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// 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 AudioInputStream for Windows using Windows Core Audio
// WASAPI for low latency capturing.
// Overview of operation:
// - An object of WASAPIAudioInputStream 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 IAudioCaptureClient.
// - Then some thread will call Start(sink).
// A thread called "wasapi_capture_thread" is started and this thread listens
// on an event signal which is set periodically by the audio engine for
// each recorded data packet. As a result, data samples will be provided
// to the registered sink.
// - At some point, a thread will call Stop(), which stops and joins the
// capture 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.
// 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
// WASAPIAudioInputStream::HardwareSampleRate() to retrieve the sample rate.
// - Calling Close() also leads to self destruction.
// Core Audio API details:
// - Utilized MMDevice interfaces:
// o IMMDeviceEnumerator
// o IMMDevice
// - Utilized WASAPI interfaces:
// o IAudioClient
// o IAudioCaptureClient
// - 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 capture thread.
// - Audio applications that use the MMDevice API and WASAPI typically use
// the ISimpleAudioVolume interface to manage stream volume levels on a
// per-session basis. It is also possible to use of the IAudioEndpointVolume
// interface to control the master volume level of an audio endpoint device.
// This implementation is using the ISimpleAudioVolume interface.
// MSDN states that "In rare cases, a specialized audio application might
// require the use of the IAudioEndpointVolume".
#include <Audioclient.h>
#include <MMDeviceAPI.h>
#include <string>
#include "base/compiler_specific.h"
#include "base/threading/non_thread_safe.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/agc_audio_stream.h"
#include "media/audio/audio_parameters.h"
#include "media/base/media_export.h"
namespace media {
class AudioManagerWin;
// AudioInputStream implementation using Windows Core Audio APIs.
class MEDIA_EXPORT WASAPIAudioInputStream
: public AgcAudioStream<AudioInputStream>,
public base::DelegateSimpleThread::Delegate,
NON_EXPORTED_BASE(public base::NonThreadSafe) {
// The ctor takes all the usual parameters, plus |manager| which is the
// the audio manager who is creating this object.
WASAPIAudioInputStream(AudioManagerWin* manager,
const AudioParameters& params,
const std::string& device_id);
// The dtor is typically called by the AudioManager only and it is usually
// triggered by calling AudioInputStream::Close().
virtual ~WASAPIAudioInputStream();
// Implementation of AudioInputStream.
virtual bool Open() OVERRIDE;
virtual void Start(AudioInputCallback* callback) OVERRIDE;
virtual void Stop() OVERRIDE;
virtual void Close() OVERRIDE;
virtual double GetMaxVolume() OVERRIDE;
virtual void SetVolume(double volume) OVERRIDE;
virtual double GetVolume() OVERRIDE;
// Retrieves the sample rate used by the audio engine for its internal
// processing/mixing of shared-mode streams given a specifed device.
static int HardwareSampleRate(const std::string& device_id);
// Retrieves the number of audio channels used by the audio engine for its
// internal processing/mixing of shared-mode streams given a specified device.
static uint32 HardwareChannelCount(const std::string& device_id);
bool started() const { return started_; }
// 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 SetCaptureDevice();
HRESULT ActivateCaptureDevice();
HRESULT GetAudioEngineStreamFormat();
bool DesiredFormatIsSupported();
HRESULT InitializeAudioEngine();
// Retrieves the stream format that the audio engine uses for its internal
// processing/mixing of shared-mode streams.
static HRESULT GetMixFormat(const std::string& device_id,
WAVEFORMATEX** device_format);
// Our creator, the audio manager needs to be notified when we close.
AudioManagerWin* manager_;
// Capturing is driven by this thread (which has no message loop).
// All OnData() callbacks will be called from this thread.
base::DelegateSimpleThread* capture_thread_;
// Contains the desired audio format which is set up at construction.
bool opened_;
bool started_;
// 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 user received in each
// OnData() 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_;
// Contains the unique name of the selected endpoint device.
// Note that AudioManagerBase::kDefaultDeviceId represents the default
// device role and is not a valid ID as such.
std::string device_id_;
// Conversion factor used in delay-estimation calculations.
// Converts a raw performance counter value to 100-nanosecond unit.
double perf_count_to_100ns_units_;
// Conversion factor used in delay-estimation calculations.
// Converts from milliseconds to audio frames.
double ms_to_frame_count_;
// Pointer to the object that will receive the recorded audio samples.
AudioInputCallback* sink_;
// Windows Multimedia Device (MMDevice) API interfaces.
// An IMMDevice interface which represents an audio endpoint device.
base::win::ScopedComPtr<IMMDevice> endpoint_device_;
// Windows Audio Session API (WASAP) interfaces.
// 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_;
// Loopback IAudioClient doesn't support event-driven mode, so a separate
// IAudioClient is needed to receive notifications when data is available in
// the buffer. For loopback input |audio_client_| is used to receive data,
// while |audio_render_client_for_loopback_| is used to get notifications
// when a new buffer is ready. See comment in InitializeAudioEngine() for
// details.
base::win::ScopedComPtr<IAudioClient> audio_render_client_for_loopback_;
// The IAudioCaptureClient interface enables a client to read input data
// from a capture endpoint buffer.
base::win::ScopedComPtr<IAudioCaptureClient> audio_capture_client_;
// The ISimpleAudioVolume interface enables a client to control the
// master volume level of an audio session.
// The volume-level is a value in the range 0.0 to 1.0.
// This interface does only work with shared-mode streams.
base::win::ScopedComPtr<ISimpleAudioVolume> simple_audio_volume_;
// The audio engine will signal this event each time a buffer has been
// recorded.
base::win::ScopedHandle audio_samples_ready_event_;
// This event will be signaled when capturing shall stop.
base::win::ScopedHandle stop_capture_event_;
} // namespace media