robert@464: /*
robert@464:  * render.cpp
robert@464:  *
robert@464:  *  Created on: Oct 24, 2014
robert@464:  *      Author: parallels
robert@464:  */
robert@464: 
robert@464: #include <Bela.h>
robert@464: #include <DigitalChannelManager.h>
robert@464: #include <cmath>
robert@464: #include <I2c_Codec.h>
robert@464: #include <PRU.h>
robert@464: #include <stdio.h>
robert@464: #include "z_libpd.h"
robert@464: #include "s_stuff.h"
robert@464: #include <UdpServer.h>
robert@464: #include <Midi.h>
robert@464: //extern t_sample* sys_soundin;
robert@464: //extern t_sample* sys_soundout;
robert@464: // if you are 100% sure of what value was used to compile libpd/puredata, then
robert@464: // you could #define this instead of getting it at runtime. It has proved to give some 0.3%
robert@464: // performance boost when it is 8 (thanks to vectorize optimizations I guess).
robert@464: int gBufLength;
robert@464: 
robert@464: float* gInBuf;
robert@464: float* gOutBuf;
robert@464: 
robert@464: void pdnoteon(int ch, int pitch, int vel) {
robert@464:   printf("noteon: %d %d %d\n", ch, pitch, vel);
robert@464: }
robert@464: 
robert@464: void Bela_printHook(const char *recv){
robert@464: 	rt_printf("%s", recv);
robert@464: }
robert@464: 
robert@464: void libpdReadFilesLoop(){
robert@464:     while(!gShouldStop){
robert@464:     	// check for modified sockets/file descriptors
robert@464:     	// (libpd would normally do this every block WITHIN the audio thread)
robert@464:     	// not sure if this is thread-safe at the moment
robert@464:     	libpd_sys_microsleep(0);
robert@464:         usleep(1000);
robert@464:     }
robert@464: }
robert@464: 
robert@464: #define PARSE_MIDI
robert@464: static AuxiliaryTask libpdReadFilesTask;
robert@464: static Midi midi;
robert@464: static DigitalChannelManager dcm;
robert@464: //UdpServer udpServer;
robert@464: 
robert@464: void sendDigitalMessage(bool state, unsigned int delay, void* receiverName){
robert@464: 	libpd_float((char*)receiverName, (float)state);
robert@464: //	rt_printf("%s: %d\n", (char*)receiverName, state);
robert@464: }
robert@464: 
robert@464: #define LIBPD_DIGITAL_OFFSET 11 // digitals are preceded by 2 audio and 8 analogs (even if using a different number of analogs)
robert@464: 
robert@464: void Bela_messageHook(const char *source, const char *symbol, int argc, t_atom *argv){
robert@464: 	if(strcmp(source, "bela_setDigital") == 0){
robert@464: 		// symbol is the direction, argv[0] is the channel, argv[1] (optional)
robert@464: 		// is signal("sig" or "~") or message("message", default) rate
robert@464: 		bool isMessageRate = true; // defaults to message rate
robert@464: 		bool direction = 0; // initialize it just to avoid the compiler's warning
robert@464: 		bool disable = false;
robert@464: 		if(strcmp(symbol, "in") == 0){
robert@464: 			direction = INPUT;
robert@464: 		} else if(strcmp(symbol, "out") == 0){
robert@464: 			direction = OUTPUT;
robert@464: 		} else if(strcmp(symbol, "disable") == 0){
robert@464: 			disable = true;
robert@464: 		} else {
robert@464: 			return;
robert@464: 		}
robert@464: 		if(argc == 0){
robert@464: 			return;
robert@464: 		} else if (libpd_is_float(&argv[0]) == false){
robert@464: 			return;
