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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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80
src/amps/dsp.c
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@@ -81,10 +81,10 @@
#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 "amps.h"
#include "frame.h"
#include "dsp.h"
@@ -180,8 +180,7 @@ static void sat_reset(amps_t *amps, const char *reason);
/* Init FSK of transceiver */
int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
{
double coeff;
int16_t *spl;
sample_t *spl;
int i;
int rc;
double RC, dt;
@@ -206,12 +205,13 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
PDEBUG(DDSP, DEBUG_DEBUG, "Use %.4f samples for full bit duration @ %d.\n", amps->fsk_bitduration, amps->sender.samplerate);
amps->fsk_tx_buffer_size = amps->fsk_bitduration + 10; /* 10 extra to avoid overflow due to rounding */
amps->fsk_tx_buffer = calloc(sizeof(int16_t), amps->fsk_tx_buffer_size);
if (!amps->fsk_tx_buffer) {
spl = calloc(sizeof(*spl), amps->fsk_tx_buffer_size);
if (!spl) {
PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
rc = -ENOMEM;
goto error;
}
amps->fsk_tx_buffer = spl;
amps->fsk_rx_window_length = ceil(amps->fsk_bitduration); /* buffer holds one bit (rounded up) */
half = amps->fsk_rx_window_length >> 1;
@@ -221,12 +221,13 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
PDEBUG(DDSP, DEBUG_DEBUG, "Bit window length: %d\n", amps->fsk_rx_window_length);
PDEBUG(DDSP, DEBUG_DEBUG, " -> Samples in window to analyse level left of edge: %d..%d\n", amps->fsk_rx_window_begin, amps->fsk_rx_window_half - 1);
PDEBUG(DDSP, DEBUG_DEBUG, " -> Samples in window to analyse level right of edge: %d..%d\n", amps->fsk_rx_window_half, amps->fsk_rx_window_end - 1);
amps->fsk_rx_window = calloc(sizeof(int16_t), amps->fsk_rx_window_length);
if (!amps->fsk_rx_window) {
spl = calloc(sizeof(*amps->fsk_rx_window), amps->fsk_rx_window_length);
if (!spl) {
PDEBUG(DDSP, DEBUG_DEBUG, "No memory!\n");
rc = -ENOMEM;
goto error;
}
amps->fsk_rx_window = spl;
/* create devation and ramp */
amps->fsk_deviation = FSK_DEVIATION; /* be sure not to overflow 32767 */
@@ -234,7 +235,7 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
/* allocate ring buffer for SAT signal detection */
amps->sat_samples = (int)((double)amps->sender.samplerate * SAT_DURATION + 0.5);
spl = calloc(1, amps->sat_samples * sizeof(*spl));
spl = calloc(sizeof(*spl), amps->sat_samples);
if (!spl) {
PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
return -ENOMEM;
@@ -243,10 +244,7 @@ int dsp_init_sender(amps_t *amps, int high_pass, int tolerant)
/* count SAT tones */
for (i = 0; i < 5; i++) {
coeff = 2.0 * cos(2.0 * PI * sat_freq[i] / (double)amps->sender.samplerate);
amps->sat_coeff[i] = coeff * 32768.0;
PDEBUG(DDSP, DEBUG_DEBUG, "sat_coeff[%d] = %d\n", i, (int)amps->sat_coeff[i]);
audio_goertzel_init(&amps->sat_goertzel[i], sat_freq[i], amps->sender.samplerate);
if (i < 3) {
amps->sat_phaseshift256[i] = 256.0 / ((double)amps->sender.samplerate / sat_freq[i]);
PDEBUG(DDSP, DEBUG_DEBUG, "sat_phaseshift256[%d] = %.4f\n", i, amps->sat_phaseshift256[i]);
@@ -300,7 +298,7 @@ void dsp_cleanup_sender(amps_t *amps)
static int fsk_encode(amps_t *amps, char bit)
{
int16_t *spl;
sample_t *spl;
double phase, bitstep, deviation;
int count;
char last;
@@ -368,10 +366,10 @@ static int fsk_encode(amps_t *amps, char bit)
return count;
}
static int fsk_frame(amps_t *amps, int16_t *samples, int length)
static int fsk_frame(amps_t *amps, sample_t *samples, int length)
{
int count = 0, len, pos, copy, i;
int16_t *spl;
sample_t *spl;
int rc;
char c;
@@ -430,7 +428,7 @@ done:
}
/* Generate audio stream with SAT signal. Keep phase for next call of function. */
static void sat_encode(amps_t *amps, int16_t *samples, int length)
static void sat_encode(amps_t *amps, sample_t *samples, int length)
{
double phaseshift, phase;
int32_t sample;
@@ -455,7 +453,7 @@ static void sat_encode(amps_t *amps, int16_t *samples, int length)
amps->sat_phase256 = phase;
}
static void test_tone_encode(amps_t *amps, int16_t *samples, int length)
static void test_tone_encode(amps_t *amps, sample_t *samples, int length)
{
double phaseshift, phase;
int i;
@@ -474,7 +472,7 @@ static void test_tone_encode(amps_t *amps, 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)
{
amps_t *amps = (amps_t *) sender;
int count;
@@ -505,12 +503,12 @@ again:
}
}
static void fsk_rx_bit(amps_t *amps, int16_t *spl, int len, int pos, int begin, int half, int end)
static void fsk_rx_bit(amps_t *amps, sample_t *spl, int len, int pos, int begin, int half, int end)
{
int i;
int32_t first, second;
double first, second;
int bit;
int32_t max = -32768, min = 32767;
double max = 0, min = 0;
/* decode one bit. substact the first half from the second half.
