annotate projects/basic_network/render.cpp @ 41:4255ecbb9bec ultra-staging

Timers to measure performances, ultra experimental

/*
 * render.cpp
 *
 *  Created on: Oct 24, 2014
 *      Author: parallels
 */

#include "../../include/RTAudioSettings.h"
#include "../../include/render.h"
#include <cmath>
#include "../../include/client.h"
#include "../../include/RTAudio.h"	// to schedule lower prio parallel process
#include <rtdk.h>
#include <native/timer.h>
#include "../../include/PRU.h"
float gFrequency;
float gPhase;
float gInverseSampleRate;
int gCount=0;
networkData networkObject;
networkAudio netAudio;
extern PRU *gPRU;
AuxiliaryTask printIntervalTask;
AuxiliaryTask transmitReceiveAudioTask;
void transmitReceiveData(){ //transmit and receive asynchronous messages
//   	printf("transmitReceiveData auxiliary task has started\n");
//	while(!gShouldStop){
//		sendMessage(&networkObject);
//		receiveMessage(networkObject);
//		usleep(1000);
//	}
//	closeSockets();
}

void transmitReceiveAudio(){ //transmit and receive audio buffers
	sendAudio(&netAudio);
}


// initialise_render() is called once before the audio rendering starts.
// Use it to perform any initialisation and allocation which is dependent
// on the period size or sample rate.
//
// userData holds an opaque pointer to a data structure that was passed
// in from the call to initAudio().
//
// Return true on success; returning false halts the program.
bool initialise_render(int numMatrixChannels, int numDigitalChannels, int numAudioChannels,
					   int numMatrixFramesPerPeriod,
					   int numAudioFramesPerPeriod,
					   float matrixSampleRate, float audioSampleRate,
					   void *userData, RTAudioSettings *settings)
{
	// Retrieve a parameter passed in from the initAudio() call
	gFrequency = *(float *)userData;

	gInverseSampleRate = 1.0 / audioSampleRate;
	gPhase = 0.0;

//	networkObject.counter=&gCount;
//	networkObject.variables[0]=&gFrequency;
//	networkObject.variables[1]=&gPhase;
//	networkObject.numVariables=2;
//	netAudio.doneOnTime=1;
//	netAudio.index=0;
//	netAudio.currentBuffer=0;
//	setupSockets(settings->receivePort, settings->transmitPort, settings->serverName);
////	transmitReceiveDataTask=createAuxiliaryTaskLoop(*transmitReceiveData, 10, "transmit-receive-data");
////	scheduleAuxiliaryTask(transmitReceiveDataTask); //here it does not work
//	transmitReceiveAudioTask=createAuxiliaryTaskLoop(*transmitReceiveAudio, 98, "transmit-receive-audio");
	return true;
}

// render() is called regularly at the highest priority by the audio engine.
// Input and output are given from the audio hardware and the other
// ADCs and DACs (if available). If only audio is available, numMatrixFrames
// will be 0.

void render(int numAnalogFrames, int numAudioFrames, int numDigitalFrames, float *audioIn, float *audioOut,
		float *analogIn, float *analogOut, uint32_t *digital)
{
	for(int n = 0; n < numAudioFrames; n++) {
		float out = 0.7f * sinf(gPhase);
		float fake=0.1;
		for(int a=0; a<24; a++){
			fake = 0.7f * sinf(fake+out);
		}
		fake/=1000000000000000;
		gPhase += 2.0 * M_PI * gFrequency * gInverseSampleRate;
		if(gPhase > 2.0 * M_PI)
			gPhase -= 2.0 * M_PI;

//		for(int channel = 0; channel < gNumAudioChannels; channel++)
//			audioOut[n * gNumAudioChannels + channel] = audioIn[n * gNumAudioChannels + 0]+audioIn[n * gNumAudioChannels + 1];
		audioOut[n * gNumAudioChannels] = fake*0.0000000001;
		audioOut[n * gNumAudioChannels+1]=out;
		if(0==gCount){
//			scheduleAuxiliaryTask(transmitReceiveDataTask);
		}
//		if(netAudio.index==(NETWORK_AUDIO_BUFFER_SIZE)){ // when the buffer is ready ...
//			netAudio.index=0; //reset the counter
//			if(netAudio.doneOnTime==0)
//				rt_printf("Network buffer underrun :-{\n");
//			netAudio.timestamp=gCount;
//			netAudio.currentBuffer=!netAudio.currentBuffer; //switch buffer
//			netAudio.doneOnTime=0;
//			scheduleAuxiliaryTask(transmitReceiveAudioTask); //send the buffer
//		}
//		netAudio.buffers[netAudio.currentBuffer][netAudio.index++]=audioOut[n*gNumAudioChannels + 0];//copy channel 0 to the buffer
		gCount++;
	}
}

// cleanup_render() is called once at the end, after the audio has stopped.
// Release any resources that were allocated in initialise_render().

void cleanup_render()
{
//	closeSockets();
}