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Move samples of int16_t format to sample_t, that is of type double

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This prepares the correction of all levels
This commit is contained in:
Andreas Eversberg
2017-01-27 16:57:34 +01:00
parent 538a959128
commit 7ea3bc188d
74 changed files with 471 additions and 447 deletions
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83
src/nmt/dsp.c
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@@ -25,9 +25,9 @@
#include <string.h>
#include <errno.h>
#include <math.h>
#include "../common/sample.h"
#include "../common/debug.h"
#include "../common/timer.h"
#include "../common/goertzel.h"
#include "nmt.h"
#include "transaction.h"
#include "dsp.h"
@@ -50,7 +50,7 @@
#define BIT_RATE 1200 /* baud rate */
#define STEPS_PER_BIT 10 /* step every 1/12000 sec */
#define DIALTONE_HZ 425.0 /* dial tone frequency */
#define TX_PEAK_DIALTONE 16000 /* dial tone peak */
#define TX_PEAK_DIALTONE 16000.0 /* dial tone peak */
#define SUPER_DURATION 0.25 /* duration of supervisory signal measurement */
#define SUPER_DETECT_COUNT 4 /* number of measures to detect supervisory signal */
#define MUTE_DURATION 0.280 /* a tiny bit more than two frames */
@@ -71,9 +71,9 @@ static double super_freq[5] = {
};
/* table for fast sine generation */
uint16_t dsp_tone_bit[2][2][256]; /* polarity, bit, phase */
uint16_t dsp_sine_super[256];
uint16_t dsp_sine_dialtone[256];
static double dsp_tone_bit[2][2][256]; /* polarity, bit, phase */
static double dsp_sine_super[256];
static double dsp_sine_dialtone[256];
/* global init for FSK */
void dsp_init(void)
@@ -85,24 +85,23 @@ void dsp_init(void)
for (i = 0; i < 256; i++) {
s = sin((double)i / 256.0 * 2.0 * PI);
/* supervisor sine */
dsp_sine_super[i] = (int)(s * TX_PEAK_SUPER);
dsp_sine_super[i] = s * TX_PEAK_SUPER;
/* dialtone sine */
dsp_sine_dialtone[i] = (int)(s * TX_PEAK_DIALTONE);
dsp_sine_dialtone[i] = s * TX_PEAK_DIALTONE;
/* bit(1) 1 cycle */
dsp_tone_bit[0][1][i] = (int)(s * TX_PEAK_FSK);
dsp_tone_bit[1][1][i] = (int)(-s * TX_PEAK_FSK);
dsp_tone_bit[0][1][i] = s * TX_PEAK_FSK;
dsp_tone_bit[1][1][i] = -s * TX_PEAK_FSK;
/* bit(0) 1.5 cycles */
s = sin((double)i / 256.0 * 3.0 * PI);
dsp_tone_bit[0][0][i] = (int)(s * TX_PEAK_FSK);
dsp_tone_bit[1][0][i] = (int)(-s * TX_PEAK_FSK);
dsp_tone_bit[0][0][i] = s * TX_PEAK_FSK;
dsp_tone_bit[1][0][i] = -s * TX_PEAK_FSK;
}
}
/* Init FSK of transceiver */
int dsp_init_sender(nmt_t *nmt)
{
double coeff;
int16_t *spl;
sample_t *spl;
int i;
/* attack (3ms) and recovery time (13.5ms) according to NMT specs */
@@ -165,20 +164,14 @@ int dsp_init_sender(nmt_t *nmt)
nmt->super_filter_spl = spl;
/* count symbols */
for (i = 0; i < 2; i++) {
coeff = 2.0 * cos(2.0 * PI * fsk_freq[i] / (double)nmt->sender.samplerate);
nmt->fsk_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_coeff[%d] = %d\n", i, (int)nmt->fsk_coeff[i]);
}
for (i = 0; i < 2; i++)
audio_goertzel_init(&nmt->fsk_goertzel[i], fsk_freq[i], nmt->sender.samplerate);
nmt->fsk_phaseshift256 = 256.0 / nmt->fsk_samples_per_bit;
PDEBUG(DDSP, DEBUG_DEBUG, "fsk_phaseshift = %.4f\n", nmt->fsk_phaseshift256);
/* count supervidory tones */
for (i = 0; i < 5; i++) {
coeff = 2.0 * cos(2.0 * PI * super_freq[i] / (double)nmt->sender.samplerate);
nmt->super_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "supervisory coeff[%d] = %d\n", i, (int)nmt->super_coeff[i]);
audio_goertzel_init(&nmt->super_goertzel[i], super_freq[i], nmt->sender.samplerate);
if (i < 4) {
nmt->super_phaseshift256[i] = 256.0 / ((double)nmt->sender.samplerate / super_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "super_phaseshift[%d] = %.4f\n", i, nmt->super_phaseshift256[i]);
@@ -301,7 +294,7 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
{
double level, result[2], softbit, quality;
int max;
int16_t *spl;
sample_t *spl;
int bit;
max = nmt->fsk_filter_size;
@@ -316,7 +309,7 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
level = 0.01;
// level = 0.63662 / 2.0;
