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-rw-r--r--src/common/call.c5
-rw-r--r--src/common/sdr.c12
-rw-r--r--src/common/sdr.h2
-rw-r--r--src/common/sender.c14
-rw-r--r--src/common/sender.h2
-rw-r--r--src/common/soapy.c19
-rw-r--r--src/common/soapy.h2
-rw-r--r--src/common/sound.h2
-rw-r--r--src/common/sound_alsa.c22
-rw-r--r--src/common/uhd.c18
-rw-r--r--src/common/uhd.h2
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);
generated by cgit v1.2.3 (git 2.25.1) at 2024-04-26 23:13:11 +0000