changeset 3:6810f166482f

_new IIR filter example

--- a/.cproject	Thu Nov 06 15:59:16 2014 +0000
+++ b/.cproject	Thu Nov 06 17:55:05 2014 +0000
@@ -83,8 +83,8 @@
 					</folderInfo>
 					<sourceEntries>
 						<entry flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="include"/>
-						<entry excluding="oscillator_bank|samples|basic_sensor|d-box|basic_analog_output|basic" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="projects"/>
 						<entry excluding="audio_routines_old.S" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="core"/>
+						<entry excluding="filter_FIR|oscillator_bank|samples|basic_sensor|d-box|basic_analog_output|basic" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="projects"/>
 					</sourceEntries>
 				</configuration>
 			</storageModule>
@@ -155,9 +155,9 @@
 						</toolChain>
 					</folderInfo>
 					<sourceEntries>
-						<entry excluding="oscillator_bank|samples|basic_sensor|d-box|basic_analog_output|basic" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="projects"/>
 						<entry flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="include"/>
 						<entry excluding="audio_routines_old.S" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="core"/>
+						<entry excluding="filter_FIR|oscillator_bank|samples|basic_sensor|d-box|basic_analog_output|basic" flags="VALUE_WORKSPACE_PATH|RESOLVED" kind="sourcePath" name="projects"/>
 					</sourceEntries>
 				</configuration>
 			</storageModule>

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/projects/filter_IIR/SampleData.h	Thu Nov 06 17:55:05 2014 +0000
@@ -0,0 +1,19 @@
+/*
+ * SampleData.h
+ *
+ *  Created on: Nov 5, 2014
+ *      Author: Victor Zappi
+ */
+
+#ifndef SAMPLEDATA_H_
+#define SAMPLEDATA_H_
+
+// User defined structure to pass between main and rendere complex data retrieved from file
+struct SampleData {
+	float *samples;	// Samples in file
+	int sampleLen;	// Total nume of samples
+};
+
+
+
+#endif /* SAMPLEDATA_H_ */

