| /* |
| * 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_ |