robert@464: 		}
robert@464: 		int channel = libpd_get_float(&argv[0]) - LIBPD_DIGITAL_OFFSET;
robert@464: 		if(disable == true){
robert@464: 			dcm.unmanage(channel);
robert@464: 			return;
robert@464: 		}
robert@464: 		if(argc >= 2){
robert@464: 			t_atom* a = &argv[1];
robert@464: 			if(libpd_is_symbol(a)){
robert@464: 				char *s = libpd_get_symbol(a);
robert@464: 				if(strcmp(s, "~") == 0  || strncmp(s, "sig", 3) == 0){
robert@464: 					isMessageRate = false;
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 		dcm.manage(channel, direction, isMessageRate);
robert@464: 	}
robert@464: }
robert@464: 
robert@464: void Bela_floatHook(const char *source, float value){
robert@464: 	// let's make this as optimized as possible for built-in digital Out parsing
robert@464: 	// the built-in digital receivers are of the form "bela_digitalOutXX" where XX is between 11 and 26
robert@464: 	static int prefixLength = 15; // strlen("bela_digitalOut")
robert@464: 	if(strncmp(source, "bela_digitalOut", prefixLength)==0){
robert@464: 		if(source[prefixLength] != 0){ //the two ifs are used instead of if(strlen(source) >= prefixLength+2)
robert@464: 			if(source[prefixLength + 1] != 0){
robert@464: 				// quickly convert the suffix to integer, assuming they are numbers, avoiding to call atoi
robert@464: 				int receiver = ((source[prefixLength] - 48) * 10);
robert@464: 				receiver += (source[prefixLength+1] - 48);
robert@464: 				unsigned int channel = receiver - 11; // go back to the actual Bela digital channel number
robert@464: 				if(channel >= 0 && channel < 16){ //16 is the hardcoded value for the number of digital channels
robert@464: 					dcm.setValue(channel, value);
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 	}
robert@464: }
robert@464: 
robert@464: char receiverNames[16][21]={
robert@464: 	{"bela_digitalIn11"},{"bela_digitalIn12"},{"bela_digitalIn13"},{"bela_digitalIn14"},{"bela_digitalIn15"},
robert@464: 	{"bela_digitalIn16"},{"bela_digitalIn17"},{"bela_digitalIn18"},{"bela_digitalIn19"},{"bela_digitalIn20"},
robert@464: 	{"bela_digitalIn21"},{"bela_digitalIn22"},{"bela_digitalIn23"},{"bela_digitalIn24"},{"bela_digitalIn25"},
robert@464: 	{"bela_digitalIn26"}
robert@464: };
robert@464: 
robert@464: static unsigned int analogChannelsInUse;
robert@464: static unsigned int gLibpdBlockSize;
robert@464: static unsigned int gChannelsInUse = 26;
robert@464: 
robert@464: bool setup(BelaContext *context, void *userData)
robert@464: {
robert@464: 	dcm.setCallback(sendDigitalMessage);
robert@464: 	analogChannelsInUse = min(context->analogChannels, gChannelsInUse - context->audioChannels - context->digitalChannels);
robert@464: 	if(context->digitalChannels > 0){
robert@464: 		for(unsigned int ch = 0; ch < context->digitalChannels; ++ch){
robert@464: 			dcm.setCallbackArgument(ch, receiverNames[ch]);
robert@464: 		}
robert@464: 	}
robert@464: 	midi.readFrom(0);
robert@464: 	midi.writeTo(0);
robert@464: #ifdef PARSE_MIDI
robert@464: 	midi.enableParser(true);
robert@464: #else
robert@464: 	midi.enableParser(false);
robert@464: #endif /* PARSE_MIDI */
robert@464: //	gChannelsInUse = min((int)(context->analogChannels+context->audioChannels), (int)gChannelsInUse);
