/* * 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 WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_IPHONE_H #include #include "audio_device_generic.h" #include "critical_section_wrapper.h" namespace webrtc { class ThreadWrapper; const WebRtc_UWord32 N_REC_SAMPLES_PER_SEC = 44000; const WebRtc_UWord32 N_PLAY_SAMPLES_PER_SEC = 44000; const WebRtc_UWord32 N_REC_CHANNELS = 1; // default is mono recording const WebRtc_UWord32 N_PLAY_CHANNELS = 1; // default is mono 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); // Number of 10 ms recording blocks in recording buffer const WebRtc_UWord16 N_REC_BUFFERS = 20; class AudioDeviceIPhone : public AudioDeviceGeneric { public: AudioDeviceIPhone(const WebRtc_Word32 id); ~AudioDeviceIPhone(); // 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, char name[kAdmMaxDeviceNameSize], char guid[kAdmMaxGuidSize]); virtual WebRtc_Word32 RecordingDeviceName(WebRtc_UWord16 index, char name[kAdmMaxDeviceNameSize], char 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); // Reset Audio Deivce (for mobile devices only) virtual WebRtc_Word32 ResetAudioDevice(); // enable or disable loud speaker (for iphone only) virtual WebRtc_Word32 SetLoudspeakerStatus(bool enable); virtual WebRtc_Word32 GetLoudspeakerStatus(bool& enabled) const; private: void Lock() { _critSect.Enter(); } void UnLock() { _critSect.Leave(); } WebRtc_Word32 Id() { return _id; } // Init and shutdown WebRtc_Word32 InitPlayOrRecord(); WebRtc_Word32 ShutdownPlayOrRecord(); void UpdateRecordingDelay(); void UpdatePlayoutDelay(); static OSStatus RecordProcess(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *WebRtc_Word32imeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); static OSStatus PlayoutProcess(void *inRefCon, AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *WebRtc_Word32imeStamp, UInt32 inBusNumber, UInt32 inNumberFrames, AudioBufferList *ioData); OSStatus RecordProcessImpl(AudioUnitRenderActionFlags *ioActionFlags, const AudioTimeStamp *WebRtc_Word32imeStamp, WebRtc_UWord32 inBusNumber, WebRtc_UWord32 inNumberFrames); OSStatus PlayoutProcessImpl(WebRtc_UWord32 inNumberFrames, AudioBufferList *ioData); static bool RunCapture(void* ptrThis); bool CaptureWorkerThread(); private: AudioDeviceBuffer* _ptrAudioBuffer; CriticalSectionWrapper& _critSect; ThreadWrapper* _captureWorkerThread; WebRtc_UWord32 _captureWorkerThreadId; WebRtc_Word32 _id; AudioUnit _auRemoteIO; private: bool _initialized; bool _isShutDown; bool _recording; bool _playing; bool _recIsInitialized; bool _playIsInitialized; bool _recordingDeviceIsSpecified; bool _playoutDeviceIsSpecified; bool _micIsInitialized; bool _speakerIsInitialized; bool _AGC; // The sampling rate to use with Audio Device Buffer WebRtc_UWord32 _adbSampFreq; // Delay calculation WebRtc_UWord32 _recordingDelay; WebRtc_UWord32 _playoutDelay; WebRtc_UWord32 _playoutDelayMeasurementCounter; WebRtc_UWord32 _recordingDelayHWAndOS; WebRtc_UWord32 _recordingDelayMeasurementCounter; // Errors and warnings count WebRtc_UWord16 _playWarning; WebRtc_UWord16 _playError; WebRtc_UWord16 _recWarning; WebRtc_UWord16 _recError; // Playout buffer, needed for 44.0 / 44.1 kHz mismatch WebRtc_Word16 _playoutBuffer[ENGINE_PLAY_BUF_SIZE_IN_SAMPLES]; WebRtc_UWord32 _playoutBufferUsed; // How much is filled // Recording buffers WebRtc_Word16 _recordingBuffer[N_REC_BUFFERS][ENGINE_REC_BUF_SIZE_IN_SAMPLES]; WebRtc_UWord32 _recordingLength[N_REC_BUFFERS]; WebRtc_UWord32 _recordingSeqNumber[N_REC_BUFFERS]; WebRtc_UWord32 _recordingCurrentSeq; // Current total size all data in buffers, used for delay estimate WebRtc_UWord32 _recordingBufferTotalSize; }; } // namespace webrtc #endif // MODULES_AUDIO_DEVICE_MAIN_SOURCE_MAC_AUDIO_DEVICE_IPHONE_H_