annotate examples/basic_libpd/render.cpp @ 349:6e454ebd9cc4 prerelease

Removed libpd_queued

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

#include <Bela.h>
#include <DigitalToMessage.h>
#include <cmath>
#include <Utilities.h>
#include <I2c_Codec.h>
#include <PRU.h>
#include <stdio.h>
#include "z_libpd.h"
#include "s_stuff.h"
#include <UdpServer.h>
#include <Midi.h>
//extern t_sample* sys_soundin;
//extern t_sample* sys_soundout;
// if you are 100% sure of what value was used to compile libpd/puredata, then
// you could #define this instead of getting it at runtime. It has proved to give some 0.3%
// performance boost when it is 8 (thanks to vectorize optimizations I guess).
int gBufLength;

float* gInBuf;
float* gOutBuf;

void pdnoteon(int ch, int pitch, int vel) {
  printf("noteon: %d %d %d\n", ch, pitch, vel);
}

void Bela_printHook(const char *recv){
	rt_printf("%s", recv);
}

void libpdReadFilesLoop(){
    while(!gShouldStop){
    	// check for modified sockets/file descriptors
    	// (libpd would normally do this every block WITHIN the audio thread)
    	// not sure if this is thread-safe at the moment
    	libpd_sys_microsleep(0);
        usleep(1000);
    }
}

#define PARSE_MIDI
AuxiliaryTask libpdReadFilesTask;
Midi midi;
//UdpServer udpServer;

void sendDigitalMessage(bool state, unsigned int delay, void* receiverName){
	libpd_float((char*)receiverName, (float)state);
//	rt_printf("%s: %d\n", (char*)receiverName, state);
}
char receiverNames[16][21]={
	{"bela_digitalIn11"},{"bela_digitalIn12"},{"bela_digitalIn13"},{"bela_digitalIn14"},{"bela_digitalIn15"},
	{"bela_digitalIn16"},{"bela_digitalIn17"},{"bela_digitalIn18"},{"bela_digitalIn19"},{"bela_digitalIn20"},
	{"bela_digitalIn21"},{"bela_digitalIn22"},{"bela_digitalIn23"},{"bela_digitalIn24"},{"bela_digitalIn25"},
	{"bela_digitalIn26"}
};

static DigitalToMessage** dtm;
static unsigned int analogChannelsInUse;
static unsigned int gLibpdBlockSize;
static unsigned int gChannelsInUse = 26;

bool setup(BelaContext *context, void *userData)
{
	analogChannelsInUse = min(context->analogChannels, gChannelsInUse - context->audioChannels - context->digitalChannels);
	dtm = new DigitalToMessage* [context->digitalChannels];
	if(context->digitalChannels > 0){
		for(unsigned int ch = 0; ch < context->digitalChannels; ++ch){
			dtm[ch] = new DigitalToMessage;
			dtm[ch]->setCallback(sendDigitalMessage, receiverNames[ch]);
			pinMode(context, 0, ch, INPUT);
		}
	}
	midi.readFrom(0);
	midi.writeTo(0);
#ifdef PARSE_MIDI
	midi.enableParser(true);
#else
	midi.enableParser(false);
#endif /* PARSE_MIDI */
//	gChannelsInUse = min((int)(context->analogChannels+context->audioChannels), (int)gChannelsInUse);
//	udpServer.bindToPort(1234);

	gLibpdBlockSize = libpd_blocksize();
	// check that we are not running with a blocksize smaller than gLibPdBlockSize
	// it would still work, but the load would be executed unevenly between calls to render
	if(context->audioFrames < gLibpdBlockSize){
		fprintf(stderr, "Error: minimum block size must be %d\n", gLibpdBlockSize);
		return false;
	}
	// set hooks before calling libpd_init
	libpd_set_printhook(Bela_printHook);
	libpd_set_floathook(Bela_floatHook);
	libpd_set_noteonhook(pdnoteon);
	//TODO: add hooks for other midi events and generate MIDI output appropriately
	libpd_init();
	//TODO: ideally, we would analyse the ASCII of the patch file and find the in/outs to use
	libpd_init_audio(gChannelsInUse, gChannelsInUse, context->audioSampleRate);

	libpd_start_message(1); // one entry in list
	libpd_add_float(1.0f);
	libpd_finish_message("pd", "dsp");

	gBufLength = max(gLibpdBlockSize, context->audioFrames);

	char file[] = "_main.pd";
	char folder[] = "./";
	// open patch       [; pd open file folder(
	libpd_openfile(file, folder);
	gInBuf = libpd_get_sys_soundin();
	gOutBuf = libpd_get_sys_soundout();
	libpdReadFilesTask = Bela_createAuxiliaryTask(libpdReadFilesLoop, 60, "libpdReadFiles");
	Bela_scheduleAuxiliaryTask(libpdReadFilesTask);