robert@464: //	udpServer.bindToPort(1234);
robert@464: 
robert@464: 	gLibpdBlockSize = libpd_blocksize();
robert@464: 	// check that we are not running with a blocksize smaller than gLibPdBlockSize
robert@464: 	// it would still work, but the load would be executed unevenly between calls to render
robert@464: 	if(context->audioFrames < gLibpdBlockSize){
robert@464: 		fprintf(stderr, "Error: minimum block size must be %d\n", gLibpdBlockSize);
robert@464: 		return false;
robert@464: 	}
robert@464: 	// set hooks before calling libpd_init
robert@464: 	libpd_set_printhook(Bela_printHook);
robert@464: 	libpd_set_floathook(Bela_floatHook);
robert@464: 	libpd_set_messagehook(Bela_messageHook);
robert@464: 	libpd_set_noteonhook(pdnoteon);
robert@464: 	//TODO: add hooks for other midi events and generate MIDI output appropriately
robert@464: 	libpd_init();
robert@464: 	//TODO: ideally, we would analyse the ASCII of the patch file and find out which in/outs to use
robert@464: 	libpd_init_audio(gChannelsInUse, gChannelsInUse, context->audioSampleRate);
robert@464: 	gInBuf = libpd_get_sys_soundin();
robert@464: 	gOutBuf = libpd_get_sys_soundout();
robert@464: 
robert@464: 	libpd_start_message(1); // one entry in list
robert@464: 	libpd_add_float(1.0f);
robert@464: 	libpd_finish_message("pd", "dsp");
robert@464: 
robert@464: 	gBufLength = max(gLibpdBlockSize, context->audioFrames);
robert@464: 
robert@464: 
robert@464: 	// bind your receivers here
robert@464: 	libpd_bind("bela_digitalOut11");
robert@464: 	libpd_bind("bela_digitalOut12");
robert@464: 	libpd_bind("bela_digitalOut13");
robert@464: 	libpd_bind("bela_digitalOut14");
robert@464: 	libpd_bind("bela_digitalOut15");
robert@464: 	libpd_bind("bela_digitalOut16");
robert@464: 	libpd_bind("bela_digitalOut17");
robert@464: 	libpd_bind("bela_digitalOut18");
robert@464: 	libpd_bind("bela_digitalOut19");
robert@464: 	libpd_bind("bela_digitalOut20");
robert@464: 	libpd_bind("bela_digitalOut21");
robert@464: 	libpd_bind("bela_digitalOut22");
robert@464: 	libpd_bind("bela_digitalOut23");
robert@464: 	libpd_bind("bela_digitalOut24");
robert@464: 	libpd_bind("bela_digitalOut25");
robert@464: 	libpd_bind("bela_digitalOut26");
robert@464: 	libpd_bind("bela_setDigital");
robert@464: 
robert@464: 	char file[] = "_main.pd";
robert@464: 	char folder[] = "./";
robert@464: 	// open patch       [; pd open file folder(
robert@464: 	libpd_openfile(file, folder);
robert@464: 	libpdReadFilesTask = Bela_createAuxiliaryTask(libpdReadFilesLoop, 60, "libpdReadFiles");
robert@464: 	Bela_scheduleAuxiliaryTask(libpdReadFilesTask);
robert@464: 
robert@464: 
robert@464: 	return true;
robert@464: }
robert@464: 
robert@464: // render() is called regularly at the highest priority by the audio engine.
robert@464: // Input and output are given from the audio hardware and the other
robert@464: // ADCs and DACs (if available). If only audio is available, numMatrixFrames
robert@464: // will be 0.
robert@464: 
robert@464: void render(BelaContext *context, void *userData)
robert@464: {
robert@464: 	int num;
robert@464: 	// the safest thread-safe option to handle MIDI input is to process the MIDI buffer
robert@464: 	// from the audio thread.
