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/*
* 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.
*/
#ifndef AUDIO_DEVICE_AUDIO_DEVICE_MAC_H_
#define AUDIO_DEVICE_AUDIO_DEVICE_MAC_H_
#include <AudioToolbox/AudioConverter.h>
#include <CoreAudio/CoreAudio.h>
#include <mach/semaphore.h>
#include <atomic>
#include <memory>
#include "absl/strings/string_view.h"
#include "modules/audio_device/audio_device_generic.h"
#include "modules/audio_device/mac/audio_mixer_manager_mac.h"
#include "rtc_base/event.h"
#include "rtc_base/logging.h"
#include "rtc_base/platform_thread.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/thread_annotations.h"
struct PaUtilRingBuffer;
namespace webrtc {
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 kBufferSizeMs = 10;
const uint32_t ENGINE_REC_BUF_SIZE_IN_SAMPLES =
N_REC_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
const uint32_t ENGINE_PLAY_BUF_SIZE_IN_SAMPLES =
N_PLAY_SAMPLES_PER_SEC * kBufferSizeMs / 1000;
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 =
ENGINE_REC_BUF_SIZE_IN_SAMPLES * N_DEVICE_CHANNELS * N_BUFFERS_IN;
const uint32_t PLAY_BUF_SIZE_IN_SAMPLES =
ENGINE_PLAY_BUF_SIZE_IN_SAMPLES * N_PLAY_CHANNELS * N_BUFFERS_OUT;
const int kGetMicVolumeIntervalMs = 1000;
class AudioDeviceMac : public AudioDeviceGeneric {
public:
AudioDeviceMac();
~AudioDeviceMac();
// Retrieve the currently utilized audio layer
virtual int32_t ActiveAudioLayer(
AudioDeviceModule::AudioLayer& audioLayer) const;
// Main initializaton and termination
virtual InitStatus Init() RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t Terminate() RTC_LOCKS_EXCLUDED(mutex_);
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) RTC_LOCKS_EXCLUDED(mutex_);
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() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool PlayoutIsInitialized() const;
virtual int32_t RecordingIsAvailable(bool& available);
virtual int32_t InitRecording() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool RecordingIsInitialized() const;
// Audio transport control
virtual int32_t StartPlayout() RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t StopPlayout() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool Playing() const;
virtual int32_t StartRecording() RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t StopRecording() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool Recording() const;
// Audio mixer initialization
virtual int32_t InitSpeaker() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool SpeakerIsInitialized() const;
virtual int32_t InitMicrophone() RTC_LOCKS_EXCLUDED(mutex_);
virtual bool MicrophoneIsInitialized() const;
// Speaker volume controls
virtual int32_t SpeakerVolumeIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
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;
// Microphone volume controls
virtual int32_t MicrophoneVolumeIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
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;
// Microphone mute control
virtual int32_t MicrophoneMuteIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t SetMicrophoneMute(bool enable);
virtual int32_t MicrophoneMute(bool& enabled) const;
// Speaker mute control
virtual int32_t SpeakerMuteIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t SetSpeakerMute(bool enable);
virtual int32_t SpeakerMute(bool& enabled) const;
// Stereo support
virtual int32_t StereoPlayoutIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
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 PlayoutDelay(uint16_t& delayMS) const;
virtual void AttachAudioBuffer(AudioDeviceBuffer* audioBuffer)
RTC_LOCKS_EXCLUDED(mutex_);
private:
int32_t InitSpeakerLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
int32_t InitMicrophoneLocked() RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
virtual int32_t MicrophoneIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t MicrophoneIsAvailableLocked(bool& available)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
virtual int32_t SpeakerIsAvailable(bool& available)
RTC_LOCKS_EXCLUDED(mutex_);
virtual int32_t SpeakerIsAvailableLocked(bool& available)
RTC_EXCLUSIVE_LOCKS_REQUIRED(mutex_);
static void AtomicSet32(int32_t* theValue, int32_t newValue);
static int32_t AtomicGet32(int32_t* theValue);
static void logCAMsg(webrtc::LoggingSeverity sev,
const char* msg,
const char* err);
int32_t GetNumberDevices(AudioObjectPropertyScope scope,
AudioDeviceID scopedDeviceIds[],
uint32_t deviceListLength);
int32_t GetDeviceName(AudioObjectPropertyScope scope,
uint16_t index,
webrtc::ArrayView<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);
OSStatus implObjectListenerProc(AudioObjectID objectId,
UInt32 numberAddresses,
const AudioObjectPropertyAddress addresses[]);
int32_t HandleDeviceChange();
int32_t HandleStreamFormatChange(AudioObjectID objectId,
AudioObjectPropertyAddress propertyAddress);
int32_t HandleDataSourceChange(AudioObjectID objectId,
AudioObjectPropertyAddress propertyAddress);
int32_t 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)
RTC_LOCKS_EXCLUDED(mutex_);
OSStatus implOutConverterProc(UInt32* numberDataPackets,
AudioBufferList* data);
OSStatus implInDeviceIOProc(const AudioBufferList* inputData,
const AudioTimeStamp* inputTime)
RTC_LOCKS_EXCLUDED(mutex_);
OSStatus implInConverterProc(UInt32* numberDataPackets,
AudioBufferList* data);
static void RunCapture(void*);
static void RunRender(void*);
bool CaptureWorkerThread();
bool RenderWorkerThread();
bool KeyPressed();
AudioDeviceBuffer* _ptrAudioBuffer;
Mutex mutex_;
webrtc::Event _stopEventRec;
webrtc::Event _stopEvent;
// Only valid/running between calls to StartRecording and StopRecording.
webrtc::PlatformThread capture_worker_thread_;
// Only valid/running between calls to StartPlayout and StopPlayout.
webrtc::PlatformThread render_worker_thread_;
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];
bool _initialized;
bool _isShutDown;
bool _recording;
bool _playing;
bool _recIsInitialized;
bool _playIsInitialized;
// Atomically set varaibles
std::atomic<int32_t> _renderDeviceIsAlive;
std::atomic<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 std::atomic<int32_t> _captureDelayUs;
mutable std::atomic<int32_t> _renderDelayUs;
int32_t _renderDelayOffsetSamples;
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
#endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_MAC_H_
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