void dsp_process_init(int cpu_hz, int hsb_hz, int pba_hz, int pbb_hz)
{
// Initialize TPA6130
tpa6130_init();
// Initialize DAC that send audio to TPA6130
tpa6130_dac_start(DAC_SAMPLING_RATE, DAC_NUM_CHANNELS,
DAC_BITS_PER_SAMPLE, DAC_SWAP_CHANNELS,
audio_callback, AUDIO_DAC_RELOAD_CB,
FOSC0);
tpa6130_set_volume(0x20);
tpa6130_get_volume();
signal_source_init(&signal1_generator, 433, 20000);
signal_source_init(&signal2_generator, 2000, 10000);
current_stereo_out_buf = stereo_out_buf1;
signal_in_buf = signal_pre_filter_buf + FIR_NUM_COEF;
filter_restore_default();
/* Fill the initial buffer for the hamming window with ones. This buffer
* will then be multiplied by the hamming window.
*/
dsp16_gen_step(fft_window, BUFFER_LENGTH, DSP16_Q(1.), DSP16_Q(1.), 0);
dsp16_win_hamm(fft_window, fft_window, BUFFER_LENGTH);
/* Run the interrupt handler manually once to start the ABDAC */
dac_reload_callback();
}
void dsp16_gen_dcomb(dsp16_t *vect1, int size, int f, int fs, dsp16_t delay)
{
int i;
int t_inc, t_low, cp;
int t_delay;
// Number of sample per period
t_inc = fs / f;
// Number of sample for the delay
t_delay = (((S32) delay) * ((S32) t_inc)) >> DSP16_QB;
// Number of sample per low signal
t_inc--;
i = 0;
t_low = t_delay;
// Main loop
while(i < size)
{
// Low part
cp = MIN(t_low, size);
for(; i<cp; i++)
vect1[i] = DSP16_Q(0.);
// High part
if (i < size)
vect1[i++] = DSP16_Q(1.);
t_low = t_inc + i;
}
}
//see header
rc_value_t delta_ramp(rc_value_t y1, rc_value_t y0, dsp16_t t)
{
if(t < DSP16_Q(0.5f))
{
//original formula is : y0 + (y1-y0)*t/(dT/2)
//it has been tweaked a little to avoid overflow
return 2*(y0/2 + 2*dsp16_op_mul((y1/2-y0/2),t) );
}
else
{
//y1 - (y1-y0)*(t-dT/2)/(dT/2)
return 2*(y1/2 - 2*dsp16_op_mul((y1/2-y0/2),(t-DSP16_Q(0.5f))) );
}
}
void dsp16_gen_rect(dsp16_t *vect1, int size, int f, int fs, dsp16_t duty, dsp16_t delay)
{
int i = 0;
int t_low_inc, t_high_inc, cp;
int t_low, t_high, t_delay;
// Number of sample per period
t_low_inc = fs / f;
// Number of sample per high signal
t_high_inc = (((S32) duty) * ((S32) t_low_inc)) >> DSP16_QB;
// Number of sample for the delay
t_delay = (((S32) delay) * ((S32) t_low_inc)) >> DSP16_QB;
// Number of sample per low signal
t_low_inc -= t_high_inc;
i = 0;
// For the delay
t_high = MIN(t_high_inc, t_delay - t_low_inc);
t_low = MIN(t_low_inc, t_delay);
if (t_high > 0)
t_low += t_high;
// Main loop
while(i < size)
{
// High part
cp = MIN(t_high, size);
for(; i<cp; i++)
vect1[i] = DSP16_Q(1.);
// Low part
cp = MIN(t_low, size);
for(; i<cp; i++)
vect1[i] = DSP16_Q(-1.);
t_high = t_high_inc + i;
t_low = t_high + t_low_inc;
}
}
dsp16_t dsp16_gen_saw(dsp16_t *vect1, int size, int f, int fs, dsp16_t duty, dsp16_t delay)
{
int i = 0;
S32 t_low_inc, t_high_inc, t;
dsp16_t delta_rise, delta_decrease, delta;
// Number of sample per period
t_low_inc = fs / f;
