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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 notes:
// - It is recommended to first acquire the native sample rate of the default
// output device and then use the same rate when creating this object.
// Use AUAudioOutputStream::HardwareSampleRate() to retrieve the sample rate.
// - Calling Close() also leads to self destruction.
// - The latency consists of two parts:
// 1) Hardware latency, which includes Audio Unit latency, audio device
// latency;
// 2) The delay between the moment getting the callback and the scheduled time
// stamp that tells when the data is going to be played out.
#include <AudioUnit/AudioUnit.h>
#include <CoreAudio/CoreAudio.h>
#include "base/compiler_specific.h"
#include "base/synchronization/lock.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_parameters.h"
namespace media {
class AudioManagerMac;
// Implementation of AudioOuputStream for Mac OS X using the
// default output Audio Unit present in OS 10.4 and later.
// The default output Audio Unit is for low-latency audio I/O.
class AUAudioOutputStream : public AudioOutputStream {
// The ctor takes all the usual parameters, plus |manager| which is the
// the audio manager who is creating this object.
AUAudioOutputStream(AudioManagerMac* manager,
const AudioParameters& params);
// The dtor is typically called by the AudioManager only and it is usually
// triggered by calling AudioOutputStream::Close().
virtual ~AUAudioOutputStream();
// Implementation of AudioOutputStream.
virtual bool Open() OVERRIDE;
virtual void Close() OVERRIDE;
virtual void Start(AudioSourceCallback* callback) OVERRIDE;
virtual void Stop() OVERRIDE;
virtual void SetVolume(double volume) OVERRIDE;
virtual void GetVolume(double* volume) OVERRIDE;
static int HardwareSampleRate();
// DefaultOutputUnit callback.
static OSStatus InputProc(void* user_data,
AudioUnitRenderActionFlags* flags,
const AudioTimeStamp* time_stamp,
UInt32 bus_number,
UInt32 number_of_frames,
AudioBufferList* io_data);
OSStatus Render(UInt32 number_of_frames, AudioBufferList* io_data,
const AudioTimeStamp* output_time_stamp);
// Sets up the stream format for the default output Audio Unit.
bool Configure();
// Gets the fixed playout device hardware latency and stores it. Returns 0
// if not available.
double GetHardwareLatency();
// Gets the current playout latency value.
double GetPlayoutLatency(const AudioTimeStamp* output_time_stamp);
// Our creator, the audio manager needs to be notified when we close.
AudioManagerMac* manager_;
size_t number_of_frames_;
// Pointer to the object that will provide the audio samples.
AudioSourceCallback* source_;
// Protects |source_|. Necessary since Render() calls seem to be in flight
// when |output_unit_| is supposedly stopped. See
base::Lock source_lock_;
// Structure that holds the stream format details such as bitrate.
AudioStreamBasicDescription format_;
// The default output Audio Unit which talks to the audio hardware.
AudioUnit output_unit_;
// The UID refers to the current output audio device.
AudioDeviceID output_device_id_;
// Volume level from 0 to 1.
float volume_;
// Fixed playout hardware latency in frames.
double hardware_latency_frames_;
// The flag used to stop the streaming.
bool stopped_;
// Container for retrieving data from AudioSourceCallback::OnMoreData().
scoped_ptr<AudioBus> audio_bus_;
} // namespace media