blob: 2baf6f1ac9a80be7b36f94debb0f92e5fbc59e95 [file] [log] [blame]
// Copyright (c) 2012 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.
// - 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).
// 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 All APIs must be called from the creating thread as well.
// - It is required to first acquire the native audio parameters of the default
// output device and then use the same rate when creating this object. Use
// e.g. WASAPIAudioOutputStream::HardwareSampleRate() to retrieve the sample
// rate. Open() will fail unless "perfect" audio parameters are utilized.
// - Calling Close() also leads to self destruction.
// - Support for 8-bit audio has not yet been verified and tested.
// Core Audio API details:
// - The public API methods (Open(), Start(), Stop() and Close()) must be
// called on constructing thread. The reason is that we want to ensure that
// the COM environment is the same for all API implementations.
// - 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.
// Threading details:
// - It is assumed that this class is created on the audio thread owned
// by the AudioManager.
// - It is a requirement to call the following methods on the same audio
// thread: Open(), Start(), Stop(), and Close().
// - Audio rendering is performed on the audio render thread, owned by this
// class, and the AudioSourceCallback::OnMoreData() method will be called
// from this thread. Stream switching also takes place on the audio-render
// thread.
// Experimental exclusive mode:
// - It is possible to open up a stream in exclusive mode by using the
// --enable-exclusive-audio command line flag.
// - The internal buffering scheme is less flexible for exclusive streams.
// Hence, some manual tuning will be required before deciding what frame
// size to use. See the WinAudioOutputTest unit test for more details.
// - If an application opens a stream in exclusive mode, the application has
// exclusive use of the audio endpoint device that plays the stream.
// - Exclusive-mode should only be utilized when the lowest possible latency
// is important.
// - In exclusive mode, the client can choose to open the stream in any audio
// format that the endpoint device supports, i.e. not limited to the device's
// current (default) configuration.
// - Initial measurements on Windows 7 (HP Z600 workstation) have shown that
// the lowest possible latencies we can achieve on this machine are:
// o ~3.3333ms @ 48kHz <=> 160 audio frames per buffer.
// o ~3.6281ms @ 44.1kHz <=> 160 audio frames per buffer.
// - See
// for more details.
#include <Audioclient.h>
#include <MMDeviceAPI.h>
#include <string>
#include "base/compiler_specific.h"
#include "base/memory/scoped_ptr.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"
namespace media {
class AudioManagerWin;
// AudioOutputStream implementation using Windows Core Audio APIs.
class MEDIA_EXPORT WASAPIAudioOutputStream :
public AudioOutputStream,
public base::DelegateSimpleThread::Delegate {
// The ctor takes all the usual parameters, plus |manager| which is the
// the audio manager who is creating this object.
WASAPIAudioOutputStream(AudioManagerWin* manager,
const std::string& device_id,
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 sample rate the audio engine uses for its internal
// processing/mixing of shared-mode streams. To fetch the settings for the
// default device, pass an empty string as the |device_id|.
static int HardwareSampleRate(const std::string& device_id);
// Returns AUDCLNT_SHAREMODE_EXCLUSIVE if --enable-exclusive-mode is used
// as command-line flag and AUDCLNT_SHAREMODE_SHARED otherwise (default).
static AUDCLNT_SHAREMODE GetShareMode();
bool started() const { return render_thread_.get() != NULL; }
// DelegateSimpleThread::Delegate implementation.
virtual void Run() OVERRIDE;
// Core part of the thread loop which controls the actual rendering.
// Checks available amount of space in the endpoint buffer and reads
// data from the client to fill up the buffer without causing audio
// glitches.
bool RenderAudioFromSource(UINT64 device_frequency);
// Called when the device will be opened in exclusive mode and use the
// application specified format.
// TODO(henrika): rewrite and move to CoreAudioUtil when removing flag
// for exclusive audio mode.
HRESULT ExclusiveModeInitialization(IAudioClient* client,
HANDLE event_handle,
uint32* endpoint_buffer_size);
// If |render_thread_| is valid, sets |stop_render_event_| and blocks until
// the thread has stopped. |stop_render_event_| is reset after the call.
// |source_| is set to NULL.
void StopThread();
// Contains the thread ID of the creating thread.
base::PlatformThreadId creating_thread_id_;
// 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.
scoped_ptr<base::DelegateSimpleThread> render_thread_;
// Contains the desired audio format which is set up at construction.
// Extended PCM waveform format structure based on WAVEFORMATEXTENSIBLE.
// Use this for multiple channel and hi-resolution PCM data.
// Set to true when stream is successfully opened.
bool opened_;
// We check if the input audio parameters are identical (bit depth is
// excluded) to the preferred (native) audio parameters during construction.
// Open() will fail if |audio_parameters_are_valid_| is false.
bool audio_parameters_are_valid_;
// Volume level from 0 to 1.
float volume_;
// 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_;
// Length of the audio endpoint buffer.
uint32 endpoint_buffer_size_frames_;
// The target device id or an empty string for the default device.
const std::string device_id_;
// Defines the role that the system has assigned to an audio endpoint device.
ERole device_role_;
// The sharing mode for the connection.
// where AUDCLNT_SHAREMODE_SHARED is the default.
// Counts the number of audio frames written to the endpoint buffer.
UINT64 num_written_frames_;
// Pointer to the client that will deliver audio samples to be played out.
AudioSourceCallback* source_;
// 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_;
// Container for retrieving data from AudioSourceCallback::OnMoreData().
scoped_ptr<AudioBus> audio_bus_;
base::win::ScopedComPtr<IAudioClock> audio_clock_;
} // namespace media