williamp/osmocom-analog
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Refactor global variables for signal processing

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These are:
	device, sample rate, buffer, latency
Called now:
	dsp_device, dsp_samplerate, dsp_buffer, dsp_latency
Call audio device:
	call_device, call_samplerate, call_buffer
This commit is contained in:
Andreas Eversberg
2021-09-18 11:43:01 +02:00
parent de685b3cb6
commit 6fa74a1296
57 changed files with 379 additions and 368 deletions
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4
src/libsound/sound.h
View File

@@ -1,10 +1,10 @@
enum paging_signal;
void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int latspl, double max_deviation, double max_modulation, double modulation_index);
void *sound_open(const char *audiodev, double *tx_frequency, double *rx_frequency, int *am, int channels, double paging_frequency, int samplerate, int buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
int sound_start(void *inst);
void sound_close(void *inst);
int sound_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);
int sound_get_tosend(void *inst, int latspl);
int sound_get_tosend(void *inst, int buffer_size);

8
src/libsound/sound_alsa.c
View File

@@ -187,7 +187,7 @@ static void dev_close(sound_t *sound)
snd_pcm_close(sound->chandle);
}
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) latspl, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_frequency, double __attribute__((unused)) *rx_frequency, int __attribute__((unused)) *am, int channels, double __attribute__((unused)) paging_frequency, int samplerate, int __attribute((unused)) buffer_size, double __attribute__((unused)) interval, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
{
sound_t *sound;
int rc;
@@ -487,7 +487,7 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double __a
* get playback buffer space
*
* return number of samples to be sent */
int sound_get_tosend(void *inst, int latspl)
int sound_get_tosend(void *inst, int buffer_size)
{
sound_t *sound = (sound_t *)inst;
int rc;
@@ -497,7 +497,7 @@ int sound_get_tosend(void *inst, int latspl)
rc = snd_pcm_delay(sound->phandle, &delay);
if (rc < 0) {
if (rc == -32)
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency and enable real time scheduling\n");
PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher buffer and enable real time scheduling\n");
else
PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
if (rc == -EPIPE) {
@@ -511,7 +511,7 @@ int sound_get_tosend(void *inst, int latspl)
return rc;
}
tosend = latspl - delay;
tosend = buffer_size - delay;
return tosend;
}