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Open sound device for capture or playback only, if full duplex is not required

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This commit is contained in:
Andreas Eversberg
2024-03-15 16:30:04 +01:00
parent ce58b765f5
commit b613123291
17 changed files with 125 additions and 81 deletions
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src/libsound/sound.h
+7 -1
View File
@@ -1,7 +1,13 @@
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 buffer_size, double interval, double max_deviation, double max_modulation, double modulation_index);
enum sound_direction {
SOUND_DIR_PLAY,
SOUND_DIR_REC,
SOUND_DIR_DUPLEX,
};
void *sound_open(int direction, 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);
src/libsound/sound_alsa.c
+59 -37
View File
@@ -32,6 +32,7 @@
static int KEEP_FRAMES=8; /* minimum frames not to read, to prevent reading from buffer before data has been received (seems to be a bug in ALSA) */
typedef struct sound {
enum sound_direction direction;
snd_pcm_t *phandle, *chandle;
int pchannels, cchannels;
int channels; /* required number of channels */
@@ -125,49 +126,57 @@ error:
static int dev_open(sound_t *sound)
{
int rc, rc_rec, rc_play;
int rc, rc_rec = 0, rc_play = 0;
rc_play = snd_pcm_open(&sound->phandle, sound->paudiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
rc_rec = snd_pcm_open(&sound->chandle, sound->caudiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if (rc_play < 0)
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports playing audio.\n", sound->paudiodev, snd_strerror(rc_play));
if (rc_rec < 0)
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports capturing audio.\n", sound->caudiodev, snd_strerror(rc_rec));
if (sound->direction == SOUND_DIR_PLAY || sound->direction == SOUND_DIR_DUPLEX) {
rc_play = snd_pcm_open(&sound->phandle, sound->paudiodev, SND_PCM_STREAM_PLAYBACK, SND_PCM_NONBLOCK);
if (rc_play < 0)
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for playback! (%s) Please select a device that supports playing audio.\n", sound->paudiodev, snd_strerror(rc_play));
}
if (sound->direction == SOUND_DIR_REC || sound->direction == SOUND_DIR_DUPLEX) {
rc_rec = snd_pcm_open(&sound->chandle, sound->caudiodev, SND_PCM_STREAM_CAPTURE, SND_PCM_NONBLOCK);
if (rc_rec < 0)
LOGP(DSOUND, LOGL_ERROR, "Failed to open '%s' for capture! (%s) Please select a device that supports capturing audio.\n", sound->caudiodev, snd_strerror(rc_rec));
}
if (rc_play < 0 || rc_rec < 0)
return (rc_play < 0) ? rc_play : rc_rec;
rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "Failed to set playback hw params\n");
return rc;
}
if (sound->pchannels < sound->channels) {
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
return rc;
}
LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
if (sound->direction == SOUND_DIR_PLAY || sound->direction == SOUND_DIR_DUPLEX) {
rc = set_hw_params(sound->phandle, sound->samplerate, &sound->pchannels);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "Failed to set playback hw params\n");
return rc;
}
if (sound->pchannels < sound->channels) {
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for playback.\n", sound->pchannels);
return rc;
}
LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "Failed to set capture hw params\n");
return rc;
}
if (sound->cchannels < sound->channels) {
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
return -EIO;
}
LOGP(DSOUND, LOGL_DEBUG, "Capture with %d channels.\n", sound->cchannels);
rc = snd_pcm_prepare(sound->phandle);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
rc = snd_pcm_prepare(sound->phandle);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
}
}
rc = snd_pcm_prepare(sound->chandle);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
if (sound->direction == SOUND_DIR_REC || sound->direction == SOUND_DIR_DUPLEX) {
rc = set_hw_params(sound->chandle, sound->samplerate, &sound->cchannels);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "Failed to set capture hw params\n");
return rc;
}
if (sound->cchannels < sound->channels) {
LOGP(DSOUND, LOGL_ERROR, "Sound card only supports %d channel for capture.\n", sound->cchannels);
return -EIO;
}
LOGP(DSOUND, LOGL_DEBUG, "Capture with %d channels.\n", sound->cchannels);
rc = snd_pcm_prepare(sound->chandle);
if (rc < 0) {
LOGP(DSOUND, LOGL_ERROR, "cannot prepare audio interface for use (%s)\n", snd_strerror(rc));
return rc;
}
}
return 0;
@@ -181,7 +190,7 @@ 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)) buffer_size, double __attribute__((unused)) interval, double max_deviation, double __attribute__((unused)) max_modulation, double __attribute__((unused)) modulation_index)
void *sound_open(int direction, 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;
const char *env;
@@ -206,6 +215,7 @@ void *sound_open(const char *audiodev, double __attribute__((unused)) *tx_freque
} else {
sound->caudiodev = sound->paudiodev;
}
sound->direction = direction;
sound->channels = channels;
sound->samplerate = samplerate;
sound->spl_deviation = max_deviation / 32767.0;
@@ -249,6 +259,9 @@ int sound_start(void *inst)
sound_t *sound = (sound_t *)inst;
int16_t buff[2];
if (sound->direction != SOUND_DIR_REC && sound->direction != SOUND_DIR_DUPLEX)
return -EINVAL;
/* trigger capturing */
snd_pcm_readi(sound->chandle, buff, 1);
@@ -319,6 +332,9 @@ int sound_write(void *inst, sample_t **samples, uint8_t __attribute__((unused))
int rc;
int i, ii;
if (sound->direction != SOUND_DIR_PLAY && sound->direction != SOUND_DIR_DUPLEX)
return -EINVAL;
if (sound->pchannels == 2) {
/* two channels */
#ifdef HAVE_MOBILE
@@ -404,6 +420,9 @@ int sound_read(void *inst, sample_t **samples, int num, int channels, double *rf
int in, rc;
int i, ii;
if (sound->direction != SOUND_DIR_REC && sound->direction != SOUND_DIR_DUPLEX)
return -EINVAL;
/* get samples in rx buffer */
in = snd_pcm_avail(sound->chandle);
/* if not more than KEEP_FRAMES frames available, try next time */
@@ -501,6 +520,9 @@ int sound_get_tosend(void *inst, int buffer_size)
snd_pcm_sframes_t delay;
int tosend;
if (sound->direction != SOUND_DIR_PLAY && sound->direction != SOUND_DIR_DUPLEX)
return -EINVAL;
rc = snd_pcm_delay(sound->phandle, &delay);
if (rc < 0) {
if (rc == -32)