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.

11 files changed, 56 insertions, 44 deletions

diff --git a/src/common/call.c b/src/common/call.c
index f2db043..a8ac8e1 100644
--- a/src/common/call.c
+++ b/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) {
-		count = call.latspl - count;
+	if (count > 0) {
 		int16_t spl[count + 10]; /* more than enough, count will be reduced by scaling with factor */
 		switch(call.state) {
 		case CALL_ALERTING:
diff --git a/src/common/sdr.c b/src/common/sdr.c
index ea080d4..372824d 100644
--- a/src/common/sdr.c
+++ b/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;
diff --git a/src/common/sdr.h b/src/common/sdr.h
index 3d1aaca..0d02e0f 100644
--- a/src/common/sdr.h
+++ b/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);
 
diff --git a/src/common/sender.c b/src/common/sender.c
index b24db0b..055f84e 100644
--- a/src/common/sender.c
+++ b/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) {
-		count = latspl - count;
+	if (count > 0) {
 		/* 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) {
diff --git a/src/common/sender.h b/src/common/sender.h
index 658d444..b371591 100644
--- a/src/common/sender.h
+++ b/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 */
diff --git a/src/common/soapy.c b/src/common/soapy.c
index b01f586..76e213e 100644
--- a/src/common/soapy.c
+++ b/src/common/soapy.c
@@ -262,25 +262,26 @@ int soapy_receive(float *buff, int max)
 	return got;
 }
 
-/* estimate current unsent number of samples */
-int soapy_get_inbuffer(void)
+/* estimate number of samples that can be sent */
+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)
-		advance = 0;
+	if (tosend < 0)
+		tosend = 0;
 
-	return advance;
+	return tosend;
 }
 
+
diff --git a/src/common/soapy.h b/src/common/soapy.h
index 36ec304..dcfadde 100644
--- a/src/common/soapy.h
+++ b/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);
 
diff --git a/src/common/sound.h b/src/common/sound.h
index 339cea6..1054eb4 100644
--- a/src/common/sound.h
+++ b/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);
 
diff --git a/src/common/sound_alsa.c b/src/common/sound_alsa.c
index 22b1b77..752f82f 100644
--- a/src/common/sound_alsa.c
+++ b/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 */
-int sound_get_inbuffer(void *inst)
+ * return number of samples to be sent */
+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)
diff --git a/src/common/uhd.c b/src/common/uhd.c
index 15c13b9..1423703 100644
--- a/src/common/uhd.c
+++ b/src/common/uhd.c
@@ -379,19 +379,24 @@ int uhd_receive(float *buff, int max)
 	return got;
 }
 
-/* estimate current unsent number of samples */
-int uhd_get_inbuffer(void)
+/* estimate number of samples that can be sent */
+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;
 }
 
diff --git a/src/common/uhd.h b/src/common/uhd.h
index 09ae519..686d658 100644
--- a/src/common/uhd.h
+++ b/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);