401 lines
14 KiB
C
401 lines
14 KiB
C
|
/*
|
||
|
|
* Copyright (c) 2011 The WebRTC project authors. All Rights Reserved.
|
||
|
|
*
|
||
|
|
* Use of this source code is governed by a BSD-style license
|
||
|
|
* that can be found in the LICENSE file in the root of the source
|
||
|
|
* tree. An additional intellectual property rights grant can be found
|
||
|
|
* in the file PATENTS. All contributing project authors may
|
||
|
|
* be found in the AUTHORS file in the root of the source tree.
|
||
|
|
*/
|
||
|
|
|
||
|
|
#ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
|
||
|
|
#define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
|
||
|
|
|
||
|
|
#include "audio_device_generic.h"
|
||
|
|
#include "critical_section_wrapper.h"
|
||
|
|
#include "audio_mixer_manager_mac.h"
|
||
|
|
|
||
|
|
#include <CoreAudio/CoreAudio.h>
|
||
|
|
#include <AudioToolbox/AudioConverter.h>
|
||
|
|
#include <mach/semaphore.h>
|
||
|
|
|
||
|
|
struct PaUtilRingBuffer;
|
||
|
|
|
||
|
|
namespace webrtc
|
||
|
|
{
|
||
|
|
class EventWrapper;
|
||
|
|
class ThreadWrapper;
|
||
|
|
|
||
|
|
const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 48000;
|
||
|
|
const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 48000;
|
||
|
|
|
||
|
|
const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording
|
||
|
|
const WebRtc_UWord32 N_PLAY_CHANNELS = 2; // default is stereo playout
|
||
|
|
const WebRtc_UWord32 N_DEVICE_CHANNELS = 8;
|
||
|
|
|
||
|
|
const WebRtc_UWord32 ENGINE_REC_BUF_SIZE_IN_SAMPLES = (N_REC_SAMPLES_PER_SEC
|
||
|
|
/ 100);
|
||
|
|
const WebRtc_UWord32 ENGINE_PLAY_BUF_SIZE_IN_SAMPLES = (N_PLAY_SAMPLES_PER_SEC
|
||
|
|
/ 100);
|
||
|
|
|
||
|
|
enum
|
||
|
|
{
|
||
|
|
N_BLOCKS_IO = 2
|
||
|
|
};
|
||
|
|
enum
|
||
|
|
{
|
||
|
|
N_BUFFERS_IN = 10
|
||
|
|
};
|
||
|
|
enum
|
||
|
|
{
|
||
|
|
N_BUFFERS_OUT = 3
|
||
|
|
}; // Must be at least N_BLOCKS_IO
|
||
|
|
|
||
|
|
const WebRtc_UWord32 TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
|
||
|
|
|
||
|
|
const WebRtc_UWord32 REC_BUF_SIZE_IN_SAMPLES = (ENGINE_REC_BUF_SIZE_IN_SAMPLES
|
||
|
|
* N_DEVICE_CHANNELS * N_BUFFERS_IN);
|
||
|
|
const WebRtc_UWord32 PLAY_BUF_SIZE_IN_SAMPLES =
|
||
|
|
(ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT);
|
||
|
|
|
||
|
|
class AudioDeviceMac: public AudioDeviceGeneric
|
||
|
|
{
|
||
|
|
public:
|
||
|
|
AudioDeviceMac(const WebRtc_Word32 id);
|
||
|
|
~AudioDeviceMac();
|
||
|
|
|
||
|
|
// Retrieve the currently utilized audio layer
|
||
|
|
virtual WebRtc_Word32
|
||
|
|
ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
|
||
|
|
|
||
|
|
// Main initializaton and termination
|
||
|
|
virtual WebRtc_Word32 Init();
|
||
|
|
virtual WebRtc_Word32 Terminate();
|
||
|
|
virtual bool Initialized() const;
|
||
|
|
|
||
|
|
// Device enumeration
|
||
|
|
virtual WebRtc_Word16 PlayoutDevices();
|
||
|
|
virtual WebRtc_Word16 RecordingDevices();
|
||
|
|
virtual WebRtc_Word32 PlayoutDeviceName(
|
||
|
|
WebRtc_UWord16 index,
|
||
|
|
WebRtc_Word8 name[kAdmMaxDeviceNameSize],
|
||
|
|
