Rename dbm0 (level) to speech (level)

In mobile communications we use speech level instead of milliwatts.
The deviation (FM) is always based on speech level, not on 1 milliwatt.

16 files changed, 105 insertions, 104 deletions

diff --git a/src/amps/dsp.c b/src/amps/dsp.c
index f8f7c14..dc99c2f 100644
--- a/src/amps/dsp.c
+++ b/src/amps/dsp.c
@@ -101,18 +101,18 @@
 
 #define AMPS_MAX_DEVIATION	8000.0
 #define AMPS_MAX_MODULATION	10000.0
-#define AMPS_DBM0_DEVIATION	2900.0  /* deviation of dBm0 at 1 kHz */
-#define AMPS_FSK_DEVIATION	(8000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
-#define AMPS_SAT_DEVIATION	(2000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
-#define AMPS_MAX_DISPLAY	(10000.0 / AMPS_DBM0_DEVIATION)	/* no emphasis */
+#define AMPS_SPEECH_DEVIATION	2900.0  /* deviation of speech at 1 kHz */
+#define AMPS_FSK_DEVIATION	(8000.0 / AMPS_SPEECH_DEVIATION)	/* no emphasis */
+#define AMPS_SAT_DEVIATION	(2000.0 / AMPS_SPEECH_DEVIATION)	/* no emphasis */
+#define AMPS_MAX_DISPLAY	(10000.0 / AMPS_SPEECH_DEVIATION)	/* no emphasis */
 #define AMPS_BITRATE		10000
 /* for some reason, 4000 Hz deviation works better */
-#define TACS_DBM0_DEVIATION	4000.0  /* 2300 Hz deviation at 1 kHz (according to panasonic manual) */
+#define TACS_SPEECH_DEVIATION	4000.0  /* 2300 Hz deviation at 1 kHz (according to panasonic manual) */
 #define TACS_MAX_DEVIATION	6400.0	/* (according to texas instruments and other sources) */
 #define TACS_MAX_MODULATION	9500.0	/* (according to panasonic manual) */
-#define TACS_FSK_DEVIATION	(6400.0 / TACS_DBM0_DEVIATION)	/* no emphasis */
-#define TACS_SAT_DEVIATION	(1700.0 / TACS_DBM0_DEVIATION)	/* no emphasis (panasonic / TI) */
-#define TACS_MAX_DISPLAY	(8000.0 / TACS_DBM0_DEVIATION)	/* no emphasis */
+#define TACS_FSK_DEVIATION	(6400.0 / TACS_SPEECH_DEVIATION)	/* no emphasis */
+#define TACS_SAT_DEVIATION	(1700.0 / TACS_SPEECH_DEVIATION)	/* no emphasis (panasonic / TI) */
+#define TACS_MAX_DISPLAY	(8000.0 / TACS_SPEECH_DEVIATION)	/* no emphasis */
 #define TACS_BITRATE		8000
 #define SAT_DURATION		0.05	/* duration of SAT signal measurement */
 #define SAT_QUALITY		0.85	/* quality needed to detect SAT signal */
@@ -202,7 +202,7 @@ int dsp_init_sender(amps_t *amps, int tolerant)
 	sender_set_fm(&amps->sender,
 		(!tacs) ? AMPS_MAX_DEVIATION : TACS_MAX_DEVIATION,
 		(!tacs) ? AMPS_MAX_MODULATION : TACS_MAX_MODULATION,
-		(!tacs) ? AMPS_DBM0_DEVIATION : TACS_DBM0_DEVIATION,
+		(!tacs) ? AMPS_SPEECH_DEVIATION : TACS_SPEECH_DEVIATION,
 		(!tacs) ? AMPS_MAX_DISPLAY : TACS_MAX_DISPLAY);
 
 	if (amps->sender.samplerate < 96000) {
diff --git a/src/anetz/dsp.c b/src/anetz/dsp.c
index 14af0be..40144a5 100644
--- a/src/anetz/dsp.c
+++ b/src/anetz/dsp.c
@@ -35,15 +35,15 @@
 #define PI		3.1415927
 
