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

src/bnetz/dialer.c

@@ -346,7 +346,7 @@ int main(int argc, char *argv[])
 
 #ifdef HAVE_ALSA
 	/* init sound */
-	audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
+	audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
 	if (!audio) {
 		LOGP(DBNETZ, LOGL_ERROR, "No sound device!\n");
 		goto exit;

src/datenklo/datenklo.c

@@ -1397,7 +1397,7 @@ int datenklo_open_audio(datenklo_t *datenklo, const char *audiodev, int buffer,
 
 #ifdef HAVE_ALSA
 	/* init sound */
-	datenklo->audio = sound_open(audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->buffer_size, 1.0, 1.0, 4000.0, 2.0);
+	datenklo->audio = sound_open(SOUND_DIR_DUPLEX, audiodev, NULL, NULL, NULL, channels, 0.0, datenklo->samplerate, datenklo->buffer_size, 1.0, 1.0, 4000.0, 2.0);
 	if (!datenklo->audio) {
 		LOGP(DDATENKLO, LOGL_ERROR, "No sound device!\n");
 		return -EIO;

src/dcf77/main.c

@@ -413,13 +413,19 @@ static int get_char()
 
 int soundif_open(const char *audiodev, int samplerate, int buffer_size)
 {
+	enum sound_direction direction = SOUND_DIR_DUPLEX;
+
 	if (!audiodev || !audiodev[0]) {
 		LOGP(DDSP, LOGL_ERROR, "No audio device given!\n");
 		return -EINVAL;
 	}
 
 	/* open audiodev */
-	soundif = sound_open(audiodev, NULL, NULL, NULL, (double_amplitude) ? 2 : 1, 0.0, samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
+	if (tx && !rx)
+		direction = SOUND_DIR_PLAY;
+	if (rx && !tx)
+		direction = SOUND_DIR_REC;
+	soundif = sound_open(direction, audiodev, NULL, NULL, NULL, (double_amplitude) ? 2 : 1, 0.0, samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
 	if (!soundif) {
 		LOGP(DDSP, LOGL_ERROR, "Failed to open sound device!\n");
 		return -EIO;
@@ -451,6 +457,7 @@ void soundif_work(int buffer_size)
 	int rc;
 	int i;
 
+	if (tx) {
 		/* encode and write */
 		count = sound_get_tosend(soundif, buffer_size);
 		if (count < 0) {
@@ -469,7 +476,9 @@ void soundif_work(int buffer_size)
 				return;
 			}
 		}
+	}
 
+	if (rx) {
 		/* read */
 		count = sound_read(soundif, samples, buffer_size, 1, rf_level_db);
 		if (count < 0) {
@@ -480,6 +489,7 @@ void soundif_work(int buffer_size)
 		/* decode */
 		dcf77_decode(dcf77, samples[0], count);
 	}
+}
 
 int main(int argc, char *argv[])
 {

src/imts/dialer.c

@@ -307,7 +307,7 @@ int main(int argc, char *argv[])
 
 #ifdef HAVE_ALSA
 	/* init sound */
-	audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, dsp_buffer, 1.0, 1.0, 4000.0, 2.0);
+	audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, dsp_buffer, 1.0, 1.0, 4000.0, 2.0);
 	if (!audio) {
 		LOGP(DBNETZ, LOGL_ERROR, "No sound device!\n");
 		goto exit;

src/libjitter/jitter.c

@@ -348,7 +348,7 @@ int32_t jitter_offset(jitter_t *jb)
 	jb->unlocked = true;
 