// Number of sample per high signal
t_high_inc = (((S32) duty) * ((S32) t_low_inc)) >> DSP16_QB;
// Number of sample for the delay
t = (((S32) delay) * ((S32) t_low_inc)) >> DSP16_QB;
// Number of sample per low signal
t_low_inc -= t_high_inc;
// Calculate the delta coefficient of the rising edge of the saw tooth
delta_rise = (DSP16_Q(1.) / t_high_inc) * 2;
// Calculate the delta coefficient of the decreasing edge of the saw tooth
delta_decrease = (DSP16_Q(1.) / t_low_inc) * 2;
// Compute the initial value
if (t < t_high_inc)
delta = DSP16_Q(-1.) + t * delta_rise;
else
delta = DSP16_Q(1.) - (t - t_high_inc) * delta_decrease;
while(i < size)
{
int lim, j;
if (i)
t = 0;
lim = Max(0, Min(t_high_inc - t, size - i));
for(j=0; j<lim; j++)
vect1[i++] = (delta += delta_rise);
t += lim;
if (j)
delta = DSP16_Q(1.);
lim = Min(t_high_inc + t_low_inc - t, size - i);
for(j=0; j<lim; j++)
vect1[i++] = (delta -= delta_decrease);
t += lim;
delta = DSP16_Q(-1.);
}
return (t << DSP16_QB) / (t_high_inc + t_low_inc);
}
//! The main function
int main(int argc, char *argv[])
{
unsigned int cycle_count;
int i;
short predicted_value, step_index;
int diff;
dsp16_t ratio;
// Switch to external Oscillator 0.
pm_switch_to_osc0(&AVR32_PM, FOSC0, OSC0_STARTUP);
// Initialize the DSP debug module
dsp_debug_initialization(FOSC0);
// Get the actual cycle count
cycle_count = Get_sys_count();
predicted_value = 0;
step_index = 0;
// Encode
dsp_adpcm_ima_encode(cbuf, sbuf, NSAMPLES, &step_index, &predicted_value);
predicted_value = 0;
step_index = 0;
// Decode
dsp_adpcm_ima_decode(new_sbuf, cbuf, NSAMPLES, &step_index, &predicted_value);
// Calculate the number of cycles
cycle_count = Get_sys_count() - cycle_count;
// Error calculation
diff = 0;
for(i=0; i<NSAMPLES; i++)
diff += ((new_sbuf[i]-sbuf[i]) > 0)?(new_sbuf[i]-sbuf[i]):(sbuf[i]-new_sbuf[i]);
ratio = DSP16_Q((diff/NSAMPLES)/(((float) (1 << 15))));
// Print the number of cycles
dsp16_debug_printf("Cycle count: %d\n", cycle_count);
// Print the error average
dsp16_debug_printf("Error average ratio: %f\n", ratio);
while(1);
}
/*
* Output range = [-1; 1] corresponding to [-pi; pi]
*/
dsp16_t dsp16_op_asin(dsp16_t num)
{
S32 num_sqr, res;
dsp16_t num_abs;
num_abs = dsp16_op_abs(num);
#if DSP16_QB < 15
// Limits
if (((S32) num) >= ((S32) DSP16_Q(1.)))
return DSP16_Q(0.5);
if (((S32) num) <= ((S32) DSP16_Q(-1.)))
return DSP16_Q(-0.5);
#endif
// If |num| > 0.5
if (num_abs > DSP16_Q(0.5))
{
num_abs = dsp16_op_sqrt((DSP16_Q(1.)-num_abs) >> 1);
num_abs = DSP16_Q(0.5) - (dsp16_op_asin(num_abs) << 1);
return (num < 0)?-num_abs:num_abs;
}
#include "compiler.h"
#include "board.h"
#include "dsp.h"
#include "dsp_debug.h"
#include "pm.h"
#include "cycle_counter.h"
#include "gpio.h"
//! \brief The size of the input signal
#define SIZE 64
//! \brief Buffer containing input data
