Low-latency AudioOutputStream implementation based on WASAPI for Windows.

media/audio/win/audio_low_latency_output_win.h

+// Copyright (c) 2011 The Chromium Authors. All rights reserved.
+// Use of this source code is governed by a BSD-style license that can be
+// found in the LICENSE file.
+//
+// Implementation of AudioOutputStream for Windows using Windows Core Audio
+// WASAPI for low latency rendering.
+//
+// Overview of operation and performance:
+//
+// - An object of WASAPIAudioOutputStream is created by the AudioManager
+//   factory.
+// - Next some thread will call Open(), at that point the underlying
+//   Core Audio APIs are utilized to create two WASAPI interfaces called
+//   IAudioClient and IAudioRenderClient.
+// - Then some thread will call Start(source).
+//   A thread called "wasapi_render_thread" is started and this thread listens
+//   on an event signal which is set periodically by the audio engine to signal
+//   render events. As a result, OnMoreData() will be called and the registered
+//   client is then expected to provide data samples to be played out.
+// - At some point, a thread will call Stop(), which stops and joins the
+//   render thread and at the same time stops audio streaming.
+// - The same thread that called stop will call Close() where we cleanup
+//   and notify the audio manager, which likely will destroy this object.
+// - Initial tests on Windows 7 shows that this implementation results in a
+//   latency of approximately 35 ms if the selected packet size is less than
+//   or equal to 20 ms. Using a packet size of 10 ms does not result in a
+//   lower latency but only affects the size of the data buffer in each
+//   OnMoreData() callback.
+// - A total typical delay of 35 ms contains three parts:
+//    o Audio endpoint device period (~10 ms).
+//    o Stream latency between the buffer and endpoint device (~5 ms).
+//    o Endpoint buffer (~20 ms to ensure glitch-free rendering).
+// - Note that, if the user selects a packet size of e.g. 100 ms, the total
+//   delay will be approximately 115 ms (10 + 5 + 100).
+//
+// Implementation notes:
+//
+// - The minimum supported client is Windows Vista.
+// - This implementation is single-threaded, hence:
+//    o Construction and destruction must take place from the same thread.
+//    o It is recommended to call all APIs from the same thread as well.
+// - It is recommended to first acquire the native sample rate of the default
+//   input device and then use the same rate when creating this object. Use
+//   WASAPIAudioOutputStream::HardwareSampleRate() to retrieve the sample rate.
+// - Calling Close() also leads to self destruction.
+//
+// Core Audio API details:
+//
+// - CoInitializeEx() is called on the creating thread and on the internal
+//   capture thread. Each thread's concurrency model and apartment is set
+//   to multi-threaded (MTA). CHECK() is called to ensure that we crash if
+//   CoInitializeEx(MTA) fails.
+// - Utilized MMDevice interfaces:
+//     o IMMDeviceEnumerator
+//     o IMMDevice
+// - Utilized WASAPI interfaces:
+//     o IAudioClient
+//     o IAudioRenderClient
+// - The stream is initialized in shared mode and the processing of the
+//   audio buffer is event driven.
+// - The Multimedia Class Scheduler service (MMCSS) is utilized to boost
+//   the priority of the render thread.
+// - Audio-rendering endpoint devices can have three roles:
+//   Console (eConsole), Communications (eCommunications), and Multimedia
+//   (eMultimedia). Search for "Device Roles" on MSDN for more details.
+//
+#ifndef MEDIA_AUDIO_WIN_AUDIO_LOW_LATENCY_OUTPUT_WIN_H_
+#define MEDIA_AUDIO_WIN_AUDIO_LOW_LATENCY_OUTPUT_WIN_H_
+
+#include <Audioclient.h>
+#include <MMDeviceAPI.h>
+
+#include "base/compiler_specific.h"
+#include "base/threading/platform_thread.h"
+#include "base/threading/simple_thread.h"
+#include "base/win/scoped_co_mem.h"
+#include "base/win/scoped_com_initializer.h"
+#include "base/win/scoped_comptr.h"
+#include "base/win/scoped_handle.h"
+#include "media/audio/audio_io.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/media_export.h"
+
+class AudioManagerWin;
+
+// AudioOutputStream implementation using Windows Core Audio APIs.
+class MEDIA_EXPORT WASAPIAudioOutputStream
+    : public AudioOutputStream,
+      public base::DelegateSimpleThread::Delegate {
+ public:
+  // The ctor takes all the usual parameters, plus |manager| which is the
+  // the audio manager who is creating this object.
+  WASAPIAudioOutputStream(AudioManagerWin* manager,
+                          const AudioParameters& params,
+                          ERole device_role);
+  // The dtor is typically called by the AudioManager only and it is usually
+  // triggered by calling AudioOutputStream::Close().
+  virtual ~WASAPIAudioOutputStream();
+
+  // Implementation of AudioOutputStream.
+  virtual bool Open() OVERRIDE;
+  virtual void Start(AudioSourceCallback* callback) OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual void Close() OVERRIDE;
+  virtual void SetVolume(double volume) OVERRIDE;
+  virtual void GetVolume(double* volume) OVERRIDE;
+
+  // Retrieves the stream format that the audio engine uses for its internal
+  // processing/mixing of shared-mode streams.
+  static double HardwareSampleRate(ERole device_role);
+
+  bool started() const { return started_; }
+
+ private:
+  // DelegateSimpleThread::Delegate implementation.
+  virtual void Run() OVERRIDE;
+
+  // Issues the OnError() callback to the |sink_|.
+  void HandleError(HRESULT err);
+
+  // The Open() method is divided into these sub methods.
+  HRESULT SetRenderDevice(ERole device_role);
+  HRESULT ActivateRenderDevice();
+  HRESULT GetAudioEngineStreamFormat();
+  bool DesiredFormatIsSupported();
+  HRESULT InitializeAudioEngine();
+
+  // Initializes the COM library for use by the calling thread and sets the
+  // thread's concurrency model to multi-threaded.
+  base::win::ScopedCOMInitializer com_init_;
+
+  // Our creator, the audio manager needs to be notified when we close.
+  AudioManagerWin* manager_;
+
+  // Rendering is driven by this thread (which has no message loop).
+  // All OnMoreData() callbacks will be called from this thread.
+  base::DelegateSimpleThread* render_thread_;
+
+  // Contains the desired audio format which is set up at construction.
+  WAVEFORMATEX format_;
+
+  // Copy of the audio format which we know the audio engine supports.
+  // It is recommended to ensure that the sample rate in |format_| is identical
+  // to the sample rate in |audio_engine_mix_format_|.
+  base::win::ScopedCoMem<WAVEFORMATEX> audio_engine_mix_format_;
+
+  bool opened_;
+  bool started_;
+
+  // Volume level from 0 to 1.
+  float volume_;
+
+  // Size in bytes of each audio frame (4 bytes for 16-bit stereo PCM).
+  size_t frame_size_;
+
+  // Size in audio frames of each audio packet where an audio packet
+  // is defined as the block of data which the source is expected to deliver
+  // in each OnMoreData() callback.
+  size_t packet_size_frames_;
+
+  // Size in bytes of each audio packet.
+  size_t packet_size_bytes_;
+
+  // Size in milliseconds of each audio packet.
+  float packet_size_ms_;
+
+  // Length of the audio endpoint buffer.
+  size_t endpoint_buffer_size_frames_;
+
+  // Defines the role that the system has assigned to an audio endpoint device.
+  ERole device_role_;
+
+  // Counts the number of audio frames written to the endpoint buffer.
+  UINT64 num_written_frames_;

