Implementation of C-Netz (German mobile telephone system)

src/common/debug.c

@@ -38,6 +38,7 @@ struct debug_cat {
 	{ "dsp", "\033[0;31m" },
 	{ "anetz", "\033[1;34m" },
 	{ "bnetz", "\033[1;34m" },
+	{ "cnetz", "\033[1;34m" },
 	{ "nmt", "\033[1;34m" },
 	{ "frame", "\033[0;36m" },
 	{ "call", "\033[1;37m" },

src/common/debug.h

@@ -9,10 +9,11 @@
 #define DDSP		2
 #define DANETZ		3
 #define DBNETZ		4
-#define DNMT		5
-#define DFRAME		6
-#define DCALL		7
-#define DMNCC		8
+#define DCNETZ		5
+#define DNMT		6
+#define DFRAME		7
+#define DCALL		8
+#define DMNCC		9
 
 #define PDEBUG(cat, level, fmt, arg...) _printdebug(__FILE__, __FUNCTION__, __LINE__, cat, level, fmt, ## arg)
 void _printdebug(const char *file, const char *function, int line, int cat, int level, const char *fmt, ...);

src/common/filter.h

@@ -1,3 +1,5 @@
+#ifndef _FILTER_H
+#define _FILTER_H
 
 typedef struct biquad_low_pass {
 	double a0, a1, a2, b1, b2;
@@ -7,3 +9,4 @@ typedef struct biquad_low_pass {
 void biquad_init(biquad_low_pass_t *bq, double frequency, int samplerate);
 void biquad_process(biquad_low_pass_t *bq, double *samples, int length, int iterations);
 
+#endif /* _FILTER_H */

src/common/sender.c

@@ -30,6 +30,7 @@
 
 sender_t *sender_head = NULL;
 static sender_t **sender_tailp = &sender_head;
+int cant_recover = 0;
 
 /* Init transceiver instance and link to list of transceivers. */
 int sender_create(sender_t *sender, const char *sounddev, int samplerate, int pre_emphasis, int de_emphasis, const char *write_wave, const char *read_wave, int kanal, int loopback, double loss_volume, int use_pilot_signal)
@@ -139,7 +140,7 @@ static void gen_pilotton(sender_t *sender, int16_t *samples, int length)
 }
 
 /* Handle audio streaming of one transceiver. */
-void process_sender(sender_t *sender, int latspl)
+void process_sender(sender_t *sender, int *quit, int latspl)
 {
 	int16_t samples[latspl], pilot[latspl];
 	int rc, count;
@@ -147,8 +148,15 @@ void process_sender(sender_t *sender, int latspl)
 	count = sound_get_inbuffer(sender->sound);
 	if (count < 0) {
 		PDEBUG(DSENDER, DEBUG_ERROR, "Failed to get samples in buffer (rc = %d)!\n", count);
-		if (count == -EPIPE)
+		if (count == -EPIPE) {
+			if (cant_recover) {
+cant_recover:
+				PDEBUG(DSENDER, DEBUG_ERROR, "Cannot recover due to measurements, quitting!\n");
+				*quit = 1;
+				return;
+			}
 			PDEBUG(DSENDER, DEBUG_ERROR, "Trying to recover!\n");
+		}
 		return;
 	}
 	if (count < latspl) {
@@ -189,8 +197,11 @@ void process_sender(sender_t *sender, int latspl)
 		}
 		if (rc < 0) {
 			PDEBUG(DSENDER, DEBUG_ERROR, "Failed to write TX data to sound device (rc = %d)\n", rc);
-			if (rc == -EPIPE)
+			if (rc == -EPIPE) {
+				if (cant_recover)
+					goto cant_recover;
 				PDEBUG(DSENDER, DEBUG_ERROR, "Trying to recover!\n");
+			}
 			return;
 		}
 		if (sender->loopback == 1) {
@@ -204,8 +215,11 @@ void process_sender(sender_t *sender, int latspl)
 //printf("count=%d time= %.4f\n", count, (double)count * 1000 / sender->samplerate);
 	if (count < 0) {
 		PDEBUG(DSENDER, DEBUG_ERROR, "Failed to read from sound device (rc = %d)!\n", count);
-		if (count == -EPIPE)
+		if (count == -EPIPE) {
+			if (cant_recover)
+				goto cant_recover;
 			PDEBUG(DSENDER, DEBUG_ERROR, "Trying to recover!\n");
+		}
 		return;
 	}
 	if (count) {
@@ -236,7 +250,7 @@ void main_loop(int *quit, int latency)
 		sender = sender_head;
 		while (sender) {
 			latspl = sender->samplerate * latency / 1000;
-			process_sender(sender, latspl);
+			process_sender(sender, quit, latspl);
 			sender = sender->next;
 		}
 

src/common/sender.h

@@ -50,6 +50,7 @@ typedef struct sender {
 
 /* list of all senders */
 extern sender_t *sender_head;
+extern int cant_recover;
 
 int sender_create(sender_t *sender, const char *sounddev, int samplerate, int pre_emphasis, int de_emphasis, const char *write_wave, const char *read_wave, int kanal, int loopback, double loss_volume, int use_pilot_signal);
 void sender_destroy(sender_t *sender);

src/common/sound_alsa.c

@@ -249,7 +249,10 @@ int sound_get_inbuffer(void *inst)
 
 	rc = snd_pcm_delay(sound->phandle, &delay);
 	if (rc < 0) {
-		PDEBUG(DSOUND, DEBUG_ERROR, "failed to get delay from interface (%s)\n", snd_strerror(rc));
+		if (rc == -32)
+			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)
 			snd_pcm_prepare(sound->phandle);
 		return rc;