 	/* get timestamp of chunk that is not in the past */
-	while ((jf = jb->frame_list)) {
+	for (jf = jb->frame_list; jf; jf = jf->next) {
 		offset_timestamp = jf->timestamp - jb->window_timestamp;
 		if (offset_timestamp >= 0)
 			break;
@@ -446,6 +446,9 @@ copy_chunk:
 		tocopy = jb->spl_len - jb->spl_pos;
 		if (tocopy > len)
 			tocopy = len;
+#ifdef HEAVY_DEBUG
+		LOGP(DJITTER, LOGL_DEBUG, "%s loading %d samples: from valid sample buffer.\n", jb->name, tocopy);
+#endif
 		/* advance jitter buffer */
 		jitter_advance(jb, tocopy);
 		memcpy(spl, jb->spl_buf + jb->spl_pos * sample_size, tocopy * sample_size);
@@ -469,6 +472,9 @@ copy_chunk:
 		/* only process as much samples as need */
 		if (offset > len)
 			offset = len;
+#ifdef HEAVY_DEBUG
+		LOGP(DJITTER, LOGL_DEBUG, "%s concealing %d samples: from invalid sample buffer.\n", jb->name, offset);
+#endif
 		/* advance jitter buffer */
 		jitter_advance(jb, offset);
 		/* if there is no buffer, allocate 20ms, filled with 0 */
@@ -501,6 +507,9 @@ copy_chunk:
 		jitter_reset(jb);
 		return;
 	}
+#ifdef HEAVY_DEBUG
+	LOGP(DJITTER, LOGL_DEBUG, "%s loading new frame to sample buffer.\n", jb->name);
+#endif
 	/* get data from frame */
 	jitter_frame_get(jf, &decoder, &decoder_priv, &payload, &payload_len, NULL, NULL, NULL, NULL);
 	/* free previous buffer */

src/libmobile/console.c

@@ -404,7 +404,7 @@ int console_open_audio(int __attribute__((unused)) buffer_size, double __attribu
 #ifdef HAVE_ALSA
 	/* open sound device for call control */
 	/* use factor 1.4 of speech level for complete range of sound card */
-	console.sound = sound_open(console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, buffer_size, interval, 1.4, 4000.0, 2.0);
+	console.sound = sound_open(SOUND_DIR_DUPLEX, console.audiodev, NULL, NULL, NULL, 1, 0.0, console.samplerate, buffer_size, interval, 1.4, 4000.0, 2.0);
 	if (!console.sound) {
 		LOGP(DSENDER, LOGL_ERROR, "No sound device!\n");
 		return -EIO;

src/libmobile/sender.c

@@ -235,7 +235,7 @@ int sender_open_audio(int buffer_size, double interval)
 		}
 
 		/* open device */
-		master->audio = master->audio_open(master->device, tx_f, rx_f, am, channels, paging_frequency, master->samplerate, buffer_size, interval, (master->max_deviation) ?: 1.0, master->max_modulation, master->modulation_index);
+		master->audio = master->audio_open(SOUND_DIR_DUPLEX, master->device, tx_f, rx_f, am, channels, paging_frequency, master->samplerate, buffer_size, interval, (master->max_deviation) ?: 1.0, master->max_modulation, master->modulation_index);
 		if (!master->audio) {
 			LOGP(DSENDER, LOGL_ERROR, "No device for transceiver!\n");
 			return -EIO;

src/libmobile/sender.h

@@ -44,7 +44,7 @@ typedef struct sender {
 	/* audio */
 	void			*audio;
 	char			device[64];		/* audio device name (alsa or sdr) */
-	void			*(*audio_open)(const char *, double *, double *, int *, int, double, int, int, double, double, double, double);
+	void			*(*audio_open)(int, const char *, double *, double *, int *, int, double, int, int, double, double, double, double);
 	int 			(*audio_start)(void *);
 	void 			(*audio_close)(void *);
 	int			(*audio_write)(void *, sample_t **, uint8_t **, int, enum paging_signal *, int *, int);

src/libsdr/sdr.c

@@ -137,7 +137,7 @@ static void show_spectrum(const char *direction, double halfbandwidth, double ce
 		LOGP(DSDR, LOGL_INFO, "Frequency P = %.4f MHz (Paging Frequency)\n", paging_frequency / 1e6);
 }
 
-void *sdr_open(const char __attribute__((__unused__)) *device, 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)
+void *sdr_open(int __attribute__((__unused__)) direction, const char __attribute__((__unused__)) *device, 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)
 {
 	sdr_t *sdr;
 	int threads = 1, oversample = 1; /* always use threads */

src/libsdr/sdr.h

@@ -2,7 +2,7 @@
 enum paging_signal;
 
 int sdr_start(void *inst);
-void *sdr_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);
+void *sdr_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);
 void sdr_close(void *inst);
 int sdr_write(void *inst, sample_t **samples, uint8_t **power, int num, enum paging_signal *paging_signal, int *on, int channels);
 int sdr_read(void *inst, sample_t **samples, int num, int channels, double *rf_level_db);

src/libsound/sound.h

@@ -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

@@ -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 */
@@ -125,17 +126,22 @@ error:
 
 static int dev_open(sound_t *sound)
 {
-	int rc, rc_rec, rc_play;
+	int rc, rc_rec = 0, rc_play = 0;
 
