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Correcting all levels and move all remaining integer samples to sample_t

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The leves are based on the standards of each mobile network. They
are adjusted to the specified frequency deviation now.
This commit is contained in:
Andreas Eversberg
2017-01-29 07:25:12 +01:00
parent bd7ccc5fa0
commit 7e45f556ce
38 changed files with 484 additions and 397 deletions
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24
src/common/sound_alsa.c
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@@ -27,6 +27,7 @@
typedef struct sound {
snd_pcm_t *phandle, *chandle;
int pchannels, cchannels;
double spl_deviation; /* how much deviation is one sample step */
double paging_phaseshift; /* phase to shift every sample */
double paging_phase; /* current phase */
} sound_t;
@@ -131,7 +132,7 @@ static int sound_prepare(sound_t *sound)
return 0;
}
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, double __attribute__((unused)) bandwidth, double __attribute__((unused)) sample_deviation)
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int channels, double __attribute__((unused)) paging_frequency, int samplerate, double max_deviation, double __attribute__((unused)) max_modulation)
{
sound_t *sound;
int rc;
@@ -147,6 +148,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
return NULL;
}
sound->spl_deviation = max_deviation / 32767.0;
sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
rc = snd_pcm_open(&sound->phandle, audiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
@@ -252,6 +254,7 @@ static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum p
int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *paging_signal, int *on, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int32_t value;
int16_t buff[num << 1];
int rc;
@@ -263,7 +266,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
int16_t paging[num << 1];
gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -273,13 +276,13 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
}
} else if (channels == 2) {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
value = samples[1][i];
value = samples[1][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -288,7 +291,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
}
} else {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -300,7 +303,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
} else {
/* one channel */
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
value = samples[0][i] / spl_deviation;
if (value > 32767)
value = 32767;
else if (value < -32767)
@@ -328,6 +331,7 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
int sound_read(void *inst, sample_t **samples, int num, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
int16_t buff[num << 1];
int32_t spl;
int in, rc;
@@ -361,17 +365,17 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
spl += buff[ii++];
samples[0][i] = (sample_t)spl;
samples[0][i] = (double)spl * spl_deviation;
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = (sample_t)buff[ii++];
samples[1][i] = (sample_t)buff[ii++];
samples[0][i] = (double)buff[ii++] * spl_deviation;
samples[1][i] = (double)buff[ii++] * spl_deviation;
}
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = (sample_t)buff[ii++];
samples[0][i] = (double)buff[ii++] * spl_deviation;
}
}