blob: bb900e07aa62fd16e27ad97b6cc389bb3bc43a2d [file] [log] [blame]
* Copyright (c) 2012 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.
#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_device/audio_device_generic.h"
#include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include <AudioToolbox/AudioConverter.h>
#include <CoreAudio/CoreAudio.h>
#include <mach/semaphore.h>
struct PaUtilRingBuffer;
namespace rtc {
class PlatformThread;
} // namespace rtc
namespace webrtc
class EventWrapper;
const uint32_t N_REC_SAMPLES_PER_SEC = 48000;
const uint32_t N_PLAY_SAMPLES_PER_SEC = 48000;
const uint32_t N_REC_CHANNELS = 1; // default is mono recording
const uint32_t N_PLAY_CHANNELS = 2; // default is stereo playout
const uint32_t N_DEVICE_CHANNELS = 64;
const int N_BLOCKS_IO = 2;
const int N_BUFFERS_IN = 2; // Must be at least N_BLOCKS_IO.
const int N_BUFFERS_OUT = 3; // Must be at least N_BLOCKS_IO.
const uint32_t TIMER_PERIOD_MS = (2 * 10 * N_BLOCKS_IO * 1000000);
const uint32_t REC_BUF_SIZE_IN_SAMPLES =
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
class AudioDeviceMac: public AudioDeviceGeneric
AudioDeviceMac(const int32_t id);
// Retrieve the currently utilized audio layer
virtual int32_t
ActiveAudioLayer(AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
virtual int32_t Init();
virtual int32_t Terminate();
virtual bool Initialized() const;
// Device enumeration
virtual int16_t PlayoutDevices();
virtual int16_t RecordingDevices();
virtual int32_t PlayoutDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]);
virtual int32_t RecordingDeviceName(
uint16_t index,
char name[kAdmMaxDeviceNameSize],
char guid[kAdmMaxGuidSize]);
// Device selection
virtual int32_t SetPlayoutDevice(uint16_t index);
virtual int32_t SetPlayoutDevice(
AudioDeviceModule::WindowsDeviceType device);
virtual int32_t SetRecordingDevice(uint16_t index);
virtual int32_t SetRecordingDevice(
AudioDeviceModule::WindowsDeviceType device);
// Audio transport initialization
virtual int32_t PlayoutIsAvailable(bool& available);
virtual int32_t InitPlayout();
virtual bool PlayoutIsInitialized() const;
virtual int32_t RecordingIsAvailable(bool& available);
virtual int32_t InitRecording();
virtual bool RecordingIsInitialized() const;
// Audio transport control
virtual int32_t StartPlayout();
virtual int32_t StopPlayout();
virtual bool Playing() const;
virtual int32_t StartRecording();
virtual int32_t StopRecording();
virtual bool Recording() const;
// Microphone Automatic Gain Control (AGC)
virtual int32_t SetAGC(bool enable);
virtual bool AGC() const;
// Volume control based on the Windows Wave API (Windows only)
virtual int32_t SetWaveOutVolume(uint16_t volumeLeft, uint16_t volumeRight);
virtual int32_t WaveOutVolume(uint16_t& volumeLeft,
uint16_t& volumeRight) const;
// Audio mixer initialization
virtual int32_t InitSpeaker();
virtual bool SpeakerIsInitialized() const;
virtual int32_t InitMicrophone();
virtual bool MicrophoneIsInitialized() const;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool& available);
virtual int32_t SetSpeakerVolume(uint32_t volume);
virtual int32_t SpeakerVolume(uint32_t& volume) const;
virtual int32_t MaxSpeakerVolume(uint32_t& maxVolume) const;
virtual int32_t MinSpeakerVolume(uint32_t& minVolume) const;
virtual int32_t SpeakerVolumeStepSize(uint16_t& stepSize) const;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool& available);
virtual int32_t SetMicrophoneVolume(uint32_t volume);
virtual int32_t MicrophoneVolume(uint32_t& volume) const;
virtual int32_t MaxMicrophoneVolume(uint32_t& maxVolume) const;
virtual int32_t MinMicrophoneVolume(uint32_t& minVolume) const;
virtual int32_t
MicrophoneVolumeStepSize(uint16_t& stepSize) const;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool& available);
virtual int32_t SetMicrophoneMute(bool enable);
virtual int32_t MicrophoneMute(bool& enabled) const;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool& available);
virtual int32_t SetSpeakerMute(bool enable);
virtual int32_t SpeakerMute(bool& enabled) const;
// Microphone boost control
virtual int32_t MicrophoneBoostIsAvailable(bool& available);