A_ALIGNED dsp16_t input_x[SIZE] = {
DSP16_Q(0.00000000000),
DSP16_Q(0.03718075139),
DSP16_Q(0.07131184859),
DSP16_Q(0.09963983090),
DSP16_Q(0.11997464035),
DSP16_Q(0.13090169944),
DSP16_Q(0.13191810690),
DSP16_Q(0.12347962859),
DSP16_Q(0.10695389264),
DSP16_Q(0.08448437894),
DSP16_Q(0.05877852523),
DSP16_Q(0.03284069954),
DSP16_Q(0.00967618536),
DSP16_Q(-0.00800483670),
DSP16_Q(-0.01805432735),
DSP16_Q(-0.01909830056),
//! The size of the input signal
#define SIZE 48
// A High-pass IIR Filter
// Butterworth model
// Fsample = 48 KHz
// Fcut-off = 2000 Hz
// Order 5
#define NUM_SIZE 6
#define NUM_PREDIV 3
#define DEN_SIZE 5
#define DEN_PREDIV 3
A_ALIGNED dsp16_t num[NUM_SIZE] = {
DSP16_Q(0.6537018 / (1 << NUM_PREDIV)),
DSP16_Q(-3.2685088 / (1 << NUM_PREDIV)),
DSP16_Q(6.5370176 / (1 << NUM_PREDIV)),
DSP16_Q(-6.5370176 / (1 << NUM_PREDIV)),
DSP16_Q(3.2685088 / (1 << NUM_PREDIV)),
DSP16_Q(-0.6537018 / (1 << NUM_PREDIV))
};
A_ALIGNED dsp16_t den[DEN_SIZE] = {
DSP16_Q(-4.1534907 / (1 << DEN_PREDIV)),
DSP16_Q(6.9612765 / (1 << DEN_PREDIV)),
DSP16_Q(-5.877997 / (1 << DEN_PREDIV)),
DSP16_Q(2.498365 / (1 << DEN_PREDIV)),
DSP16_Q(-0.427326 / (1 << DEN_PREDIV))
};
#if defined(FORCE_ALL_GENERICS) || \
defined(FORCE_GENERIC_OP16_SIN) || \
!defined(TARGET_SPECIFIC_OP16_SIN)
#define DSP16_MODULO_1_MASK (((U16) -1) >> (16 - (DSP16_QB+1)))
extern dsp16_t dsp16_op_kernel_cosfix(dsp16_t angle);
extern dsp16_t dsp16_op_kernel_sinfix(dsp16_t angle);
/*
* Input range = [-1; 1] corresponding to [-pi; pi]
*/
dsp16_t dsp16_op_sin(dsp16_t angle)
{
#if DSP16_QA > 1
U16 n = ((angle + DSP16_Q(0.25)) & DSP16_MODULO_1_MASK) >> (DSP16_QB-1);
#else
U16 n = ((U16) (angle + DSP16_Q(0.25))) >> (DSP16_QB-1);
#endif
// Translate input down to +/- pi/4
angle -= n*DSP16_Q(0.5);
#if DSP16_QA > 1
// angle modulo 1 (signed values)
angle <<= (16 - (DSP16_QB+1));
angle >>= (16 - (DSP16_QB+1));
#endif
switch(n)
{
static bool state_machine_source(int source_id, enum state_function *state)
{
static dsp16_t volume;
static unsigned int frequency;
struct signal_source *source = NULL;
if (source_id == 1)
source = &signal1_generator;
else if (source_id == 2)
source = &signal2_generator;
switch (*state)
{
case STATE_FCT_IDLE:
if (source_id == 1)
{
if (new_state_fct)
{
gui_set_selection(GUI_SOURCE1_ID);
gui_text_print(GUI_COMMENT_ID, TEXT_SOURCE1);
}
else
{
if (controller_wheel_right(2))
switch_state(STATE_SOURCE2);
return false;
}
}
else if (source_id == 2)
{
if (new_state_fct)
{
gui_set_selection(GUI_SOURCE2_ID);
gui_text_print(GUI_COMMENT_ID, TEXT_SOURCE2);
}
else
{
if (controller_wheel_left(2))
switch_state(STATE_SOURCE1);
else if (controller_wheel_right(2))
switch_state(STATE_INPUT);
return false;
}
}
break;
// Amplitude
case STATE_FCT_FUNCTION1:
volume = signal_source_get_volume(source);
if (controller_wheel_right(1) && volume < DSP16_Q(1.))
