// // OutputAVFoundation.h // Cog // // Created by Christopher Snowhill on 6/23/22. // Copyright 2022 Christopher Snowhill. All rights reserved. // #import #import #import #import #import #import #import #ifdef __cplusplus #import using std::atomic_long; #else #import #endif #import "Downmix.h" #import #import "HeadphoneFilter.h" //#define OUTPUT_LOG #ifdef OUTPUT_LOG #import #endif @class OutputNode; @class FSurroundFilter; @interface OutputAVFoundation : NSObject { OutputNode *outputController; BOOL r8bDone; void *r8bstate, *r8bold; void *r8bvis; double lastVisRate; BOOL stopInvoked; BOOL stopCompleted; BOOL running; BOOL stopping; BOOL stopped; BOOL started; BOOL paused; BOOL restarted; BOOL commandStop; BOOL eqEnabled; BOOL eqInitialized; BOOL streamFormatStarted; BOOL streamFormatChanged; double secondsHdcdSustained; BOOL defaultdevicelistenerapplied; BOOL currentdevicelistenerapplied; BOOL devicealivelistenerapplied; BOOL observersapplied; BOOL outputdevicechanged; float volume; float eqPreamp; AudioDeviceID outputDeviceID; AudioStreamBasicDescription realStreamFormat; // stream format pre-hrtf AudioStreamBasicDescription streamFormat; // stream format last seen in render callback AudioStreamBasicDescription realNewFormat; // in case of resampler flush AudioStreamBasicDescription newFormat; // in case of resampler flush AudioStreamBasicDescription visFormat; // Mono format for vis uint32_t realStreamChannelConfig; uint32_t streamChannelConfig; uint32_t realNewChannelConfig; uint32_t newChannelConfig; AVSampleBufferAudioRenderer *audioRenderer; AVSampleBufferRenderSynchronizer *renderSynchronizer; CMAudioFormatDescriptionRef audioFormatDescription; id currentPtsObserver; NSLock *currentPtsLock; CMTime currentPts, lastPts; double secondsLatency; CMTime outputPts, trackPts, lastCheckpointPts; AudioTimeStamp timeStamp; size_t _bufferSize; AudioUnit _eq; DownmixProcessor *downmixerForVis; VisualizationController *visController; BOOL enableHrtf; HeadphoneFilter *hrtf; BOOL enableFSurround; BOOL FSurroundDelayRemoved; int inputBufferLastTime; FSurroundFilter *fsurround; BOOL resetStreamFormat; float *samplePtr; float tempBuffer[512 * 32]; float r8bTempBuffer[4096 * 32]; float inputBuffer[4096 * 32]; // 4096 samples times maximum supported channel count float fsurroundBuffer[8192 * 6]; float hrtfBuffer[4096 * 2]; float eqBuffer[4096 * 32]; float visAudio[4096]; float visTemp[8192]; #ifdef OUTPUT_LOG FILE *_logFile; #endif } - (id)initWithController:(OutputNode *)c; - (BOOL)setup; - (OSStatus)setOutputDeviceByID:(AudioDeviceID)deviceID; - (BOOL)setOutputDeviceWithDeviceDict:(NSDictionary *)deviceDict; - (void)start; - (void)pause; - (void)resume; - (void)stop; - (double)latency; - (void)setVolume:(double)v; - (void)setEqualizerEnabled:(BOOL)enabled; - (void)sustainHDCD; @end