[Raymond Toy (Google), 2011/11/03 22:49:56]

No problem here, but I was just wondering if you really need to count up to
64bits of frames written.

[henrika (OOO until Aug 14), 2011/11/04 11:26:15]

Simply want to avoid wrapping even for very long audio sessions.

+
+  // Pointer to the client that will deliver audio samples to be played out.
+  AudioSourceCallback* source_;
+
+  // An IMMDevice interface which represents an audio endpoint device.
+  base::win::ScopedComPtr<IMMDevice> endpoint_device_;
+
+  // An IAudioClient interface which enables a client to create and initialize
+  // an audio stream between an audio application and the audio engine.
+  base::win::ScopedComPtr<IAudioClient> audio_client_;
+
+  // The IAudioRenderClient interface enables a client to write output
+  // data to a rendering endpoint buffer.
+  base::win::ScopedComPtr<IAudioRenderClient> audio_render_client_;
+
+  // The audio engine will signal this event each time a buffer becomes
+  // ready to be filled by the client.
+  base::win::ScopedHandle audio_samples_render_event_;
+
+  // This event will be signaled when rendering shall stop.
+  base::win::ScopedHandle stop_render_event_;
+
+  DISALLOW_COPY_AND_ASSIGN(WASAPIAudioOutputStream);
+};
+
+#endif  // MEDIA_AUDIO_WIN_AUDIO_LOW_LATENCY_OUTPUT_WIN_H_