williamp/osmocom-analog
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prepare audio handling for multi carrier SDR

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This commit is contained in:
Andreas Eversberg
2017-01-03 12:31:59 +01:00
parent 713b1ab118
commit d54d3ac265
23 changed files with 131 additions and 176 deletions
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49
src/common/sound_alsa.c
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@@ -187,24 +187,29 @@ void sound_close(void *inst)
free(sound);
}
int sound_write(void *inst, int16_t *samples_left, int16_t *samples_right, int num)
int sound_write(void *inst, int16_t **samples, int num, int channels)
{
sound_t *sound = (sound_t *)inst;
int16_t buff[num << 1], *samples;
int16_t buff[num << 1];
int rc;
int i, ii;
if (sound->pchannels == 2) {
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = *samples_right++;
buff[ii++] = *samples_left++;
if (channels < 2) {
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = samples[0][i];
buff[ii++] = samples[0][i];
}
} else {
for (i = 0, ii = 0; i < num; i++) {
buff[ii++] = samples[0][i];
buff[ii++] = samples[1][i];
}
}
samples = buff;
rc = snd_pcm_writei(sound->phandle, buff, num);
} else
samples = samples_left;
rc = snd_pcm_writei(sound->phandle, samples[0], num);
rc = snd_pcm_writei(sound->phandle, samples, num);
if (rc < 0) {
PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE)
@@ -220,10 +225,11 @@ int sound_write(void *inst, int16_t *samples_left, int16_t *samples_right, int n
#define KEEP_FRAMES 8 /* minimum frames not to read, due to bug in ALSA */
int sound_read(void *inst, int16_t *samples_left, int16_t *samples_right, int num)
int sound_read(void *inst, int16_t **samples, int num, int channels)
{
sound_t *sound = (sound_t *)inst;
int16_t buff[num << 1];
int32_t spl;
int in, rc;
int i, ii;
@@ -241,7 +247,7 @@ int sound_read(void *inst, int16_t *samples_left, int16_t *samples_right, int nu
if (sound->cchannels == 2)
rc = snd_pcm_readi(sound->chandle, buff, in);
else
rc = snd_pcm_readi(sound->chandle, samples_left, in);
rc = snd_pcm_readi(sound->chandle, samples[0], in);
if (rc < 0) {
if (errno == EAGAIN)
@@ -254,12 +260,23 @@ int sound_read(void *inst, int16_t *samples_left, int16_t *samples_right, int nu
}
if (sound->cchannels == 2) {
for (i = 0, ii = 0; i < rc; i++) {
*samples_right++ = buff[ii++];
*samples_left++ = buff[ii++];
if (channels < 2) {
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;
}
} else {
for (i = 0, ii = 0; i < rc; i++) {
samples[0][i] = buff[ii++];
samples[1][i] = buff[ii++];
}
}
} else
memcpy(samples_right, samples_left, num << 1);
}
return rc;
}