WebRtc_Word8 guid[kAdmMaxGuidSize]);
|
||
|
|
virtual WebRtc_Word32 RecordingDeviceName(
|
||
|
|
WebRtc_UWord16 index,
|
||
|
|
WebRtc_Word8 name[kAdmMaxDeviceNameSize],
|
||
|
|
WebRtc_Word8 guid[kAdmMaxGuidSize]);
|
||
|
|
|
||
|
|
// Device selection
|
||
|
|
virtual WebRtc_Word32 SetPlayoutDevice(WebRtc_UWord16 index);
|
||
|
|
virtual WebRtc_Word32 SetPlayoutDevice(
|
||
|
|
AudioDeviceModule::WindowsDeviceType device);
|
||
|
|
virtual WebRtc_Word32 SetRecordingDevice(WebRtc_UWord16 index);
|
||
|
|
virtual WebRtc_Word32 SetRecordingDevice(
|
||
|
|
AudioDeviceModule::WindowsDeviceType device);
|
||
|
|
|
||
|
|
// Audio transport initialization
|
||
|
|
virtual WebRtc_Word32 PlayoutIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 InitPlayout();
|
||
|
|
virtual bool PlayoutIsInitialized() const;
|
||
|
|
virtual WebRtc_Word32 RecordingIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 InitRecording();
|
||
|
|
virtual bool RecordingIsInitialized() const;
|
||
|
|
|
||
|
|
// Audio transport control
|
||
|
|
virtual WebRtc_Word32 StartPlayout();
|
||
|
|
virtual WebRtc_Word32 StopPlayout();
|
||
|
|
virtual bool Playing() const;
|
||
|
|
virtual WebRtc_Word32 StartRecording();
|
||
|
|
virtual WebRtc_Word32 StopRecording();
|
||
|
|
virtual bool Recording() const;
|
||
|
|
|
||
|
|
// Microphone Automatic Gain Control (AGC)
|
||
|
|
virtual WebRtc_Word32 SetAGC(bool enable);
|
||
|
|
virtual bool AGC() const;
|
||
|
|
|
||
|
|
// Volume control based on the Windows Wave API (Windows only)
|
||
|
|
virtual WebRtc_Word32 SetWaveOutVolume(WebRtc_UWord16 volumeLeft,
|
||
|
|
WebRtc_UWord16 volumeRight);
|
||
|
|
virtual WebRtc_Word32 WaveOutVolume(WebRtc_UWord16& volumeLeft,
|
||
|
|
WebRtc_UWord16& volumeRight) const;
|
||
|
|
|
||
|
|
// Audio mixer initialization
|
||
|
|
virtual WebRtc_Word32 SpeakerIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 InitSpeaker();
|
||
|
|
virtual bool SpeakerIsInitialized() const;
|
||
|
|
virtual WebRtc_Word32 MicrophoneIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 InitMicrophone();
|
||
|
|
virtual bool MicrophoneIsInitialized() const;
|
||
|
|
|
||
|
|
// Speaker volume controls
|
||
|
|
virtual WebRtc_Word32 SpeakerVolumeIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetSpeakerVolume(WebRtc_UWord32 volume);
|
||
|
|
virtual WebRtc_Word32 SpeakerVolume(WebRtc_UWord32& volume) const;
|
||
|
|
virtual WebRtc_Word32 MaxSpeakerVolume(WebRtc_UWord32& maxVolume) const;
|
||
|
|
virtual WebRtc_Word32 MinSpeakerVolume(WebRtc_UWord32& minVolume) const;
|
||
|
|
virtual WebRtc_Word32 SpeakerVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||
|
|
|
||
|
|
// Microphone volume controls
|
||
|
|
virtual WebRtc_Word32 MicrophoneVolumeIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetMicrophoneVolume(WebRtc_UWord32 volume);
|
||
|
|
virtual WebRtc_Word32 MicrophoneVolume(WebRtc_UWord32& volume) const;
|
||
|
|
virtual WebRtc_Word32 MaxMicrophoneVolume(WebRtc_UWord32& maxVolume) const;
|
||
|
|
virtual WebRtc_Word32 MinMicrophoneVolume(WebRtc_UWord32& minVolume) const;