Binary file projects/filter_IIR/longsample.wav has changed

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/projects/filter_IIR/main.cpp	Thu Nov 06 17:55:05 2014 +0000
@@ -0,0 +1,195 @@
+/*
+ * main.cpp
+ *
+ *  Created on: Oct 24, 2014
+ *      Author: Andrew McPherson and Victor Zappi
+ */
+
+#include <iostream>
+#include <cstdlib>
+#include <libgen.h>
+#include <signal.h>
+#include <string>
+#include <sndfile.h>				// to load audio files
+#include "../../include/RTAudio.h"
+#include "SampleData.h"
+
+using namespace std;
+
+float gCutFreq = 100;
+
+// Load samples from file
+int initFile(string file, SampleData *smp)//float *& smp)
+{
+	SNDFILE *sndfile ;
+	SF_INFO sfinfo ;
+
+	if (!(sndfile = sf_open (file.c_str(), SFM_READ, &sfinfo))) {
+		cout << "Couldn't open file " << file << endl;
+		return 1;
+	}
+
+	int numChan = sfinfo.channels;
+	if(numChan != 1)
+	{
+		cout << "Error: " << file << " is not a mono file" << endl;
+		return 1;
+	}
+
+	smp->sampleLen = sfinfo.frames * numChan;
+	smp->samples = new float[smp->sampleLen];
+	if(smp == NULL){
+		cout << "Could not allocate buffer" << endl;
+		return 1;
+	}
+
+	int subformat = sfinfo.format & SF_FORMAT_SUBMASK;
+	int readcount = sf_read_float(sndfile, smp->samples, smp->sampleLen);
+
+	// Pad with zeros in case we couldn't read whole file
+	for(int k = readcount; k <smp->sampleLen; k++)
+		smp->samples[k] = 0;
+
+	if (subformat == SF_FORMAT_FLOAT || subformat == SF_FORMAT_DOUBLE) {
+		double	scale ;
+		int 	m ;
+
+		sf_command (sndfile, SFC_CALC_SIGNAL_MAX, &scale, sizeof (scale)) ;
+		if (scale < 1e-10)
+			scale = 1.0 ;
+		else
+			scale = 32700.0 / scale ;
+		cout << "File samples scale = " << scale << endl;
+
+		for (m = 0; m < smp->sampleLen; m++)
+			smp->samples[m] *= scale;
+	}
+
+	sf_close(sndfile);
+
+	return 0;
+}
+
+
+// Handle Ctrl-C by requesting that the audio rendering stop
+void interrupt_handler(int var)
+{
+	//rt_task_delete ((RT_TASK *) &gTriggerSamplesTask);
+	gShouldStop = true;
+}
+
+// Print usage information
+void usage(const char * processName)
+{
+	cerr << "Usage: " << processName << " [-h] [-v] [-p period] [-f frequency]" << endl;
+	cerr << "   -h:           Print this menu\n";
+	cerr << "   -v:           Enable verbose messages\n";
+	cerr << "   -p period:    Set the period (hardware buffer) size in sensor frames\n";
+	cerr << "   -m:           Enable the matrix (ADC and DAC) as well as audio\n";
+	cerr << "   -f filename:  Name of the file to load (default is \"sample.wav\")\n";
+	cerr << "   -c freq:      Set the cut off frequency of the filter in Hz\n";
+}
+
+int main(int argc, char *argv[])
+{
+	int verbose = 0;			// Verbose printing level
+	int periodSize = 8;			// Period size in sensor frames
+	int useMatrix = 0;			// Whether to use the matrix or just audio
+	string fileName;			// Name of the sample to load
+
+	SampleData sampleData;		// User define structure to pass data retrieved from file to render function
+	sampleData.samples = 0;
+	sampleData.sampleLen = -1;
+
+	// Parse command-line arguments
+	while (1) {
+		int c;
+		if ((c = getopt(argc, argv, "hp:vms:")) < 0)
+				break;
+		switch (c) {
+		case 'h':
+				usage(basename(argv[0]));
+				exit(0);
+		case 'p':
+				periodSize = atoi(optarg);
+				if(periodSize < 1)
+					periodSize = 1;
+				break;
+		case 'v':
+				verbose = 1;
+				break;
+		case 'm':
+				useMatrix = 1;
+				break;
+		case 'f':
+				fileName = string((char *)optarg);
+				break;
+		case 'c':
+				gCutFreq = atof(optarg);
+				break;
+		case '?':
+		default:
+				usage(basename(argv[0]));
+				exit(1);
+		}
+	}
+
+	if(fileName.empty()){
+		fileName = "filter/longsample.wav";
+	}
+
+	// Set verbose logging information (optional by using value > 0; default is 0)
+	setVerboseLevel(verbose);
+
+	if(verbose) {
+		cout << "Starting with period size " << periodSize << endl;
+		if(useMatrix)
+			cout << "Matrix enabled\n";
+		else
+			cout << "Matrix disabled\n";
+		cout << "Loading file " << fileName << endl;
+	}
+
+	// Load file
+	if(initFile(fileName, &sampleData) != 0)
+	{
+		cout << "Error: unable to load samples " << endl;
+		return -1;
+	}
+
+	if(verbose)
+		cout << "File contains " << sampleData.sampleLen << " samples" << endl;
+
+	// Initialise the PRU audio device
+	if(initAudio(periodSize, useMatrix, &sampleData) != 0) {
+		cout << "Error: unable to initialise audio" << endl;
+		return -1;
+	}
+
+	// Start the audio device running
+	if(startAudio()) {
+		cout << "Error: unable to start real-time audio" << endl;
+		return -1;
+	}
+
+	// Set up interrupt handler to catch Control-C
+	signal(SIGINT, interrupt_handler);
+
+	// Run until told to stop
+	while(!gShouldStop) {
+		usleep(100000);
+	}
+
+	// Stop the audio device
+	stopAudio();
+
+	if(verbose) {
+		cout << "Cleaning up..." << endl;
+	}
+
+	// Clean up any resources allocated for audio
+	cleanupAudio();
+
+	// All done!
+	return 0;
+}