robert@464: #ifdef PARSE_MIDI
robert@464: 	while((num = midi.getParser()->numAvailableMessages()) > 0){
robert@464: 		static MidiChannelMessage message;
robert@464: 		message = midi.getParser()->getNextChannelMessage();
robert@464: 		//message.prettyPrint(); // use this to print beautified message (channel, data bytes)
robert@464: 		switch(message.getType()){
robert@464: 			case kmmNoteOn:
robert@464: 			{
robert@464: 				int noteNumber = message.getDataByte(0);
robert@464: 				int velocity = message.getDataByte(1);
robert@464: 				int channel = message.getChannel();
robert@464: 				libpd_noteon(channel, noteNumber, velocity);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmNoteOff:
robert@464: 			{
robert@464: 				/* PureData does not seem to handle noteoff messages as per the MIDI specs,
robert@464: 				 * so that the noteoff velocity is ignored. Here we convert them to noteon
robert@464: 				 * with a velocity of 0.
robert@464: 				 */
robert@464: 				int noteNumber = message.getDataByte(0);
robert@464: //				int velocity = message.getDataByte(1); // would be ignored by Pd
robert@464: 				int channel = message.getChannel();
robert@464: 				libpd_noteon(channel, noteNumber, 0);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmControlChange:
robert@464: 			{
robert@464: 				int channel = message.getChannel();
robert@464: 				int controller = message.getDataByte(0);
robert@464: 				int value = message.getDataByte(1);
robert@464: 				libpd_controlchange(channel, controller, value);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmProgramChange:
robert@464: 			{
robert@464: 				int channel = message.getChannel();
robert@464: 				int program = message.getDataByte(0);
robert@464: 				libpd_programchange(channel, program);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmPolyphonicKeyPressure:
robert@464: 			{
robert@464: 				int channel = message.getChannel();
robert@464: 				int pitch = message.getDataByte(0);
robert@464: 				int value = message.getDataByte(1);
robert@464: 				libpd_polyaftertouch(channel, pitch, value);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmChannelPressure:
robert@464: 			{
robert@464: 				int channel = message.getChannel();
robert@464: 				int value = message.getDataByte(0);
robert@464: 				libpd_aftertouch(channel, value);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmPitchBend:
robert@464: 			{
robert@464: 				int channel = message.getChannel();
robert@464: 				int value =  (message.getDataByte(1) << 7)| message.getDataByte(0);
robert@464: 				libpd_pitchbend(channel, value);
robert@464: 				break;
robert@464: 			}
robert@464: 			case kmmNone:
robert@464: 			case kmmAny:
robert@464: 				break;
robert@464: 		}
robert@464: 	}
robert@464: #else
robert@464: 	int input;
robert@464: 	while((input = midi.getInput()) >= 0){
robert@464: 		libpd_midibyte(0, input);
robert@464: 	}
robert@464: #endif /* PARSE_MIDI */
robert@464: 
robert@464: 	static unsigned int numberOfPdBlocksToProcess = gBufLength / gLibpdBlockSize;
robert@464: 
robert@464: 	// these are reset at every audio callback. Persistence across audio callbacks
robert@464: 	// is handled by the core code.
