Rework on audio buffer management

Use function to get samples to be sent to fill audio buffers to a
level. This replaces the function that only shows how much data
is in the buffer.

This way the function itself can control how much data will be
sent.

src/common/call.c

@@ -681,15 +681,14 @@ void process_call(int c)
 	int count;
 	int rc;
 
-	count = sound_get_inbuffer(call.sound);
+	count = sound_get_tosend(call.sound, call.latspl);
 	if (count < 0) {
 		PDEBUG(DSENDER, DEBUG_ERROR, "Failed to get samples in buffer (rc = %d)!\n", count);
 		if (count == -EPIPE)
 			PDEBUG(DSENDER, DEBUG_ERROR, "Trying to recover.\n");
 		return;
 	}
-	if (count < call.latspl) {
+	if (count > 0) {
-		count = call.latspl - count;
 		int16_t spl[count + 10]; /* more than enough, count will be reduced by scaling with factor */
 		switch(call.state) {
 		case CALL_ALERTING:

src/common/sdr.c

@@ -234,10 +234,12 @@ int sdr_start(void __attribute__((__unused__)) *inst)
 //	sdr_t *sdr = (sdr_t *)inst;
 
 #ifdef HAVE_UHD
-	return uhd_start();
+	if (sdr_use_uhd)
+		return uhd_start();
 #endif
 #ifdef HAVE_SOAPY
-	return soapy_start();
+	if (sdr_use_soapy)
+		return soapy_start();
 #endif
 	return -EINVAL;
 }
@@ -356,18 +358,18 @@ int sdr_read(void *inst, sample_t **samples, int num, int channels)
 }
 
 /* how many delay (in audio sample duration) do we have in the buffer */
-int sdr_get_inbuffer(void __attribute__((__unused__)) *inst)
+int sdr_get_tosend(void __attribute__((__unused__)) *inst, int latspl)
 {
 //	sdr_t *sdr = (sdr_t *)inst;
 	int count = 0;
 
 #ifdef HAVE_UHD
 	if (sdr_use_uhd)
-		count = uhd_get_inbuffer();
+		count = uhd_get_tosend(latspl);
 #endif
 #ifdef HAVE_SOAPY
 	if (sdr_use_soapy)
-		count = soapy_get_inbuffer();
+		count = soapy_get_tosend(latspl);
 #endif
 	if (count < 0)
 		return count;

src/common/sdr.h

@@ -5,5 +5,5 @@ void *sdr_open(const char *audiodev, double *tx_frequency, double *rx_frequency,
 void sdr_close(void *inst);
 int sdr_write(void *inst, sample_t **samples, int num, enum paging_signal *paging_signal, int *on, int channels);
 int sdr_read(void *inst, sample_t **samples, int num, int channels);
-int sdr_get_inbuffer(void *inst);
+int sdr_get_tosend(void *inst, int latspl);
 

src/common/sender.c

@@ -101,7 +101,7 @@ int sender_create(sender_t *sender, int kanal, double sendefrequenz, double empf
 			sender->audio_close = sdr_close;
 			sender->audio_read = sdr_read;
 			sender->audio_write = sdr_write;
-			sender->audio_get_inbuffer = sdr_get_inbuffer;
+			sender->audio_get_tosend = sdr_get_tosend;
 		} else
 #endif
 		{
@@ -110,7 +110,7 @@ int sender_create(sender_t *sender, int kanal, double sendefrequenz, double empf
 			sender->audio_close = sound_close;
 			sender->audio_read = sound_read;
 			sender->audio_write = sound_write;
-			sender->audio_get_inbuffer = sound_get_inbuffer;
+			sender->audio_get_tosend = sound_get_tosend;
 		}
 	}
 
@@ -282,12 +282,8 @@ void process_sender_audio(sender_t *sender, int *quit, int latspl)
 		samples[i] = buff[i];
 	}
 
-	count = sender->audio_get_inbuffer(sender->audio);
+	count = sender->audio_get_tosend(sender->audio, latspl);
 	if (count < 0) {
-		/* special case when the device is not yet ready to transmit packets */
-		if (count == -EAGAIN) {
-			goto transmit_later;
-		}
 		PDEBUG(DSENDER, DEBUG_ERROR, "Failed to get samples in buffer (rc = %d)!\n", count);
 		if (count == -EPIPE) {
 			if (cant_recover) {
@@ -300,8 +296,7 @@ cant_recover:
 		}
 		return;
 	}
-	if (count < latspl) {
+	if (count > 0) {
-		count = latspl - count;
 		/* loop through all channels */
 		for (i = 0, inst = sender; inst; i++, inst = inst->slave) {
 			/* load TX data from audio loop or from sender instance */
@@ -338,7 +333,6 @@ cant_recover:
 			return;
 		}
 	}
-transmit_later:
 
 	count = sender->audio_read(sender->audio, samples, latspl, num_chan);
 	if (count < 0) {

src/common/sender.h

@@ -46,7 +46,7 @@ typedef struct sender {
 	void 			(*audio_close)(void *);
 	int			(*audio_write)(void *, sample_t **, int, enum paging_signal *, int *, int);
 	int			(*audio_read)(void *, sample_t **, int, int);
-	int			(*audio_get_inbuffer)(void *);
+	int			(*audio_get_tosend)(void *, int);
 	int			samplerate;
 	samplerate_t		srstate;		/* sample rate conversion state */
 	double			rx_gain;		/* factor of level to apply on rx samples */

src/common/soapy.c

@@ -262,25 +262,26 @@ int soapy_receive(float *buff, int max)
 	return got;
 }
 
-/* estimate current unsent number of samples */
+/* estimate number of samples that can be sent */
-int soapy_get_inbuffer(void)
+int soapy_get_tosend(int latspl)
 {
-	long long advance;
+	int tosend;
 