{
if (volume < DSP16_Q(1.) - DSP16_Q(1./16))
volume += DSP16_Q(1./16);
else
volume = DSP16_Q(1.);
new_state_fct = true;
}
else if (controller_wheel_left(1))
{
if (volume > DSP16_Q(1./16))
volume -= DSP16_Q(1./16);
else
volume = 0;
new_state_fct = true;
}
if (new_state_fct)
{
signal_source_set_volume(source, volume);
gui_text_printf(GUI_COMMENT_ID, "Source%i - " TEXT_FUNC1 "\nAmplitude %f\n\n\n\n" TEXT_WHEEL, source_id, volume);
}
break;
// Frequency
case STATE_FCT_FUNCTION2:
frequency = signal_source_get_freq(source);
if (controller_wheel_right(1) && frequency < 10000)
{
frequency *= 1.1;
new_state_fct = true;
}
else if (controller_wheel_left(1) && frequency > 100)
{
frequency *= 0.9;
new_state_fct = true;
}
if (new_state_fct)
{
signal_source_set_freq(source, frequency);
gui_text_printf(GUI_COMMENT_ID, "Source%i - " TEXT_FUNC2 "\nFrequency %iHz\n\n\n\n" TEXT_WHEEL, source_id, frequency);
}
break;
case STATE_FCT_FUNCTION3:
break;
// Zoom
case STATE_FCT_ZOOM:
if (new_state_fct)
{
zoom_view = true;
if (source_id == 1)
zoom_view_id = GUI_SOURCE1_ID;
else if (source_id == 2)
zoom_view_id = GUI_SOURCE2_ID;
controller_reset();
}
break;
}
return true;
}
#include "pm.h"
#include "cycle_counter.h"
// Length of the output signal
#define VECT1_SIZE 11
// Length of the first input signal
#define VECT2_SIZE 11
// Length of the second input signal
#define VECT3_SIZE 11
// The output signal
A_ALIGNED dsp16_t vect1[VECT1_SIZE];
// The first input signal
A_ALIGNED dsp16_t vect2[VECT2_SIZE] = {
DSP16_Q(0.012010),
DSP16_Q(0.059717),
DSP16_Q(0.101397),
DSP16_Q(0.0583150),
DSP16_Q(0.0000000),
DSP16_Q(0.0921177),
DSP16_Q(0.0951057),
DSP16_Q(0.0884270),
DSP16_Q(0.0716020),
DSP16_Q(0.515080),
DSP16_Q(0.0583150)
};
// The second input signal
A_ALIGNED dsp16_t vect3[VECT3_SIZE] = {
DSP16_Q(0.5),
DSP16_Q(0.101397),
DSP32_Q(-0.01913417162), DSP32_Q(-0.05000000000), DSP32_Q(-0.07373934164), DSP32_Q(-0.08535533906), DSP32_Q(-0.08296893720), DSP32_Q(-0.06830127019), DSP32_Q(-0.04619397663), DSP32_Q(-0.02329629131), DSP32_Q(-0.00627097288), DSP32_Q(0.00000000000), DSP32_Q(-0.00627097288), DSP32_Q(-0.02329629131), DSP32_Q(-0.04619397663), DSP32_Q(-0.06830127019), DSP32_Q(-0.08296893720), DSP16_Q(-0.08535533906), DSP32_Q(-0.07373934164), DSP32_Q(-0.05000000000), DSP32_Q(-0.01913417162), DSP32_Q(0.01205904774), DSP32_Q(0.03677496058), DSP32_Q(0.05000000000), DSP32_Q(0.04982757980), DSP32_Q(0.03794095226), DSP32_Q(0.01913417162), DSP32_Q(0.00000000000), DSP32_Q(-0.01286319874), DSP32_Q(-0.01464466094), DSP32_Q(-0.00363360317), DSP32_Q(0.01830127019), DSP32_Q(0.04619397663),
void dac_overrun_callback(void);
void dac_reload_callback(void);
static A_ALIGNED dsp16_t fft_window[BUFFER_LENGTH];
unsigned int active_filter;
const char * filter_description[NUM_FILTERS] = {
"No filtering",
"High-pass filter,\n1000Hz cut-off",
"Low-pass filter,\n1000Hz cut-off",
};
A_ALIGNED dsp16_t filter_coef[NUM_FILTERS - 1][FIR_NUM_COEF] = {
{
DSP16_Q(-0.0184183),
DSP16_Q(-0.0207874),
DSP16_Q(-0.0231255),
DSP16_Q(-0.0254103),