+	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);
-	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 (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;
 
+	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");
@@ -147,6 +153,14 @@ static int dev_open(sound_t *sound)
 		}
 		LOGP(DSOUND, LOGL_DEBUG, "Playback with %d channels.\n", sound->pchannels);
 
+		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;
+		}
+	}
+
+	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");
@@ -158,17 +172,12 @@ static int dev_open(sound_t *sound)
 		}
 		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->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)

src/magnetic/main.c

@@ -268,7 +268,7 @@ inval_number:
 
 #ifdef HAVE_ALSA
 	/* open audio device */
-	sound = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
+	sound = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 0.0, 2.0);
 	if (!sound) {
 		rc = -EIO;
 		LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");

src/mate/matesimulator.c

@@ -272,7 +272,7 @@ int main(int argc, char *argv[])
 
 #ifdef HAVE_ALSA
 	/* init sound */
-	audio = sound_open(dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
+	audio = sound_open(SOUND_DIR_PLAY, dsp_audiodev, NULL, NULL, NULL, 1, 0.0, dsp_samplerate, buffer_size, 1.0, 1.0, 4000.0, 2.0);
 	if (!audio) {
 		LOGP(DDSP, LOGL_ERROR, "No sound device!\n");
 		goto exit;

src/radio/main.c

@@ -396,7 +396,7 @@ int main(int argc, char *argv[])
 	tx_frequencies[0] = frequency;
 	rx_frequencies[0] = frequency;
 	am[0] = 0;
-	sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
+	sdr = sdr_open(0, NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
 	if (!sdr)
 		goto error;
 	sdr_start(sdr);

src/radio/radio.c

@@ -100,7 +100,7 @@ int radio_init(radio_t *radio, int buffer_size, int samplerate, double frequency
 		/* open audio device */
 		radio->tx_audio_samplerate = 48000;
 		radio->tx_audio_channels = (stereo) ? 2 : 1;
-		radio->tx_sound = sound_open(tx_audiodev, NULL, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
+		radio->tx_sound = sound_open(SOUND_DIR_PLAY, tx_audiodev, NULL, NULL, NULL, radio->tx_audio_channels, 0.0, radio->tx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
 		if (!radio->tx_sound) {
 			rc = -EIO;
 			LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");
@@ -164,10 +164,7 @@ int radio_init(radio_t *radio, int buffer_size, int samplerate, double frequency
 		radio->rx_audio_samplerate = 48000;
 		radio->rx_audio_channels = (stereo) ? 2 : 1;
 		/* check if we use same device */
-		if (radio->tx_sound && !strcmp(tx_audiodev, rx_audiodev))
+		radio->rx_sound = sound_open(SOUND_DIR_REC, rx_audiodev, NULL, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
-			radio->rx_sound = radio->tx_sound;
-		else
-			radio->rx_sound = sound_open(rx_audiodev, NULL, NULL, NULL, radio->rx_audio_channels, 0.0, radio->rx_audio_samplerate, radio->buffer_size, 1.0, 1.0, 0.0, 2.0);
 		if (!radio->rx_sound) {
 			rc = -EIO;
 			LOGP(DRADIO, LOGL_ERROR, "Failed to open sound device!\n");

src/tv/main.c

@@ -342,7 +342,7 @@ static void tx_bas(sample_t *sample_bas, __attribute__((__unused__)) sample_t *s
 		tx_frequencies[0] = frequency;
 		rx_frequencies[0] = frequency;
 		am[0] = 0;
-		sdr = sdr_open(NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
+		sdr = sdr_open(0, NULL, tx_frequencies, rx_frequencies, am, 0, 0.0, dsp_samplerate, buffer_size, 1.0, 0.0, 0.0, 0.0);
 		if (!sdr)
 			goto error;
 		sdr_start(sdr);