|
||
|
|
virtual WebRtc_Word32
|
||
|
|
MicrophoneVolumeStepSize(WebRtc_UWord16& stepSize) const;
|
||
|
|
|
||
|
|
// Microphone mute control
|
||
|
|
virtual WebRtc_Word32 MicrophoneMuteIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetMicrophoneMute(bool enable);
|
||
|
|
virtual WebRtc_Word32 MicrophoneMute(bool& enabled) const;
|
||
|
|
|
||
|
|
// Speaker mute control
|
||
|
|
virtual WebRtc_Word32 SpeakerMuteIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetSpeakerMute(bool enable);
|
||
|
|
virtual WebRtc_Word32 SpeakerMute(bool& enabled) const;
|
||
|
|
|
||
|
|
// Microphone boost control
|
||
|
|
virtual WebRtc_Word32 MicrophoneBoostIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetMicrophoneBoost(bool enable);
|
||
|
|
virtual WebRtc_Word32 MicrophoneBoost(bool& enabled) const;
|
||
|
|
|
||
|
|
// Stereo support
|
||
|
|
virtual WebRtc_Word32 StereoPlayoutIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetStereoPlayout(bool enable);
|
||
|
|
virtual WebRtc_Word32 StereoPlayout(bool& enabled) const;
|
||
|
|
virtual WebRtc_Word32 StereoRecordingIsAvailable(bool& available);
|
||
|
|
virtual WebRtc_Word32 SetStereoRecording(bool enable);
|
||
|
|
virtual WebRtc_Word32 StereoRecording(bool& enabled) const;
|
||
|
|
|
||
|
|
// Delay information and control
|
||
|
|
virtual WebRtc_Word32
|
||
|
|
SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
|
||
|
|
WebRtc_UWord16 sizeMS);
|
||
|
|
virtual WebRtc_Word32 PlayoutBuffer(AudioDeviceModule::BufferType& type,
|
||
|
|
WebRtc_UWord16& sizeMS) const;
|
||
|
|
virtual WebRtc_Word32 PlayoutDelay(WebRtc_UWord16& delayMS) const;
|
||
|
|
virtual WebRtc_Word32 RecordingDelay(WebRtc_UWord16& delayMS) const;
|
||
|
|
|
||
|
|
// CPU load
|
||
|
|
virtual WebRtc_Word32 CPULoad(WebRtc_UWord16& load) const;
|
||
|
|
|
||
|
|
public:
|
||
|
|
virtual bool PlayoutWarning() const;
|
||
|
|
virtual bool PlayoutError() const;
|
||
|
|
virtual bool RecordingWarning() const;
|
||
|
|
virtual bool RecordingError() const;
|
||
|
|
virtual void ClearPlayoutWarning();
|
||
|
|
virtual void ClearPlayoutError();
|
||
|
|
virtual void ClearRecordingWarning();
|
||
|
|
virtual void ClearRecordingError();
|
||
|
|
|
||
|
|
public:
|
||
|
|
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
|
||
|
|
|
||
|
|
private:
|
||
|
|
void Lock()
|
||
|
|
{
|
||
|
|
_critSect.Enter();
|
||
|
|
}
|
||
|
|
;
|
||
|
|
void UnLock()
|
||
|
|
{
|
||
|
|
_critSect.Leave();
|
||
|
|
}
|
||
|
|
;
|
||
|
|
WebRtc_Word32 Id()
|
||
|
|
{
|
||
|
|
return _id;
|
||
|
|
}
|
||
|
|
|
||
|
|
static void AtomicSet32(int32_t* theValue, int32_t newValue);
|
||
|
|
static int32_t AtomicGet32(int32_t* theValue);
|
||
|
|
|
||
|
|
static void logCAMsg(const TraceLevel level,
|
||
|
|
const TraceModule module,
|
||
|
|
const WebRtc_Word32 id, const char *msg,
|
||
|
|
const char *err);
|
||
|
|
|
||
|
|
WebRtc_Word32 GetNumberDevices(const AudioObjectPropertyScope scope,
|
||
|
|
AudioDeviceID scopedDeviceIds[],
|
||
|
|
const WebRtc_UWord32 deviceListLength);
|
||