virtual int32_t SetMicrophoneBoost(bool enable);
virtual int32_t MicrophoneBoost(bool& enabled) const;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool& available);
virtual int32_t SetStereoPlayout(bool enable);
virtual int32_t StereoPlayout(bool& enabled) const;
virtual int32_t StereoRecordingIsAvailable(bool& available);
virtual int32_t SetStereoRecording(bool enable);
virtual int32_t StereoRecording(bool& enabled) const;
// Delay information and control
virtual int32_t
SetPlayoutBuffer(const AudioDeviceModule::BufferType type,
uint16_t sizeMS);
virtual int32_t PlayoutBuffer(AudioDeviceModule::BufferType& type,
uint16_t& sizeMS) const;
virtual int32_t PlayoutDelay(uint16_t& delayMS) const;
virtual int32_t RecordingDelay(uint16_t& delayMS) const;
// CPU load
virtual int32_t CPULoad(uint16_t& load) const;
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();
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer);
virtual int32_t MicrophoneIsAvailable(bool& available);
virtual int32_t SpeakerIsAvailable(bool& available);
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 int32_t id, const char *msg,
const char *err);
int32_t GetNumberDevices(const AudioObjectPropertyScope scope,
AudioDeviceID scopedDeviceIds[],
const uint32_t deviceListLength);
int32_t GetDeviceName(const AudioObjectPropertyScope scope,
const uint16_t index, char* name);
int32_t InitDevice(uint16_t userDeviceIndex,
AudioDeviceID& deviceId, bool isInput);
// Always work with our preferred playout format inside VoE.
// Then convert the output to the OS setting using an AudioConverter.
OSStatus SetDesiredPlayoutFormat();
static OSStatus
objectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[],
void* clientData);
implObjectListenerProc(AudioObjectID objectId, UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[]);
int32_t HandleDeviceChange();
HandleStreamFormatChange(AudioObjectID objectId,
AudioObjectPropertyAddress propertyAddress);
HandleDataSourceChange(AudioObjectID objectId,
AudioObjectPropertyAddress propertyAddress);
HandleProcessorOverload(AudioObjectPropertyAddress propertyAddress);
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();
bool KeyPressed();
AudioDeviceBuffer* _ptrAudioBuffer;
CriticalSectionWrapper& _critSect;
EventWrapper& _stopEventRec;
EventWrapper& _stopEvent;
// TODO(pbos): Replace with direct members, just start/stop, no need to
// recreate the thread.
// Only valid/running between calls to StartRecording and StopRecording.
rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;
// Only valid/running between calls to StartPlayout and StopPlayout.
rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;
int32_t _id;
AudioMixerManagerMac _mixerManager;
uint16_t _inputDeviceIndex;
uint16_t _outputDeviceIndex;
AudioDeviceID _inputDeviceID;
AudioDeviceID _outputDeviceID;
AudioDeviceIOProcID _inDeviceIOProcID;
AudioDeviceIOProcID _deviceIOProcID;
bool _inputDeviceIsSpecified;
bool _outputDeviceIsSpecified;
uint8_t _recChannels;
uint8_t _playChannels;
Float32* _captureBufData;
SInt16* _renderBufData;
SInt16 _renderConvertData[PLAY_BUF_SIZE_IN_SAMPLES];
AudioDeviceModule::BufferType _playBufType;
bool _initialized;
bool _isShutDown;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
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;
AudioConverterRef _captureConverter;
AudioConverterRef _renderConverter;
AudioStreamBasicDescription _outStreamFormat;
AudioStreamBasicDescription _outDesiredFormat;
AudioStreamBasicDescription _inStreamFormat;
AudioStreamBasicDescription _inDesiredFormat;
uint32_t _captureLatencyUs;
uint32_t _renderLatencyUs;
// Atomically set variables
mutable int32_t _captureDelayUs;
mutable int32_t _renderDelayUs;
int32_t _renderDelayOffsetSamples;
uint16_t _playBufDelayFixed; // fixed playback delay
uint16_t _playWarning;
uint16_t _playError;
uint16_t _recWarning;
uint16_t _recError;
PaUtilRingBuffer* _paCaptureBuffer;
PaUtilRingBuffer* _paRenderBuffer;
semaphore_t _renderSemaphore;
semaphore_t _captureSemaphore;
int _captureBufSizeSamples;
int _renderBufSizeSamples;
// Typing detection
// 0x5c is key "9", after that comes function keys.
bool prev_key_state_[0x5d];
} // namespace webrtc