 	/* we need the rx time stamp to determine how much data is already sent in advance */
 	if (rx_count == 0)
-		return -EAGAIN;
+		return 0;
 
 	/* if we have not yet sent any data, we set initial tx time stamp */
 	if (tx_count == 0)
-		tx_count = rx_count;
+		tx_count = rx_count + latspl;
 
 	/* we check how advance our transmitted time stamp is */
-	advance = tx_count - rx_count;
+	tosend = latspl - (tx_count - rx_count);
 	/* in case of underrun: */
-	if (advance < 0)
+	if (tosend < 0)
-		advance = 0;
+		tosend = 0;
 
-	return advance;
+	return tosend;
 }
 
+

src/common/soapy.h

@@ -4,5 +4,5 @@ int soapy_start(void);
 void soapy_close(void);
 int soapy_send(float *buff, int num);
 int soapy_receive(float *buff, int max);
-int soapy_get_inbuffer(void);
+int soapy_get_tosend(int latspl);
 

src/common/sound.h

@@ -6,5 +6,5 @@ int sound_start(void *inst);
 void sound_close(void *inst);
 int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *paging_signal, int *on, int channels);
 int sound_read(void *inst, sample_t **samples, int num, int channels);
-int sound_get_inbuffer(void *inst);
+int sound_get_tosend(void *inst, int latspl);
 

src/common/sound_alsa.c

@@ -323,8 +323,10 @@ int sound_write(void *inst, sample_t **samples, int num, enum paging_signal *pag
 
 	if (rc < 0) {
 		PDEBUG(DSOUND, DEBUG_ERROR, "failed to write audio to interface (%s)\n", snd_strerror(rc));
-		if (rc == -EPIPE)
+		if (rc == -EPIPE) {
 			sound_prepare(sound);
+			sound_start(sound);
+		}
 		return rc;
 	}
 
@@ -363,8 +365,10 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
 			return 0;
 		PDEBUG(DSOUND, DEBUG_ERROR, "failed to read audio from interface (%s)\n", snd_strerror(rc));
 		/* recover read */
-		if (rc == -EPIPE)
+		if (rc == -EPIPE) {
 			sound_prepare(sound);
+			sound_start(sound);
+		}
 		return rc;
 	}
 
@@ -391,14 +395,15 @@ int sound_read(void *inst, sample_t **samples, int num, int channels)
 }
 
 /* 
- * get playback buffer fill
+ * get playback buffer space
  *
- * return number of frames */
+ * return number of samples to be sent */
-int sound_get_inbuffer(void *inst)
+int sound_get_tosend(void *inst, int latspl)
 {
 	sound_t *sound = (sound_t *)inst;
 	int rc;
 	snd_pcm_sframes_t delay;
+	int tosend;
 
 	rc = snd_pcm_delay(sound->phandle, &delay);
 	if (rc < 0) {
@@ -406,12 +411,15 @@ int sound_get_inbuffer(void *inst)
 			PDEBUG(DSOUND, DEBUG_ERROR, "Buffer underrun: Please use higher latency 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)
+		if (rc == -EPIPE) {
 			sound_prepare(sound);
+			sound_start(sound);
+		}
 		return rc;
 	}
 
-	return delay;
+	tosend = latspl - delay;
+	return tosend;
 }
 
 int sound_is_stereo_capture(void *inst)

src/common/uhd.c

@@ -379,19 +379,24 @@ int uhd_receive(float *buff, int max)
 	return got;
 }
 
-/* estimate current unsent number of samples */
+/* estimate number of samples that can be sent */
-int uhd_get_inbuffer(void)
+int uhd_get_tosend(int latspl)
 {
 	double advance;
+	int tosend;
 
 	/* we need the rx time stamp to determine how much data is already sent in advance */
 	if (rx_time_secs == 0 && rx_time_fract_sec == 0.0)
-		return -EAGAIN;
+		return 0;
 
 	/* if we have not yet sent any data, we set initial tx time stamp */
 	if (tx_time_secs == 0 && tx_time_fract_sec == 0.0) {
 		tx_time_secs = rx_time_secs;
-		tx_time_fract_sec = rx_time_fract_sec;
+		tx_time_fract_sec = rx_time_fract_sec + (double)latspl / samplerate;
+		if (tx_time_fract_sec >= 1.0) {
+			tx_time_fract_sec -= 1.0;
+			tx_time_secs++;
+		}
 	}
 
 	/* we check how advance our transmitted time stamp is */
@@ -399,7 +404,10 @@ int uhd_get_inbuffer(void)
 	/* in case of underrun: */
 	if (advance < 0)
 		advance = 0;
+	tosend = latspl - (int)(advance * samplerate);
+	if (tosend < 0)
+		tosend = 0;
 
-	return (int)(advance * samplerate);
+	return tosend;
 }
 

src/common/uhd.h

@@ -4,5 +4,5 @@ int uhd_start(void);
 void uhd_close(void);
 int uhd_send(float *buff, int num);
 int uhd_receive(float *buff, int max);
-int uhd_get_inbuffer(void);
+int uhd_get_tosend(int latspl);