|
|
|
||
|
|
WebRtc_Word32 GetDeviceName(const AudioObjectPropertyScope scope,
|
||
|
|
const WebRtc_UWord16 index, char* name);
|
||
|
|
|
||
|
|
WebRtc_Word32 InitDevice(WebRtc_UWord16 userDeviceIndex,
|
||
|
|
AudioDeviceID& deviceId, bool isInput);
|
||
|
|
|
||
|
|
static OSStatus
|
||
|
|
objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
|
||
|
|
const AudioObjectPropertyAddress addresses[],
|
||
|
|
void* clientData);
|
||
|
|
|
||
|
|
OSStatus
|
||
|
|
implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
|
||
|
|
const AudioObjectPropertyAddress addresses[]);
|
||
|
|
|
||
|
|
WebRtc_Word32 HandleDeviceChange();
|
||
|
|
|
||
|
|
WebRtc_Word32
|
||
|
|
HandleStreamFormatChange(AudioObjectID objectId,
|
||
|
|
AudioObjectPropertyAddress propertyAddress);
|
||
|
|
|
||
|
|
WebRtc_Word32
|
||
|
|
HandleDataSourceChange(AudioObjectID objectId,
|
||
|
|
AudioObjectPropertyAddress propertyAddress);
|
||
|
|
|
||
|
|
WebRtc_Word32
|
||
|
|
HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
|
||
|
|
|
||
|
|
private:
|
||
|
|
static OSStatus deviceIOProc(AudioDeviceID device,
|
||
|
|
const AudioTimeStamp *now,
|
||
|
|
const AudioBufferList *inputData,
|
||
|
|
const AudioTimeStamp *inputTime,
|
||
|
|
AudioBufferList *outputData,
|
||
|
|
const AudioTimeStamp* outputTime,
|
||
|
|
void *clientData);
|
||
|
|
|
||
|
|
static OSStatus
|
||
|
|
outConverterProc(AudioConverterRef audioConverter,
|
||
|
|
UInt32 *numberDataPackets, AudioBufferList *data,
|
||
|
|
AudioStreamPacketDescription **dataPacketDescription,
|
||
|
|
void *userData);
|
||
|
|
|
||
|
|
static OSStatus inDeviceIOProc(AudioDeviceID device,
|
||
|
|
const AudioTimeStamp *now,
|
||
|
|
const AudioBufferList *inputData,
|
||
|
|
const AudioTimeStamp *inputTime,
|
||
|
|
AudioBufferList *outputData,
|
||
|
|
const AudioTimeStamp *outputTime,
|
||
|
|
void *clientData);
|
||
|
|
|
||
|
|
static OSStatus
|
||
|
|
inConverterProc(AudioConverterRef audioConverter,
|
||
|
|
UInt32 *numberDataPackets, AudioBufferList *data,
|
||
|
|
AudioStreamPacketDescription **dataPacketDescription,
|
||
|
|
void *inUserData);
|
||
|
|
|
||
|
|
OSStatus implDeviceIOProc(const AudioBufferList *inputData,
|
||
|
|
const AudioTimeStamp *inputTime,
|
||
|
|
AudioBufferList *outputData,
|
||
|
|
const AudioTimeStamp *outputTime);
|
||
|
|
|
||
|
|
OSStatus implOutConverterProc(UInt32 *numberDataPackets,
|
||
|
|
AudioBufferList *data);
|
||
|
|
|
||
|
|
OSStatus implInDeviceIOProc(const AudioBufferList *inputData,
|
||
|
|
const AudioTimeStamp *inputTime);
|
||
|
|
|
||
|
|
OSStatus implInConverterProc(UInt32 *numberDataPackets,
|
||
|
|
AudioBufferList *data);
|
||
|
|
|
||
|
|
static bool RunCapture(void*);
|
||
|
|
static bool RunRender(void*);
|
||
|
|
bool CaptureWorkerThread();
|
||
|
|
bool RenderWorkerThread();
|
||
|
|
|
||
|
|
private:
|
||
|
|
AudioDeviceBuffer* _ptrAudioBuffer;
|
||
|
|
|
||
|
|
CriticalSectionWrapper& _critSect;
|
||
|
|
CriticalSectionWrapper& _critSectCb;
|
||
|
|
|
||
|
|
EventWrapper& _stopEventRec;
|
||
|
|
EventWrapper& _stopEvent;
