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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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71
src/common/sound_alsa.c
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@@ -20,6 +20,7 @@
#include <stdlib.h>
#include <stdint.h>
#include <alsa/asoundlib.h>
#include "sample.h"
#include "debug.h"
#include "sender.h"
@@ -248,35 +249,66 @@ static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum p
}
}
int sound_write(void *inst, int16_t **samples, int num, enum paging_signal *paging_signal, int *on, int channels)
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;
int32_t value;
int16_t buff[num << 1];
int rc;
int i, ii;
if (sound->pchannels == 2) {
/* two channels */
if (paging_signal && on && paging_signal[0] != PAGING_SIGNAL_NONE) {
int16_t paging[num << 1];
gen_paging_tone(sound, paging, num, paging_signal[0], on[0]);
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = samples[0][i];
value = samples[0][i];
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
buff[ii++] = paging[i];
}
} else if (channels == 2) {
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = samples[0][i];
buff[ii++] = samples[1][i];
value = samples[0][i];
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
value = samples[1][i];
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
}
} else {
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = samples[0][i];
buff[ii++] = samples[0][i];
value = samples[0][i];
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
buff[ii++] = value;
}
}
rc = snd_pcm_writei(sound->phandle, buff, num);
} else
rc = snd_pcm_writei(sound->phandle, samples[0], num);
} else {
/* one channel */
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i];
if (value > 32767)
value = 32767;
else if (value < -32767)
value = -32767;
buff[ii++] = value;
}
}
rc = snd_pcm_writei(sound->phandle, buff, num);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
@@ -293,7 +325,7 @@ int sound_write(void *inst, int16_t **samples, int num, enum paging_signal *pagi
#define KEEP_FRAMES 8 /* minimum frames not to read, due to bug in ALSA */
int sound_read(void *inst, int16_t **samples, int num, int channels)
int sound_read(void *inst, sample_t **samples, int num, int channels)
{
sound_t *sound = (sound_t *)inst;
int16_t buff[num << 1];
@@ -312,10 +344,7 @@ int sound_read(void *inst, int16_t **samples, int num, int channels)
if (in > num)
in = num;
if (sound->cchannels == 2)
rc = snd_pcm_readi(sound->chandle, buff, in);
else
rc = snd_pcm_readi(sound->chandle, samples[0], in);
rc = snd_pcm_readi(sound->chandle, buff, in);
if (rc < 0) {
if (errno == EAGAIN)
@@ -332,18 +361,18 @@ int sound_read(void *inst, int16_t **samples, int num, int channels)
for (i = 0, ii = 0; i < rc; i++) {
spl = buff[ii++];
spl += buff[ii++];
if (spl > 32767)
spl = 32767;
else if (spl < -32768)
spl = -32768;
samples[0][i] = spl;
samples[0][i] = (sample_t)spl;
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = buff[ii++];
samples[1][i] = buff[ii++];
samples[0][i] = (sample_t)buff[ii++];
samples[1][i] = (sample_t)buff[ii++];
}
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = (sample_t)buff[ii++];
}
}
return rc;