audio_goertzel(spl, max, pos, nmt->fsk_coeff, result, 2);
audio_goertzel(nmt->fsk_goertzel, spl, max, pos, result, 2);
/* calculate soft bit from both frequencies */
softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
@@ -368,14 +361,12 @@ static inline void fsk_decode_step(nmt_t *nmt, int pos)
}
/* compare supervisory signal against noise floor on 3900 Hz */
static void super_decode(nmt_t *nmt, int16_t *samples, int length)
static void super_decode(nmt_t *nmt, sample_t *samples, int length)
{
int coeff[2];
double result[2], quality;
coeff[0] = nmt->super_coeff[nmt->supervisory - 1];
coeff[1] = nmt->super_coeff[4]; /* noise floor detection */
audio_goertzel(samples, length, 0, coeff, result, 2);
audio_goertzel(&nmt->super_goertzel[nmt->supervisory - 1], samples, length, 0, &result[0], 1);
audio_goertzel(&nmt->super_goertzel[4], samples, length, 0, &result[1], 1); /* noise floor detection */
#if 0
/* normalize levels */
@@ -424,10 +415,10 @@ void super_reset(nmt_t *nmt)
}
/* Process received audio stream from radio unit. */
void sender_receive(sender_t *sender, int16_t *samples, int length)
void sender_receive(sender_t *sender, sample_t *samples, int length)
{
nmt_t *nmt = (nmt_t *) sender;
int16_t *spl;
sample_t *spl;
int max, pos;
double step, bps;
int i;
@@ -477,18 +468,17 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
if ((nmt->dsp_mode == DSP_MODE_AUDIO || nmt->dsp_mode == DSP_MODE_DTMF)
&& nmt->trans && nmt->trans->callref) {
int16_t down[length]; /* more than enough */
int count;
count = samplerate_downsample(&nmt->sender.srstate, samples, length, down);
count = samplerate_downsample(&nmt->sender.srstate, samples, length);
if (nmt->compandor)
expand_audio(&nmt->cstate, down, count);
expand_audio(&nmt->cstate, samples, count);
if (nmt->dsp_mode == DSP_MODE_DTMF)
dtmf_tone(&nmt->dtmf, down, count);
dtmf_tone(&nmt->dtmf, samples, count);
spl = nmt->sender.rxbuf;
pos = nmt->sender.rxbuf_pos;
for (i = 0; i < count; i++) {
spl[pos++] = down[i];
spl[pos++] = samples[i];
if (pos == 160) {
call_tx_audio(nmt->trans->callref, spl, 160);
pos = 0;
@@ -500,7 +490,7 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
}
/* render frame */
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
int fsk_render_frame(nmt_t *nmt, const char *frame, int length, sample_t *sample)
{
int bit, polarity;
double phaseshift, phase;
@@ -528,10 +518,10 @@ int fsk_render_frame(nmt_t *nmt, const char *frame, int length, int16_t *sample)
return count;
}
static int fsk_frame(nmt_t *nmt, int16_t *samples, int length)
static int fsk_frame(nmt_t *nmt, sample_t *samples, int length)
{
const char *frame;
int16_t *spl;
sample_t *spl;
int i;
int count, max;
@@ -575,23 +565,16 @@ next_frame:
}
/* Generate audio stream with supervisory signal. Keep phase for next call of function. */
static void super_encode(nmt_t *nmt, int16_t *samples, int length)
static void super_encode(nmt_t *nmt, sample_t *samples, int length)
{
double phaseshift, phase;
int32_t sample;
int i;
phaseshift = nmt->super_phaseshift256[nmt->supervisory - 1];
phase = nmt->super_phase256;
for (i = 0; i < length; i++) {
sample = *samples;
sample += dsp_sine_super[(uint8_t)phase];
if (sample > 32767)
sample = 32767;
else if (sample < -32767)
sample = -32767;
*samples++ = sample;
*samples++ += dsp_sine_super[(uint8_t)phase];
phase += phaseshift;
if (phase >= 256)
phase -= 256;
@@ -601,7 +584,7 @@ static void super_encode(nmt_t *nmt, int16_t *samples, int length)
}
/* Generate audio stream from dial tone. Keep phase for next call of function. */
static void dial_tone(nmt_t *nmt, int16_t *samples, int length)
static void dial_tone(nmt_t *nmt, sample_t *samples, int length)
{
double phaseshift, phase;
int i;
@@ -620,7 +603,7 @@ static void dial_tone(nmt_t *nmt, int16_t *samples, int length)
}
/* Provide stream of audio toward radio unit */
void sender_send(sender_t *sender, int16_t *samples, int length)
void sender_send(sender_t *sender, sample_t *samples, int length)
{
nmt_t *nmt = (nmt_t *) sender;
int len;