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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 Mac OS X using the special AUHAL
// input Audio Unit present in OS 10.4 and later.
// The AUHAL input Audio Unit is for low-latency audio I/O.
//
// Overview of operation:
//
// - An object of AUAudioInputStream is created by the AudioManager
// factory: audio_man->MakeAudioInputStream().
// - Next some thread will call Open(), at that point the underlying
// AUHAL output Audio Unit is created and configured.
// - Then some thread will call Start(sink).
// Then the Audio Unit is started which creates its own thread which
// periodically will provide the sink with more data as buffers are being
// produced/recorded.
// - At some point some thread will call Stop(), which we handle by directly
// stopping the AUHAL output Audio Unit.
// - 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:
//
// - 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 AUAudioInputStream::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 actual recording instant and the time when the
// data packet is provided as a callback.
//
#ifndef MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_
#define MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_
#include <AudioUnit/AudioUnit.h>
#include <CoreAudio/CoreAudio.h>
#include "base/atomicops.h"
#include "base/cancelable_callback.h"
#include "base/memory/scoped_ptr.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "base/time/time.h"
#include "base/timer/timer.h"
#include "media/audio/agc_audio_stream.h"
#include "media/audio/audio_io.h"
#include "media/audio/audio_parameters.h"
#include "media/base/audio_block_fifo.h"
namespace media {
class AudioBus;
class AudioManagerMac;
class DataBuffer;
class AUAudioInputStream : public AgcAudioStream<AudioInputStream> {
public:
// The ctor takes all the usual parameters, plus |manager| which is the
// the audio manager who is creating this object.
AUAudioInputStream(AudioManagerMac* manager,
const AudioParameters& input_params,
AudioDeviceID audio_device_id);
// The dtor is typically called by the AudioManager only and it is usually
// triggered by calling AudioInputStream::Close().
~AUAudioInputStream() override;
// Implementation of AudioInputStream.
bool Open() override;
void Start(AudioInputCallback* callback) override;
void Stop() override;
void Close() override;
double GetMaxVolume() override;
void SetVolume(double volume) override;
double GetVolume() override;
bool IsMuted() override;
// Returns the current hardware sample rate for the default input device.
MEDIA_EXPORT static int HardwareSampleRate();
bool started() const { return started_; }
AudioUnit audio_unit() const { return audio_unit_; }
AudioBufferList* audio_buffer_list() { return &audio_buffer_list_; }
AudioDeviceID device_id() const { return input_device_id_; }
size_t requested_buffer_size() const { return number_of_frames_; }
private:
// AudioOutputUnit callback.
static OSStatus InputProc(void* user_data,
AudioUnitRenderActionFlags* flags,
const AudioTimeStamp* time_stamp,
UInt32 bus_number,
UInt32 number_of_frames,
AudioBufferList* io_data);
// Pushes recorded data to consumer of the input audio stream.
OSStatus Provide(UInt32 number_of_frames, AudioBufferList* io_data,
const AudioTimeStamp* time_stamp);
// Gets the fixed capture hardware latency and store it during initialization.
// Returns 0 if not available.
double GetHardwareLatency();
// Gets the current capture delay value.
double GetCaptureLatency(const AudioTimeStamp* input_time_stamp);
// Gets the number of channels for a stream of audio data.
int GetNumberOfChannelsFromStream();
// Issues the OnError() callback to the |sink_|.
void HandleError(OSStatus err);
// Helper function to check if the volume control is avialable on specific
// channel.
bool IsVolumeSettableOnChannel(int channel);
// Helper methods to set and get atomic |input_callback_is_active_|.
void SetInputCallbackIsActive(bool active);
bool GetInputCallbackIsActive();
// Checks if a stream was started successfully and the audio unit also starts
// to call InputProc() as it should. This method is called once when a timer
// expires 5 seconds after calling Start().
void CheckInputStartupSuccess();
// Uninitializes the audio unit if needed.
void CloseAudioUnit();
// Adds extra UMA stats when it has been detected that startup failed.
void AddHistogramsForFailedStartup();
// Verifies that Open(), Start(), Stop() and Close() are all called on the
// creating thread which is the main browser thread (CrBrowserMain) on Mac.
base::ThreadChecker thread_checker_;
// Our creator, the audio manager needs to be notified when we close.
AudioManagerMac* const manager_;
// Contains the desired number of audio frames in each callback.
const size_t number_of_frames_;
// Pointer to the object that will receive the recorded audio samples.
AudioInputCallback* sink_;
// Structure that holds the desired output format of the stream.
// Note that, this format can differ from the device(=input) format.
AudioStreamBasicDescription format_;
// The special Audio Unit called AUHAL, which allows us to pass audio data
// directly from a microphone, through the HAL, and to our application.
// The AUHAL also enables selection of non default devices.
AudioUnit audio_unit_;
// The UID refers to the current input audio device.
const AudioDeviceID input_device_id_;
// Provides a mechanism for encapsulating one or more buffers of audio data.
AudioBufferList audio_buffer_list_;
// Temporary storage for recorded data. The InputProc() renders into this
// array as soon as a frame of the desired buffer size has been recorded.
scoped_ptr<uint8_t[]> audio_data_buffer_;
// True after successful Start(), false after successful Stop().
bool started_;
// Fixed capture hardware latency in frames.
double hardware_latency_frames_;
// The number of channels in each frame of audio data, which is used
// when querying the volume of each channel.
int number_of_channels_in_frame_;
// FIFO used to accumulates recorded data.
media::AudioBlockFifo fifo_;
// Used to defer Start() to workaround http://crbug.com/160920.
base::CancelableClosure deferred_start_cb_;
// Contains time of last successful call to AudioUnitRender().
// Initialized first time in Start() and then updated for each valid
// audio buffer. Used to detect long error sequences and to take actions
// if length of error sequence is above a certain limit.
base::TimeTicks last_success_time_;
// Is set to true on the internal AUHAL IO thread in the first input callback
// after Start() has bee called.
base::subtle::Atomic32 input_callback_is_active_;
// Timer which triggers CheckInputStartupSuccess() to verify that input
// callbacks have started as intended after a successful call to Start().
// This timer lives on the main browser thread.
scoped_ptr<base::OneShotTimer> input_callback_timer_;
// Set to true if the Start() call was delayed.
// See AudioManagerMac::ShouldDeferStreamStart() for details.
bool start_was_deferred_;
// Set to true if the audio unit's IO buffer was changed when Open() was
// called.
bool buffer_size_was_changed_;
DISALLOW_COPY_AND_ASSIGN(AUAudioInputStream);
};
} // namespace media
#endif // MEDIA_AUDIO_MAC_AUDIO_LOW_LATENCY_INPUT_MAC_H_