|
||
|
|
|
||
|
|
ThreadWrapper* _captureWorkerThread;
|
||
|
|
ThreadWrapper* _renderWorkerThread;
|
||
|
|
WebRtc_UWord32 _captureWorkerThreadId;
|
||
|
|
WebRtc_UWord32 _renderWorkerThreadId;
|
||
|
|
|
||
|
|
WebRtc_Word32 _id;
|
||
|
|
|
||
|
|
AudioMixerManagerMac _mixerManager;
|
||
|
|
|
||
|
|
WebRtc_UWord16 _inputDeviceIndex;
|
||
|
|
WebRtc_UWord16 _outputDeviceIndex;
|
||
|
|
AudioDeviceID _inputDeviceID;
|
||
|
|
AudioDeviceID _outputDeviceID;
|
||
|
|
#if __MAC_OS_X_VERSION_MAX_ALLOWED >= 1050
|
||
|
|
AudioDeviceIOProcID _inDeviceIOProcID;
|
||
|
|
AudioDeviceIOProcID _deviceIOProcID;
|
||
|
|
#endif
|
||
|
|
bool _inputDeviceIsSpecified;
|
||
|
|
bool _outputDeviceIsSpecified;
|
||
|
|
|
||
|
|
WebRtc_UWord8 _recChannels;
|
||
|
|
WebRtc_UWord8 _playChannels;
|
||
|
|
|
||
|
|
Float32* _captureBufData;
|
||
|
|
SInt16* _renderBufData;
|
||
|
|
|
||
|
|
SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
|
||
|
|
|
||
|
|
AudioDeviceModule::BufferType _playBufType;
|
||
|
|
|
||
|
|
private:
|
||
|
|
bool _initialized;
|
||
|
|
bool _isShutDown;
|
||
|
|
bool _recording;
|
||
|
|
bool _playing;
|
||
|
|
bool _recIsInitialized;
|
||
|
|
bool _playIsInitialized;
|
||
|
|
bool _startRec;
|
||
|
|
bool _stopRec;
|
||
|
|
bool _stopPlay;
|
||
|
|
bool _AGC;
|
||
|
|
|
||
|
|
// Atomically set varaibles
|
||
|
|
int32_t _renderDeviceIsAlive;
|
||
|
|
int32_t _captureDeviceIsAlive;
|
||
|
|
|
||
|
|
bool _twoDevices;
|
||
|
|
bool _doStop; // For play if not shared device or play+rec if shared device
|
||
|
|
bool _doStopRec; // For rec if not shared device
|
||
|
|
bool _macBookPro;
|
||
|
|
bool _macBookProPanRight;
|
||
|
|
bool _stereoRender;
|
||
|
|
bool _stereoRenderRequested;
|
||
|
|
|
||
|
|
AudioConverterRef _captureConverter;
|
||
|
|
AudioConverterRef _renderConverter;
|
||
|
|
|
||
|
|
AudioStreamBasicDescription _outStreamFormat;
|
||
|
|
AudioStreamBasicDescription _outDesiredFormat;
|
||
|
|
AudioStreamBasicDescription _inStreamFormat;
|
||
|
|
AudioStreamBasicDescription _inDesiredFormat;
|
||
|
|
|
||
|
|
WebRtc_UWord32 _captureLatencyUs;
|
||
|
|
WebRtc_UWord32 _renderLatencyUs;
|
||
|
|
|
||
|
|
// Atomically set variables
|
||
|
|
mutable int32_t _captureDelayUs;
|
||
|
|
mutable int32_t _renderDelayUs;
|
||
|
|
|
||
|
|
WebRtc_Word32 _renderDelayOffsetSamples;
|
||
|
|
|
||
|
|
private:
|
||
|
|
WebRtc_UWord16 _playBufDelay; // playback delay
|
||
|
|
WebRtc_UWord16 _playBufDelayFixed; // fixed playback delay
|
||
|
|
|
||
|
|
WebRtc_UWord16 _playWarning;
|
||
|
|
WebRtc_UWord16 _playError;
|
||
|
|
WebRtc_UWord16 _recWarning;
|
||
|
|
WebRtc_UWord16 _recError;
|
||
|
|
|
||
|
|
PaUtilRingBuffer* _paCaptureBuffer;
|
||
|
|
PaUtilRingBuffer* _paRenderBuffer;
|
||
|
|
|
||
|
|
semaphore_t _renderSemaphore;
|
||
|
|
semaphore_t _captureSemaphore;
|
||
|
|
|
||
|
|
WebRtc_UWord32 _captureBufSizeSamples;
|
||
|
|
WebRtc_UWord32 _renderBufSizeSamples;
|
||
|
|
};
|
||
|
|
|
||
|
|
} // namespace webrtc
|
||
|
|
|
||
|
|
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
|