DSP16_Q(-0.0276199),
DSP16_Q(-0.0297327),
DSP16_Q(-0.0317281),
DSP16_Q(-0.0335863),
DSP16_Q(-0.0352887),
DSP16_Q(-0.0368182),
DSP16_Q(-0.0381593),
DSP16_Q(-0.0392986),
DSP16_Q(-0.0402242),
DSP16_Q(-0.0409269),
DSP16_Q(-0.0413995),
DSP16_Q(0.9583631),
DSP16_Q(0.261138916015625),
DSP16_Q(0.3179931640625),
DSP16_Q(0.261138916015625),
DSP16_Q(0.127838134765625),
DSP16_Q(0),
DSP16_Q(-0.05859375),
DSP16_Q(-0.043792724609375),
DSP16_Q(0),
DSP16_Q(0.025848388671875),
DSP16_Q(0.0198974609375)
};
*/
// calculated by Torsten Schultze DG1HT
A_ALIGNED const dsp16_t RXLowPassCoeff[] = {
DSP16_Q(0.019823264989714),
DSP16_Q(0.025848388671875),
DSP16_Q(0),
DSP16_Q(-0.043792724609375),
DSP16_Q(-0.05859375),
DSP16_Q(0),
DSP16_Q(0.127838134765625),
DSP16_Q(0.261138916015625),
DSP16_Q(0.3179931640625),
DSP16_Q(0.261138916015625),
DSP16_Q(0.127838134765625),
DSP16_Q(0),
DSP16_Q(-0.058593752349941),
DSP16_Q(-0.041093832349941),
DSP16_Q(0),
DSP16_Q(0.026812695985862),
#include "dsp.h"
#include "dsp_debug.h"
#include "pm.h"
#include "cycle_counter.h"
//! The size of the input signal
#define SIZE 72
//! The output buffer
A_ALIGNED dsp16_t vect1[SIZE];
//! The input signal
A_ALIGNED dsp16_t vect2[SIZE] = {
DSP16_Q(0.10000000000),
DSP16_Q(-0.06909830056),
DSP16_Q(0.15877852523),
DSP16_Q(-0.01909830056),
DSP16_Q(0.19510565163),
DSP16_Q(0.00000000000),
DSP16_Q(0.19510565163),
DSP16_Q(-0.01909830056),
DSP16_Q(0.15877852523),
DSP16_Q(-0.06909830056),
DSP16_Q(0.10000000000),
DSP16_Q(-0.13090169944),
DSP16_Q(0.04122147477),
DSP16_Q(-0.18090169944),
DSP16_Q(0.00489434837),
DSP16_Q(-0.20000000000),
#include "pm.h"
#include "cycle_counter.h"
//! Sampling rate of the original signal
#define F_INPUT 22050
//! Sampling rate of the output signal
#define F_OUTPUT 48510
//! Number of point to process
#define N 200
#define NB_CUTS 40
//! Filter order
#define FILTER_ORDER 6
A_ALIGNED dsp16_t s[N] = {
DSP16_Q(0.00000000000),
DSP16_Q(0.28111111333),
DSP16_Q(0.53955074319),
DSP16_Q(0.75447585092),
DSP16_Q(0.90855282432),
DSP16_Q(0.98935542552),
DSP16_Q(0.99036696149),
DSP16_Q(0.91150585231),
DSP16_Q(0.75913221057),
DSP16_Q(0.54553490121),
DSP16_Q(0.28794045010),
DSP16_Q(0.00712373261),
DSP16_Q(-0.27426751067),
DSP16_Q(-0.53353920393),
DSP16_Q(-0.74978120297),
DSP16_Q(-0.90555368889),
#include "dsp_debug.h"
#include "pm.h"
#include "cycle_counter.h"
//! The size of the input signal
#define SIZE 64
//! The number of tap
#define FIR_COEF_SIZE 25
//! The output buffer
A_ALIGNED dsp16_t y[SIZE - FIR_COEF_SIZE + 1 + 4]; // <- "+4" for avr32-uc3 optimized FIR version
//! The input signal
A_ALIGNED dsp16_t x[SIZE] = {
DSP16_Q(0.00000000000),
DSP16_Q(0.05971733276),
DSP16_Q(0.10139671591),
DSP16_Q(0.11013997327),
DSP16_Q(0.07684940146),
DSP16_Q(0.00000000000),
DSP16_Q(-0.11375674283),
DSP16_Q(-0.25026678831),
DSP16_Q(-0.38974690945),
DSP16_Q(-0.50960156497),
DSP16_Q(-0.58778525229),
DSP16_Q(-0.60622623909),
DSP16_Q(-0.55381024215),
DSP16_Q(-0.42847700858),
DSP16_Q(-0.23810168641),
DSP16_Q(-0.00000000000),
