diff core/default_libpd_render.cpp @ 467:03a2cd5f151b prerelease

Libpd headers moved to include/, rm useless basic_libpd example, fixed Makefile to actually build default_libpd_render

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/core/default_libpd_render.cpp	Mon Jun 20 16:57:35 2016 +0100
@@ -0,0 +1,403 @@
+/*
+ * render.cpp
+ *
+ *  Created on: Oct 24, 2014
+ *      Author: parallels
+ */
+
+#include <Bela.h>
+#include <DigitalChannelManager.h>
+#include <cmath>
+#include <I2c_Codec.h>
+#include <PRU.h>
+#include <stdio.h>
+#include <libpd/z_libpd.h>
+#include <libpd/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
+static AuxiliaryTask libpdReadFilesTask;
+static Midi midi;
+static DigitalChannelManager dcm;
+//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);
+}
+
+#define LIBPD_DIGITAL_OFFSET 11 // digitals are preceded by 2 audio and 8 analogs (even if using a different number of analogs)
+
+void Bela_messageHook(const char *source, const char *symbol, int argc, t_atom *argv){
+	if(strcmp(source, "bela_setDigital") == 0){
+		// symbol is the direction, argv[0] is the channel, argv[1] (optional)
+		// is signal("sig" or "~") or message("message", default) rate
+		bool isMessageRate = true; // defaults to message rate
+		bool direction = 0; // initialize it just to avoid the compiler's warning
+		bool disable = false;
+		if(strcmp(symbol, "in") == 0){
+			direction = INPUT;
+		} else if(strcmp(symbol, "out") == 0){
+			direction = OUTPUT;
+		} else if(strcmp(symbol, "disable") == 0){
+			disable = true;
+		} else {
+			return;
+		}
+		if(argc == 0){
+			return;
+		} else if (libpd_is_float(&argv[0]) == false){
+			return;
+		}
+		int channel = libpd_get_float(&argv[0]) - LIBPD_DIGITAL_OFFSET;
+		if(disable == true){
+			dcm.unmanage(channel);
+			return;
+		}
+		if(argc >= 2){
+			t_atom* a = &argv[1];
+			if(libpd_is_symbol(a)){
+				char *s = libpd_get_symbol(a);
+				if(strcmp(s, "~") == 0  || strncmp(s, "sig", 3) == 0){
+					isMessageRate = false;
+				}
+			}
+		}
+		dcm.manage(channel, direction, isMessageRate);
+	}
+}
+
+void Bela_floatHook(const char *source, float value){
+	// let's make this as optimized as possible for built-in digital Out parsing
+	// the built-in digital receivers are of the form "bela_digitalOutXX" where XX is between 11 and 26
+	static int prefixLength = 15; // strlen("bela_digitalOut")
+	if(strncmp(source, "bela_digitalOut", prefixLength)==0){
+		if(source[prefixLength] != 0){ //the two ifs are used instead of if(strlen(source) >= prefixLength+2)
+			if(source[prefixLength + 1] != 0){
+				// quickly convert the suffix to integer, assuming they are numbers, avoiding to call atoi
+				int receiver = ((source[prefixLength] - 48) * 10);
+				receiver += (source[prefixLength+1] - 48);
+				unsigned int channel = receiver - 11; // go back to the actual Bela digital channel number
+				if(channel >= 0 && channel < 16){ //16 is the hardcoded value for the number of digital channels
+					dcm.setValue(channel, value);
+				}
+			}
+		}
+	}
+}
+
+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 unsigned int analogChannelsInUse;
+static unsigned int gLibpdBlockSize;
+static unsigned int gChannelsInUse = 26;
+
+bool setup(BelaContext *context, void *userData)
+{
+	dcm.setCallback(sendDigitalMessage);
+	analogChannelsInUse = min(context->analogChannels, gChannelsInUse - context->audioChannels - context->digitalChannels);
+	if(context->digitalChannels > 0){
+		for(unsigned int ch = 0; ch < context->digitalChannels; ++ch){
+			dcm.setCallbackArgument(ch, receiverNames[ch]);
+		}
+	}
+	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_messagehook(Bela_messageHook);
+	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 out which in/outs to use
+	libpd_init_audio(gChannelsInUse, gChannelsInUse, context->audioSampleRate);
+	gInBuf = libpd_get_sys_soundin();
+	gOutBuf = libpd_get_sys_soundout();
+
+	libpd_start_message(1); // one entry in list
+	libpd_add_float(1.0f);
+	libpd_finish_message("pd", "dsp");
+
+	gBufLength = max(gLibpdBlockSize, context->audioFrames);
+
+
+	// bind your receivers here
+	libpd_bind("bela_digitalOut11");
+	libpd_bind("bela_digitalOut12");
+	libpd_bind("bela_digitalOut13");
+	libpd_bind("bela_digitalOut14");
+	libpd_bind("bela_digitalOut15");
+	libpd_bind("bela_digitalOut16");
+	libpd_bind("bela_digitalOut17");
+	libpd_bind("bela_digitalOut18");
+	libpd_bind("bela_digitalOut19");
+	libpd_bind("bela_digitalOut20");
+	libpd_bind("bela_digitalOut21");
+	libpd_bind("bela_digitalOut22");
+	libpd_bind("bela_digitalOut23");
+	libpd_bind("bela_digitalOut24");
+	libpd_bind("bela_digitalOut25");
+	libpd_bind("bela_digitalOut26");
+	libpd_bind("bela_setDigital");
+
+	char file[] = "_main.pd";
+	char folder[] = "./";
+	// open patch       [; pd open file folder(
+	libpd_openfile(file, folder);
+	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;
+
+	// these are reset at every audio callback. Persistence across audio callbacks
+	// is handled by the core code.
+//	setDataOut = 0;
+//	clearDataOut = 0;
+
+	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
+		// note: in multiple places below we assume that the number of digitals is same as number of audio
+		// digital in at message-rate
+		dcm.processInput(&context->digital[audioFrameBase], gLibpdBlockSize);
+
+		// digital in at signal-rate
+		for (j = 0, p0 = gInBuf; j < gLibpdBlockSize; j++, p0++) {
+			unsigned int digitalFrame = audioFrameBase + j;
+			for (k = 0, p1 = p0 + gLibpdBlockSize * (context->audioChannels + 8);
+					k < 16; ++k, p1 += gLibpdBlockSize) {
+				if(dcm.isSignalRate(k) && dcm.isInput(k)){ // only process input channels that are handled at signal rate
+					*p1 = digitalRead(context, digitalFrame, k);
+				}
+			}
+		}
+
+		libpd_process_sys(); // process the block
+
+		//digital out
+		// digital out at signal-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) {
+				if(dcm.isSignalRate(k) && dcm.isOutput(k)){ // only process output channels that are handled at signal rate
+					digitalWriteOnce(context, digitalFrame, k, *p1 > 0.5);
+				}
+			}
+		}
+
+		// digital out at message-rate
+		dcm.processOutput(&context->digital[audioFrameBase], gLibpdBlockSize);
+
+		//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)
+{
+}