(), RenderError> {
+        self.win_buffer = std::ptr::null_mut();
+
         // This unsafe block will get the playback buffer and return the size of the buffer
+        //
+        // This is safe because the contents of `win_buffer` will be
+        // released when `ReleaseBuffer` is called in the `BufferCommit` implementation.
         unsafe {
             let hr = self.audio_render_client.GetBuffer(
                 self.audio_shared_format
                     .shared_audio_engine_period_in_frames as u32,
-                self.win_buffer,
+                &mut self.win_buffer,
             );
             check_hresult!(
                 hr,
@@ -732,29 +809,51 @@ impl DeviceRenderer {
     }
 
     fn playback_buffer(&mut self) -> Result<PlaybackBuffer, RenderError> {
-        if self.win_buffer.is_null() {
-            return Err(RenderError::InvalidBuffer);
-        }
-
-        let frame_size_bytes = self.audio_shared_format.bit_depth as usize
-            * self.audio_shared_format.channels as usize
-            / 8;
-
         // Safe because `win_buffer` is allocated and retrieved from WASAPI. The size requested,
         // which we specified in `next_win_buffer` is exactly
         // `shared_audio_engine_period_in_frames`, so the size parameter should be valid.
-        let buffer_slice = unsafe {
-            std::slice::from_raw_parts_mut(
-                *self.win_buffer,
+        let (frame_size_bytes, buffer_slice) = unsafe {
+            Self::get_frame_size_and_buffer_slice(
+                self.audio_shared_format.bit_depth as usize,
+                self.audio_shared_format.channels as usize,
+                self.win_buffer,
                 self.audio_shared_format
-                    .shared_audio_engine_period_in_frames
-                    * frame_size_bytes,
-            )
+                    .shared_audio_engine_period_in_frames,
+            )?
         };
 
         PlaybackBuffer::new(frame_size_bytes, buffer_slice, self)
             .map_err(RenderError::PlaybackBuffer)
     }
+
+    /// # Safety
+    ///
+    /// Safe only if:
+    ///   1. `win_buffer` is pointing to a valid buffer used for holding audio bytes.
+    ///   2. `bit_depth`, `channels`, and `shared_audio_engine_period_in_frames` are accurate with
+    ///      respect to `win_buffer`, so that a valid slice can be made.
+    ///   3. The variables mentioned in reason "2." must calculate a size no greater than the size
+    ///      of the buffer pointed to by `win_buffer`.
+    unsafe fn get_frame_size_and_buffer_slice<'a>(
+        bit_depth: usize,
+        channels: usize,
+        win_buffer: *mut u8,
+        shared_audio_engine_period_in_frames: usize,
+    ) -> Result<(usize, &'a mut [u8]), RenderError> {
+        if win_buffer.is_null() {
+            return Err(RenderError::InvalidBuffer);
+        }
+
+        let frame_size_bytes = bit_depth * channels / 8;
+
+        Ok((
+            frame_size_bytes,
+            std::slice::from_raw_parts_mut(
+                win_buffer,
+                shared_audio_engine_period_in_frames * frame_size_bytes,
+            ),
+        ))
+    }
 }
 
 impl BufferCommit for DeviceRenderer {
@@ -789,7 +888,7 @@ impl Drop for DeviceRenderer {
     }
 }
 
-unsafe impl<'a> Send for DeviceRenderer {}
+unsafe impl Send for DeviceRenderer {}
 
 #[derive(Debug, ThisError)]
 pub enum RenderError {
@@ -814,6 +913,10 @@ pub enum RenderError {
     GenericError,
     #[error("Invalid guest channel count {0} is > than 2")]
     InvalidChannelCount(usize),
+    #[error("Async related error: {0}: {1}")]
+    AsyncError(std::io::Error, String),
+    #[error("Ready to read async event was not set during win_audio initialization.")]
+    MissingEventAsync,
 }
 
 impl From<i32> for RenderError {
@@ -877,7 +980,7 @@ mod tests {
     #[test]
     fn test_create_win_audio_renderer_no_co_initliazed() {
         let _shared = SERIALIZE_LOCK.lock();
-        let win_audio_renderer = DeviceRenderer::new(2, 48000, 720);
+        let win_audio_renderer = DeviceRenderer::new(2, 48000, 720, None);
         assert!(win_audio_renderer.is_err());
     }
 
@@ -886,7 +989,7 @@ mod tests {
     fn test_create_win_audio_renderer() {
         let _shared = SERIALIZE_LOCK.lock();
         let _co_init = SafeCoInit::new_coinitialize();
-        let win_audio_renderer_result = DeviceRenderer::new(2, 48000, 480);
+        let win_audio_renderer_result = DeviceRenderer::new(2, 48000, 480, None);
         assert!(win_audio_renderer_result.is_ok());
         let win_audio_renderer = win_audio_renderer_result.unwrap();
         assert_eq!(
@@ -916,7 +1019,7 @@ mod tests {
     // there is no way to copy audio samples over succiently.
     fn test_guest_buffer_size_bigger_than_audio_render_client_buffer_size() {
         let _shared = SERIALIZE_LOCK.lock();
-        let win_audio_renderer = DeviceRenderer::new(2, 48000, 100000);
+        let win_audio_renderer = DeviceRenderer::new(2, 48000, 100000, None);
 
         assert!(win_audio_renderer.is_err());
     }

win_audio/src/win_audio_impl/wave_format.rs

@@ -55,6 +55,7 @@ impl WaveAudioFormat {
     /// Unsafe if `wave_format_ptr` is pointing to null. This function will assume it's not null
     /// and dereference it.
     /// Also `format_ptr` will be deallocated after this function completes, so it cannot be used.
+    #[allow(clippy::let_and_return)]
     pub unsafe fn new(format_ptr: *mut WAVEFORMATEX) -> Self {
         let format_tag = { (*format_ptr).wFormatTag };
         let result = if format_tag != WAVE_FORMAT_EXTENSIBLE {
@@ -278,12 +279,20 @@ impl Debug for WaveAudioFormat {
 
                 let subformat = wave_format_extensible.SubFormat;
 
-                if !IsEqualGUID(
-                    &{ wave_format_extensible.SubFormat },
-                    &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
-                ) {
-                    warn!("Audio Engine format is NOT IEEE FLOAT");
-                }
+                // TODO(b/240186720): Passing in `KSDATAFORMAT_SUBTYPE_PCM` will cause a
+                // freeze. IsEqualGUID is unsafe even though it isn't marked as such. Look into
+                // fixing or possibily write our own, that works.
+                //
+                // This check would be a nice to have, but not necessary. Right now, the subformat
+                // used will always be `IEEE_FLOAT`, so this check will be useless if nothing
+                // changes.
+                //
+                // if !IsEqualGUID(
+                //     &{ wave_format_extensible.SubFormat },
+                //     &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT,
+                // ) {
+                //     warn!("Audio Engine format is NOT IEEE FLOAT");
+                // }
 
                 let audio_engine_extensible_format = format!(
                     "\nSamples: {}, \ndwChannelMask: {}, \nSubFormat: {}-{}-{}-{:?}",