* the result shows the direction of the bit change: 1 == positive.
@@ -522,9 +520,9 @@ static void fsk_rx_bit(amps_t *amps, int16_t *spl, int len, int pos, int begin,
pos += len;
//printf("second %d: %d\n", pos, spl[pos]);
second += spl[pos];
if (spl[pos] > max)
if (i == 0 || spl[pos] > max)
max = spl[pos];
if (spl[pos] < min)
if (i == 0 || spl[pos] < min)
min = spl[pos];
}
second /= (half - begin);
@@ -683,13 +681,13 @@ static void fsk_rx_dotting(amps_t *amps, double _elapsed)
}
/* decode frame */
static void sender_receive_frame(amps_t *amps, int16_t *samples, int length)
static void sender_receive_frame(amps_t *amps, sample_t *samples, int length)
{
int i;
for (i = 0; i < length; i++) {
#ifdef DEBUG_DECODER
puts(debug_amplitude((double)samples[i] / (double)FSK_DEVIATION));
puts(debug_amplitude(samples[i] / (double)FSK_DEVIATION));
#endif
/* push sample to detection window and shift */
amps->fsk_rx_window[amps->fsk_rx_window_pos++] = samples[i];
@@ -731,15 +729,13 @@ static void sender_receive_frame(amps_t *amps, int16_t *samples, int length)
/* decode signaling tone */
/* compare supervisory signal against noise floor on 5800 Hz */
static void sat_decode(amps_t *amps, int16_t *samples, int length)
static void sat_decode(amps_t *amps, sample_t *samples, int length)
{
int coeff[3];
double result[3], quality[2];
coeff[0] = amps->sat_coeff[amps->sat];
coeff[1] = amps->sat_coeff[3]; /* noise floor detection */
coeff[2] = amps->sat_coeff[4]; /* signaling tone */
audio_goertzel(samples, length, 0, coeff, result, 3);
audio_goertzel(&amps->sat_goertzel[amps->sat], samples, length, 0, &result[0], 1);
audio_goertzel(&amps->sat_goertzel[3], samples, length, 0, &result[1], 1);
audio_goertzel(&amps->sat_goertzel[4], samples, length, 0, &result[2], 1);
quality[0] = (result[0] - result[1]) / result[0];
if (quality[0] < 0)
@@ -805,10 +801,10 @@ static void sat_decode(amps_t *amps, int16_t *samples, int length)
* time is between SIG_TONE_MINBITS and SIG_TONE_MAXBITS. If it is, the
* frequency is close to the singalling tone, so it is detected
*/
static void sender_receive_audio(amps_t *amps, int16_t *samples, int length)
static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
{
transaction_t *trans = amps->trans_list;
int16_t *spl;
sample_t *spl;
int max, pos;
int i;
@@ -830,21 +826,19 @@ static void sender_receive_audio(amps_t *amps, int16_t *samples, int length)
if ((amps->dsp_mode == DSP_MODE_AUDIO_RX_AUDIO_TX || amps->dsp_mode == DSP_MODE_AUDIO_RX_FRAME_TX)
&& trans && trans->callref && trans->sat_detected) {
int16_t down[length]; /* more than enough */
int pos, count;
int16_t *spl;
int i;
/* de-emphasis */
if (amps->de_emphasis)
de_emphasis(&amps->estate, samples, length);
/* downsample */
count = samplerate_downsample(&amps->sender.srstate, samples, length, down);
expand_audio(&amps->cstate, down, count);
count = samplerate_downsample(&amps->sender.srstate, samples, length);
expand_audio(&amps->cstate, samples, count);
spl = amps->sender.rxbuf;
pos = amps->sender.rxbuf_pos;
for (i = 0; i < count; i++) {
spl[pos++] = down[i];
spl[pos++] = samples[i];
if (pos == 160) {
call_tx_audio(trans->callref, spl, 160);
pos = 0;
@@ -856,11 +850,10 @@ static void sender_receive_audio(amps_t *amps, int16_t *samples, int length)
}
/* 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)
{
amps_t *amps = (amps_t *) sender;
double x, y, x_last, y_last, factor;
int32_t value;
int i;
/* high pass filter to remove 0-level
@@ -874,12 +867,7 @@ void sender_receive(sender_t *sender, int16_t *samples, int length)
y = factor * (y_last + x - x_last);
x_last = x;
y_last = y;
value = (int32_t)(y + 0.5);
if (value < -32768.0)
value = -32768.0;
else if (value > 32767)
value = 32767;
samples[i] = value;
samples[i] = y;
}
amps->highpass_x_last = x_last;
amps->highpass_y_last = y_last;