williamp/osmocom-analog
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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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33
src/bnetz/dsp.c
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@@ -25,10 +25,10 @@
#include <string.h>
#include <errno.h>
#include <math.h>
#include "../common/sample.h"
#include "../common/debug.h"
#include "../common/timer.h"
#include "../common/call.h"
#include "../common/goertzel.h"
#include "bnetz.h"
#include "dsp.h"
@@ -54,7 +54,7 @@ static double fsk_bits[2] = {
};
/* table for fast sine generation */
int dsp_sine[256];
static sample_t dsp_sine[256];
/* global init for FSK */
void dsp_init(void)
@@ -75,8 +75,7 @@ void dsp_init(void)
/* Init transceiver instance. */
int dsp_init_sender(bnetz_t *bnetz)
{
double coeff;
int16_t *spl;
sample_t *spl;
int i;
if ((bnetz->sender.samplerate % 1000)) {
@@ -114,10 +113,7 @@ int dsp_init_sender(bnetz_t *bnetz)
/* count symbols */
for (i = 0; i < 2; i++) {
coeff = 2.0 * cos(2.0 * PI * fsk_bits[i] / (double)bnetz->sender.samplerate);
bnetz->fsk_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "coeff[%d] = %d (must be -3601 and 2573 at 8000hz)\n", i, (int)bnetz->fsk_coeff[i]);
audio_goertzel_init(&bnetz->fsk_goertzel[i], fsk_bits[i], bnetz->sender.samplerate);
bnetz->phaseshift256[i] = 256.0 / ((double)bnetz->sender.samplerate / fsk_bits[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "phaseshift[%d] = %.4f (must be arround 64 at 8000hz)\n", i, bnetz->phaseshift256[i]);
}
@@ -205,7 +201,7 @@ static inline void fsk_decode_step(bnetz_t *bnetz, int pos)
{
double level, result[2], softbit, quality;
int max;
int16_t *spl;
sample_t *spl;
int bit;
max = bnetz->samples_per_bit;
@@ -216,7 +212,7 @@ static inline void fsk_decode_step(bnetz_t *bnetz, int pos)
if (audio_detect_loss(&bnetz->sender.loss, level))
bnetz_loss_indication(bnetz);
audio_goertzel(spl, max, pos, bnetz->fsk_coeff, result, 2);
audio_goertzel(bnetz->fsk_goertzel, spl, max, pos, result, 2);
/* calculate soft bit from both frequencies */
softbit = (result[1] / level - result[0] / level + 1.0) / 2.0;
@@ -266,10 +262,10 @@ static inline void fsk_decode_step(bnetz_t *bnetz, int pos)
}
/* 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)
{
bnetz_t *bnetz = (bnetz_t *) sender;
int16_t *spl;
sample_t *spl;
int max, pos, step;
int i;
@@ -290,14 +286,13 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
bnetz->fsk_filter_pos = pos;
if (bnetz->dsp_mode == DSP_MODE_AUDIO && bnetz->callref) {
int16_t down[length]; /* more than enough */
int count;
count = samplerate_downsample(&bnetz->sender.srstate, samples, length, down);
count = samplerate_downsample(&bnetz->sender.srstate, samples, length);
spl = bnetz->sender.rxbuf;
pos = bnetz->sender.rxbuf_pos;
for (i = 0; i < count; i++) {
spl[pos++] = down[i];
spl[pos++] = samples[i];
if (pos == 160) {
call_tx_audio(bnetz->callref, spl, 160);
pos = 0;
@@ -308,7 +303,7 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
bnetz->sender.rxbuf_pos = 0;
}
static void fsk_tone(bnetz_t *bnetz, int16_t *samples, int length, int tone)
static void fsk_tone(bnetz_t *bnetz, sample_t *samples, int length, int tone)
{
double phaseshift, phase;
int i;
@@ -326,9 +321,9 @@ static void fsk_tone(bnetz_t *bnetz, int16_t *samples, int length, int tone)
bnetz->phase256 = phase;
}
static int fsk_telegramm(bnetz_t *bnetz, int16_t *samples, int length)
static int fsk_telegramm(bnetz_t *bnetz, sample_t *samples, int length)
{
int16_t *spl;
sample_t *spl;
const char *telegramm;
int i, j;
double phaseshift, phase;
@@ -382,7 +377,7 @@ next_telegramm:
}
/* 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)
{
bnetz_t *bnetz = (bnetz_t *) sender;
int len;