robert@464: //	setDataOut = 0;
robert@464: //	clearDataOut = 0;
robert@464: 
robert@464: 	for(unsigned int tick = 0; tick < numberOfPdBlocksToProcess; ++tick){
robert@464: 		unsigned int audioFrameBase = gLibpdBlockSize * tick;
robert@464: 		unsigned int j;
robert@464: 		unsigned int k;
robert@464: 		float* p0;
robert@464: 		float* p1;
robert@464: 		for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 			for (k = 0, p1 = p0; k < context->audioChannels; k++, p1 += gLibpdBlockSize) {
robert@464: 				*p1 = audioRead(context, audioFrameBase + j, k);
robert@464: 			}
robert@464: 		}
robert@464: 		// then analogs
robert@464: 		// this loop resamples by ZOH, as needed, using m
robert@464: 		if(context->analogChannels == 8 ){ //hold the value for two frames
robert@464: 			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					unsigned int analogFrame = (audioFrameBase + j) / 2;
robert@464: 					*p1 = analogRead(context, analogFrame, k);
robert@464: 				}
robert@464: 			}
robert@464: 		} else if(context->analogChannels == 4){ //write every frame
robert@464: 			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					unsigned int analogFrame = audioFrameBase + j;
robert@464: 					*p1 = analogRead(context, analogFrame, k);
robert@464: 				}
robert@464: 			}
robert@464: 		} else if(context->analogChannels == 2){ //drop every other frame
robert@464: 			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					unsigned int analogFrame = (audioFrameBase + j) * 2;
robert@464: 					*p1 = analogRead(context, analogFrame, k);
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 
robert@464: 		//then digital
robert@464: 		// note: in multiple places below we assume that the number of digitals is same as number of audio
robert@464: 		// digital in at message-rate
robert@464: 		dcm.processInput(&context->digital[audioFrameBase], gLibpdBlockSize);
robert@464: 
robert@464: 		// digital in at signal-rate
robert@464: 		for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 			unsigned int digitalFrame = audioFrameBase + j;
robert@464: 			for (k = 0, p1 = p0 + gLibpdBlockSize * (context->audioChannels + 8);
robert@464: 					k < 16; ++k, p1 += gLibpdBlockSize) {
robert@464: 				if(dcm.isSignalRate(k) && dcm.isInput(k)){ // only process input channels that are handled at signal rate
robert@464: 					*p1 = digitalRead(context, digitalFrame, k);
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 
robert@464: 		libpd_process_sys(); // process the block
robert@464: 
robert@464: 		//digital out
robert@464: 		// digital out at signal-rate
robert@464: 		for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) {
robert@464: 			unsigned int digitalFrame = (audioFrameBase + j);
robert@464: 			for (k = 0, p1 = p0  + gLibpdBlockSize * (context->audioChannels + 8);
robert@464: 					k < context->digitalChannels; k++, p1 += gLibpdBlockSize) {
robert@464: 				if(dcm.isSignalRate(k) && dcm.isOutput(k)){ // only process output channels that are handled at signal rate
robert@464: 					digitalWriteOnce(context, digitalFrame, k, *p1 > 0.5);
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 
robert@464: 		// digital out at message-rate
robert@464: 		dcm.processOutput(&context->digital[audioFrameBase], gLibpdBlockSize);
robert@464: 
robert@464: 		//audio
robert@464: 		for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 			for (k = 0, p1 = p0; k < context->audioChannels; k++, p1 += gLibpdBlockSize) {
robert@464: 				audioWrite(context, audioFrameBase + j, k, *p1);
robert@464: 			}
robert@464: 		}
robert@464: 
robert@464: 		//analog
robert@464: 		if(context->analogChannels == 8){
robert@464: 			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j += 2, p0 += 2) { //write every two frames
robert@464: 				unsigned int analogFrame = (audioFrameBase + j) / 2;
robert@464: 				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					analogWriteOnce(context, analogFrame, k, *p1);
robert@464: 				}
robert@464: 			}
robert@464: 		} else if(context->analogChannels == 4){ //write every frame
robert@464: 			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) {
robert@464: 				unsigned int analogFrame = (audioFrameBase + j);
robert@464: 				for (k = 0, p1 = p0  + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					analogWriteOnce(context, analogFrame, k, *p1);
robert@464: 				}
robert@464: 			}
robert@464: 		} else if(context->analogChannels == 2){ //write every frame twice
robert@464: 			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j++, p0++) {
robert@464: 				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
robert@464: 					int analogFrame = audioFrameBase * 2 + j * 2;
robert@464: 					analogWriteOnce(context, analogFrame, k, *p1);
robert@464: 					analogWriteOnce(context, analogFrame + 1, k, *p1);
robert@464: 				}
robert@464: 			}
robert@464: 		}
robert@464: 	}
robert@464: }
robert@464: 
robert@464: // cleanup() is called once at the end, after the audio has stopped.
robert@464: // Release any resources that were allocated in setup().
robert@464: 
robert@464: void cleanup(BelaContext *context, void *userData)
robert@464: {
robert@464: }