--- /dev/null	Thu Jan 01 00:00:00 1970 +0000
+++ b/projects/filter_IIR/render.cpp	Thu Nov 06 17:55:05 2014 +0000
@@ -0,0 +1,227 @@
+/*
+ * render.cpp
+ *
+ *  Created on: Oct 24, 2014
+ *      Author: Andrew McPherson and Victor Zappi
+ */
+
+
+#include "../../include/render.h"
+#include "../../include/RTAudio.h"	// to schedule lower prio parallel process
+#include <rtdk.h>
+#include <cmath>
+#include <stdio.h>
+#include "SampleData.h"
+
+SampleData gSampleData;	// User defined structure to get complex data from main
+int gReadPtr;			// Position of last read sample from file
+int gNumChannels;
+
+// filter vars
+float gLastX[2];
+float gLastY[2];
+double lb0, lb1, lb2, la1, la2 = 0.0;
+
+// communication vars between the 2 auxiliary tasks
+int gChangeCoeff = 0;
+int gFreqDelta = 0;
+
+void initialise_filter(float freq);
+
+void calculate_coeff(float cutFreq);
+
+bool initialise_aux_tasks();
+
+// Task for handling the update of the frequencies using the matrix
+AuxiliaryTask gChangeCoeffTask;
+
+void check_coeff();
+
+// Task for handling the update of the frequencies using the matrix
+AuxiliaryTask gInputTask;
+
+void read_input();
+
+
+extern float gCutFreq;
+
+
+// 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 numChannels, int numMatrixFramesPerPeriod,
+					   int numAudioFramesPerPeriod, float matrixSampleRate,
+					   float audioSampleRate, void *userData)
+{
+
+	// Retrieve a parameter passed in from the initAudio() call
+	gSampleData = *(SampleData *)userData;
+
+	gReadPtr = -1;
+	gNumChannels = numChannels;
+
+	initialise_filter(200);
+
+	// Initialise auxiliary tasks
+	if(!initialise_aux_tasks())
+		return false;
+
+	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 numMatrixFrames, int numAudioFrames, float *audioIn, float *audioOut,
+			uint16_t *matrixIn, uint16_t *matrixOut)
+{
+	for(int n = 0; n < numAudioFrames; n++) {
+		float sample = 0;
+		float out = 0;
+
+		// If triggered...
+		if(gReadPtr != -1)
+			sample += gSampleData.samples[gReadPtr++];	// ...read each sample...
+
+		if(gReadPtr >= gSampleData.sampleLen)
+			gReadPtr = -1;
+
+		out = lb0*sample+lb1*gLastX[0]+lb2*gLastX[1]-la1*gLastY[0]-la2*gLastY[1];
+
+		gLastX[1] = gLastX[0];
+		gLastX[0] = out;
+		gLastY[1] = gLastY[0];
+		gLastY[0] = out;
+
+		for(int channel = 0; channel < gNumChannels; channel++)
+			audioOut[n * gNumChannels + channel] = out;	// ...and put it in both left and right channel
+
+	}
+
+	// Request that the lower-priority tasks run at next opportunity
+	scheduleAuxiliaryTask(gChangeCoeffTask);
+	scheduleAuxiliaryTask(gInputTask);
+}
+
+// First calculation of coefficients
+
+void initialise_filter(float freq)
+{
+	calculate_coeff(freq);
+}
+
+
+// Calculate the filter coefficients
+// second order low pass butterworth filter
+
+void calculate_coeff(float cutFreq)
+{
+	// Initialise any previous state (clearing buffers etc.)
+	// to prepare for calls to render()
+	float sampleRate = 44100;
+	double f = 2*M_PI*cutFreq/sampleRate;
+	double denom = 4+2*sqrt(2)*f+f*f;
+	lb0 = f*f/denom;
+	lb1 = 2*lb0;
+	lb2 = lb0;
+	la1 = (2*f*f-8)/denom;
+	la2 = (f*f+4-2*sqrt(2)*f)/denom;
+	gLastX[0] = gLastX [1] = 0;
+	gLastY[0] = gLastY[1] = 0;
+
+}
+
+
+// Initialise the auxiliary tasks
+// and print info
+
+bool initialise_aux_tasks()
+{
+	if((gChangeCoeffTask = createAuxiliaryTaskLoop(&check_coeff, 90, "beaglert-check-coeff")) == 0)
+		return false;
+
+	if((gInputTask = createAuxiliaryTaskLoop(&read_input, 50, "beaglert-read-input")) == 0)
+		return false;
+
+	rt_printf("Press 'a' to trigger sample, 's' to stop\n");
+	rt_printf("Press 'z' to low down cut-off freq of 100 Hz, 'x' to raise it up\n");
+	rt_printf("Press 'q' to quit\n");
+
+	return true;
+}
+
+// Check if cut-off freq has been changed
+// and new coefficients are needed
+
+void check_coeff()
+{
+	if(gChangeCoeff == 1)
+	{
+		gCutFreq += gFreqDelta;
+		gCutFreq = gCutFreq < 0 ? 0 : gCutFreq;
+		gCutFreq = gCutFreq > 22050 ? 22050 : gCutFreq;
+
+		rt_printf("Cut-off frequency: %f\n", gCutFreq);
+
+		calculate_coeff(gCutFreq);
+		gChangeCoeff = 0;
+	}
+}
+
+// This is a lower-priority call to periodically read keyboard input
+// and trigger samples. By placing it at a lower priority,
+// it has minimal effect on the audio performance but it will take longer to
+// complete if the system is under heavy audio load.
+
+void read_input()
+{
+	// This is not a real-time task!
+	// Cos getchar is a system call, not handled by Xenomai.
+	// This task will be automatically down graded.
+
+	char keyStroke = '.';
+
+	keyStroke =	getchar();
+	while(getchar()!='\n'); // to read the first stroke
+
+	switch (keyStroke)
+	{
+		case 'a':
+			gReadPtr = 0;
+			break;
+		case 's':
+			gReadPtr = -1;
+			break;
+		case 'z':
+			gChangeCoeff = 1;
+			gFreqDelta = -100;
+			break;
+		case 'x':
+			gChangeCoeff = 1;
+			gFreqDelta = 100;
+			break;
+		case 'q':
+			gShouldStop = true;
+			break;
+		default:
+			break;
+	}
+}
+
+
+
+// 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()
+{
+	delete[] gSampleData.samples;
+}