comparison projects/basic_libpd/render.cpp @ 230:af211ee57867 mergingClockSync

Working basic_libpd

+/*
+ * render.cpp
+ *
+ *  Created on: Oct 24, 2014
+ *      Author: parallels
+ */
+
+#include <BeagleRT.h>
+#include <cmath>
+#include <Utilities.h>
+#include <I2c_Codec.h>
+#include <PRU.h>
+#include <stdio.h>
+#include "z_libpd.h"
+#include <UdpServer.h>
+
+// setup() 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.
+#define DEFDACBLKSIZE 8u  //make sure this matches the one used to compile libpd
+
+const int gChannelsInUse = 6;
+int gBufLength;
+
+float* gInBuf;
+float* gOutBuf;
+
+void pdnoteon(int ch, int pitch, int vel) {
+  printf("noteon: %d %d %d\n", ch, pitch, vel);
+}
+
+void BeagleRT_printHook(const char *recv){
+	rt_printf("%s", recv);
+}
+
+UdpServer udpServer;
+
+void udpRead(){
+    char dest[100] = {0};
+    while(!gShouldStop){
+    	libpd_sys_microsleep(0);
+        usleep(1000);
+    }
+}
+
+AuxiliaryTask  udpReadTask;
+bool setup(BeagleRTContext *context, void *userData)
+{
+	udpServer.bindToPort(1234);
+
+	// check that we are not running with a blocksize smaller than DEFDACBLKSIZE
+	// it would still work, but the load would be executed unevenly between calls to render
+	if(context->audioFrames < DEFDACBLKSIZE){
+		fprintf(stderr, "Error: minimum block size must be %d\n", DEFDACBLKSIZE);
+		return false;
+	}
+
+	// check that the sampling rate of the analogs is the same as audio if running with
+	// more than 2 channels (that is with analog). If we fix the TODO in render, then
+	// this test is not needed.
+	if(context->analogFrames != context->audioFrames){
+		fprintf(stderr, "Error: analog and audio sampling rates must be the same\n");
+		return false;
+	}
+	//following lines borrowed from libpd/samples/c/pdtest/pdtest.c
+	// init pd
+	libpd_set_printhook(BeagleRT_printHook); // set this before calling libpd_init
+	libpd_set_noteonhook(pdnoteon);
+	libpd_init();
+	//TODO: 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(DEFDACBLKSIZE, context->audioFrames);
+	unsigned int bufferSize = sizeof(float)*gChannelsInUse*gBufLength;
+	gInBuf = (float*)malloc(bufferSize);
+	gOutBuf = (float*)malloc(bufferSize);
+	// no need to memset to zero
+
+
+	char file[] = "_main.pd";
+	char folder[] = "./";
+	// open patch       [; pd open file folder(
+	libpd_openfile(file, folder);
+    
+	udpReadTask = BeagleRT_createAuxiliaryTask(udpRead, 60, "udpReadTask");
+	BeagleRT_scheduleAuxiliaryTask(udpReadTask);
+	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.
+BeagleRTContext *c;
+void render(BeagleRTContext *context, void *userData)
+{
+	static int inW = 0;
+	static int outR = 0;
+/*
+ *	NOTE: if you are only using audio (or only analogs) and you are using interleaved buffers
+ *	 and the blocksize of Bela is the same as DEFDACBLKSIZE, then you probably
+ *	 do not need the for loops before and after libpd_process_float, so you can save quite some
+ *	 memory operations.
+ */
+
+	for(unsigned int n = 0; n < context->audioFrames; n++){ //pd buffers are interleaved
+		for(unsigned int k = 0; k < context->audioChannels; k++){
+			gInBuf[inW++] = audioReadFrame(context, n, k);
+		}
+		for(unsigned int k = 0; k < gChannelsInUse - context->audioChannels; k ++){ // add analogs
+			gInBuf[inW++] = analogReadFrame(context, n, k);
+			// TODO: Apply here sampling rate conversion from analogs to audio
+		}
+		if(inW == gBufLength * gChannelsInUse){
+			inW = 0;
+		}
+	}
+
+	if(inW == 0){ //if the buffer is full, process it
+		int numberOfPdBlocksToProcess = gBufLength/DEFDACBLKSIZE;
+		libpd_process_float(numberOfPdBlocksToProcess, gInBuf, gOutBuf);
+		outR = 0; // reset the read pointer. NOTE: hopefully this is not needed EXCEPT the first time
+	}
+
+	for(unsigned int n = 0; n < context->audioFrames; n++){ //pd buffers are interleaved
+		for(unsigned int k = 0; k < context->audioChannels; k++){
+			audioWriteFrame(context, n, k, gOutBuf[outR++]);
+		}
+		//add analogs here, limit them to channelsInUse
+		for(unsigned int k = 0; k < gChannelsInUse - context->audioChannels; k ++){ // add analogs
+			analogWriteFrame(context, n, k, gOutBuf[outR++]);
+			// TODO: Apply here sampling rate conversion from analogs to audio
+		}
+		if(outR == gBufLength * gChannelsInUse){
+			outR = 0;
+		}
+	}
+}
+// cleanup() is called once at the end, after the audio has stopped.
+// Release any resources that were allocated in setup().
+
+void cleanup(BeagleRTContext *context, void *userData)
+{
+	free(gInBuf);
+	free(gOutBuf);
+}