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authorAndreas Eversberg <jolly@eversberg.eu>2021-03-14 11:57:52 +0100
committerAndreas Eversberg <jolly@eversberg.eu>2021-03-28 13:05:58 +0200
commit65b7d3f2e77f20b1e7e5719302688e15c7c2da06 (patch)
tree234e97bdd857f9ceac8b142cbaf23f9329a076a1
parent6ed53c5699fbf48ada37e4cbc8f3ddad1262ce41 (diff)
Make Alsa sound interface indipendent from libmobile
-rw-r--r--src/libsound/sound_alsa.c23
1 files changed, 20 insertions, 3 deletions
diff --git a/src/libsound/sound_alsa.c b/src/libsound/sound_alsa.c
index eff6b81..6216b73 100644
--- a/src/libsound/sound_alsa.c
+++ b/src/libsound/sound_alsa.c
@@ -23,7 +23,11 @@
#include <alsa/asoundlib.h>
#include "../libsample/sample.h"
#include "../libdebug/debug.h"
+#ifdef HAVE_MOBILE
#include "../libmobile/sender.h"
+#else
+#include "sound.h"
+#endif
typedef struct sound {
snd_pcm_t *phandle, *chandle;
@@ -32,10 +36,12 @@ typedef struct sound {
int samplerate; /* required sample rate */
char *audiodev; /* required device */
double spl_deviation; /* how much deviation is one sample step */
+#ifdef HAVE_MOBILE
double paging_phaseshift; /* phase to shift every sample */
double paging_phase; /* current phase */
double rx_frequency[2]; /* rx frequency of radio connected to channel */
dispmeasparam_t *dmp[2];
+#endif
} sound_t;
static int set_hw_params(snd_pcm_t *handle, int samplerate, int *channels)
@@ -201,12 +207,15 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
sound->channels = channels;
sound->samplerate = samplerate;
sound->spl_deviation = max_deviation / 32767.0;
+#ifdef HAVE_MOBILE
sound->paging_phaseshift = 1.0 / ((double)samplerate / 1000.0);
+#endif
rc = dev_open(sound);
if (rc < 0)
goto error;
+#ifdef HAVE_MOBILE
if (rx_frequency) {
sender_t *sender;
int i;
@@ -218,6 +227,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
sound->dmp[i] = display_measurements_add(&sender->dispmeas, "RX Level", "%.1f dB", DISPLAY_MEAS_PEAK, DISPLAY_MEAS_LEFT, -96.0, 0.0, -INFINITY);
}
}
+#endif
return sound;
@@ -247,6 +257,7 @@ void sound_close(void *inst)
free(sound);
}
+#ifdef HAVE_MOBILE
static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum paging_signal paging_signal, int on)
{
double phaseshift, phase;
@@ -290,8 +301,9 @@ static void gen_paging_tone(sound_t *sound, int16_t *samples, int length, enum p
break;
}
}
+#endif
-int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused)) **power, int num, enum paging_signal *paging_signal, int *on, int channels)
+int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused)) **power, int num, enum paging_signal __attribute__((unused)) *paging_signal, int __attribute__((unused)) *on, int channels)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
@@ -302,6 +314,7 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
if (sound->pchannels == 2) {
/* two channels */
+#ifdef HAVE_MOBILE
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]);
@@ -314,7 +327,9 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
buff[ii++] = value;
buff[ii++] = paging[i];
}
- } else if (channels == 2) {
+ } else
+#endif
+ if (channels == 2) {
for (i = 0, ii = 0; i < num; i++) {
value = samples[0][i] / spl_deviation;
if (value > 32767)
@@ -374,7 +389,7 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
#define KEEP_FRAMES 8 /* minimum frames not to read, due to bug in ALSA */
-int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db)
+int sound_read(void *inst, sample_t **samples, int num, int channels, double __attribute__((unused)) *rf_level_db)
{
sound_t *sound = (sound_t *)inst;
double spl_deviation = sound->spl_deviation;
@@ -453,6 +468,7 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf
}
}
+#ifdef HAVE_MOBILE
sender_t *sender;
for (i = 0; i < channels; i++) {
sender = get_sender_by_empfangsfrequenz(sound->rx_frequency[i]);
@@ -462,6 +478,7 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf
if (rf_level_db)
rf_level_db[i] = 0.0;
}
+#endif
return rc;
}