	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(BelaContext *context, void *userData)
{
	int num;
	// the safest thread-safe option to handle MIDI input is to process the MIDI buffer
	// from the audio thread.
#ifdef PARSE_MIDI
	while((num = midi.getParser()->numAvailableMessages()) > 0){
		static MidiChannelMessage message;
		message = midi.getParser()->getNextChannelMessage();
		//message.prettyPrint(); // use this to print beautified message (channel, data bytes)
		switch(message.getType()){
			case kmmNoteOn:
			{
				int noteNumber = message.getDataByte(0);
				int velocity = message.getDataByte(1);
				int channel = message.getChannel();
				libpd_noteon(channel, noteNumber, velocity);
				break;
			}
			case kmmNoteOff:
			{
				/* PureData does not seem to handle noteoff messages as per the MIDI specs,
				 * so that the noteoff velocity is ignored. Here we convert them to noteon
				 * with a velocity of 0.
				 */
				int noteNumber = message.getDataByte(0);
//				int velocity = message.getDataByte(1); // would be ignored by Pd
				int channel = message.getChannel();
				libpd_noteon(channel, noteNumber, 0);
				break;
			}
			case kmmControlChange:
			{
				int channel = message.getChannel();
				int controller = message.getDataByte(0);
				int value = message.getDataByte(1);
				libpd_controlchange(channel, controller, value);
				break;
			}
			case kmmProgramChange:
			{
				int channel = message.getChannel();
				int program = message.getDataByte(0);
				libpd_programchange(channel, program);
				break;
			}
			case kmmPolyphonicKeyPressure:
			{
				int channel = message.getChannel();
				int pitch = message.getDataByte(0);
				int value = message.getDataByte(1);
				libpd_polyaftertouch(channel, pitch, value);
				break;
			}
			case kmmChannelPressure:
			{
				int channel = message.getChannel();
				int value = message.getDataByte(0);
				libpd_aftertouch(channel, value);
				break;
			}
			case kmmPitchBend:
			{
				int channel = message.getChannel();
				int value =  (message.getDataByte(1) << 7)| message.getDataByte(0);
				libpd_pitchbend(channel, value);
				break;
			}
			case kmmNone:
			case kmmAny:
				break;
		}
	}
#else
	int input;
	while((input = midi.getInput()) >= 0){
		libpd_midibyte(0, input);
	}
#endif /* PARSE_MIDI */

	static unsigned int numberOfPdBlocksToProcess = gBufLength / gLibpdBlockSize;
	for(unsigned int tick = 0; tick < numberOfPdBlocksToProcess; ++tick){
		unsigned int audioFrameBase = gLibpdBlockSize * tick;
		unsigned int j;
		unsigned int k;
		float* p0;
		float* p1;
		for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
			for (k = 0, p1 = p0; k < context->audioChannels; k++, p1 += gLibpdBlockSize) {
				*p1 = audioRead(context, audioFrameBase + j, k);
			}
		}
		// then analogs
		// this loop resamples by ZOH, as needed, using m
		if(context->analogChannels == 8 ){ //hold the value for two frames
			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					unsigned int analogFrame = (audioFrameBase + j) / 2;
					*p1 = analogRead(context, analogFrame, k);
				}
			}
		} else if(context->analogChannels == 4){ //write every frame
			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					unsigned int analogFrame = audioFrameBase + j;
					*p1 = analogRead(context, analogFrame, k);
				}
			}
		} else if(context->analogChannels == 2){ //drop every other frame
			for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					unsigned int analogFrame = (audioFrameBase + j) * 2;
					*p1 = analogRead(context, analogFrame, k);
				}
			}
		}

		//then digital
		 //TODO: in multiple places below we assume that the number of digitals is same as number of audio
		// digital in at message-rate
		for(unsigned int n = 0; n < context->digitalChannels; ++n){
			// note that we consider only the first sample of the block
			// considering all of them is notably more expensive
			// TODO: only process the channels marked as such
			dtm[n]->process(n + 16, &context->digital[audioFrameBase], 1);
		}
		// digital in at signal-rate
		for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
			for (k = 0, p1 = p0 + gLibpdBlockSize * (context->audioChannels + 8);
					k < 16; ++k, p1 += gLibpdBlockSize) {
				// note that we only write the last sample of the block
				// writing all of them is notably more expensive.

				// TODO: only process the channels marked as such
				unsigned int digitalFrame = audioFrameBase + j;
				*p1 = digitalRead(context, digitalFrame, k);
			}
		}

		libpd_process_sys(); // process the block

		//digital out
		// digital at message-rate
		for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) {
			unsigned int digitalFrame = (audioFrameBase + j);
			for (k = 0, p1 = p0  + gLibpdBlockSize * (context->audioChannels + 8);
					k < context->digitalChannels; k++, p1 += gLibpdBlockSize) {
				// TODO: only process the channels marked as such
				digitalWriteOnce(context, digitalFrame, k, *p1 > 0.5);
			}
		}

		//audio
		for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j++, p0++) {
			for (k = 0, p1 = p0; k < context->audioChannels; k++, p1 += gLibpdBlockSize) {
				audioWrite(context, audioFrameBase + j, k, *p1);
			}
		}
		//analog
		if(context->analogChannels == 8){
			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j += 2, p0 += 2) { //write every two frames
				unsigned int analogFrame = (audioFrameBase + j) / 2;
				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					analogWriteOnce(context, analogFrame, k, *p1);
				}
			}
		} else if(context->analogChannels == 4){ //write every frame
			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; ++j, ++p0) {
				unsigned int analogFrame = (audioFrameBase + j);
				for (k = 0, p1 = p0  + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					analogWriteOnce(context, analogFrame, k, *p1);
				}
			}
		} else if(context->analogChannels == 2){ //write every frame twice
			for (j = 0, p0 = gOutBuf; j < gLibpdBlockSize; j++, p0++) {
				for (k = 0, p1 = p0 + gLibpdBlockSize * context->audioChannels; k < analogChannelsInUse; k++, p1 += gLibpdBlockSize) {
					int analogFrame = audioFrameBase * 2 + j * 2;
					analogWriteOnce(context, analogFrame, k, *p1);
					analogWriteOnce(context, analogFrame + 1, k, *p1);
				}
			}
		}
	}
}

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

void cleanup(BelaContext *context, void *userData)
{
	delete[] dtm;
}