 /* signaling */
-#define MAX_DEVIATION	15000.0
-#define MAX_MODULATION	4000.0
-#define DBM0_DEVIATION	10500.0	/* deviation of dBm0 at 1 kHz */
-#define TX_PEAK_TONE	(10500.0 / DBM0_DEVIATION)	/* 10.5 kHz, no emphasis */
-#define TX_PEAK_PAGE	(15000.0 / DBM0_DEVIATION)	/* 15 kHz, no emphasis */
-#define MAX_DISPLAY	(15000.0 / DBM0_DEVIATION)	/* 15 kHz, no emphasis */
-#define CHUNK_DURATION	0.010	/* 10 ms */
-#define TONE_THRESHOLD	0.05
-#define QUAL_THRESHOLD	0.5
+#define MAX_DEVIATION		15000.0
+#define MAX_MODULATION		4000.0
+#define SPEECH_DEVIATION	10500.0	/* deviation of speech at 1 kHz */
+#define TX_PEAK_TONE		(10500.0 / SPEECH_DEVIATION)	/* 10.5 kHz, no emphasis */
+#define TX_PEAK_PAGE		(15000.0 / SPEECH_DEVIATION)	/* 15 kHz, no emphasis */
+#define MAX_DISPLAY		(15000.0 / SPEECH_DEVIATION)	/* 15 kHz, no emphasis */
+#define CHUNK_DURATION		0.010	/* 10 ms */
+#define TONE_THRESHOLD		0.05
+#define QUAL_THRESHOLD		0.5
 
 // FIXME: how long until we detect a tone?
 #define TONE_DETECT_TH	8	/* chunk intervals to detect continuous tone */
@@ -89,7 +89,7 @@ int dsp_init_sender(anetz_t *anetz, double page_gain, int page_sequence, double
 	squelch_init(&anetz->squelch, anetz->sender.kanal, squelch_db, MUTE_TIME, LOSS_TIME);
 
 	/* set modulation parameters */
-	sender_set_fm(&anetz->sender, MAX_DEVIATION * page_gain, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
+	sender_set_fm(&anetz->sender, MAX_DEVIATION * page_gain, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
 
 	anetz->page_gain = page_gain;
 	anetz->page_sequence = page_sequence;
diff --git a/src/bnetz/dsp.c b/src/bnetz/dsp.c
index 7315d26..c360771 100644
--- a/src/bnetz/dsp.c
+++ b/src/bnetz/dsp.c
@@ -37,25 +37,25 @@
  *
  * At 2000 Hz the deviation shall be 4 kHz, so with emphasis the deviation
  * at 1000 Hz would be theoretically 2 kHz. This is factor 0.714 below
- * 2.8 kHz deviation we want at dBm0.
+ * 2.8 kHz deviation we want at speech level.
  */
 
 /* signaling */
-#define MAX_DEVIATION	4000.0
-#define MAX_MODULATION	3000.0
-#define DBM0_DEVIATION	2800.0	/* deviation of dBm0 at 1 kHz */
-#define TX_PEAK_FSK	(4000.0 / 2000.0 * 1000.0 / DBM0_DEVIATION)
-#define TX_PEAK_METER	(2000.0 / 2900.0 * 1000.0 / DBM0_DEVIATION) /* FIXME: what is the metering pulse deviation??? we use half of the 4kHz deviation, so we can still use -6dB of the speech level */
-#define DAMPEN_METER	0.5	/* use -6dB to dampen speech while sending metering pulse (according to FTZ 1727 Pfl 32 Clause 3.2.6.6.5) */
-#define MAX_DISPLAY	1.4	/* something above dBm0 */
-#define BIT_RATE	100.0
-#define BIT_ADJUST	0.5	/* full adjustment on bit change */
-#define F0		2070.0
-#define F1		1950.0
-#define METERING_HZ	2900	/* metering pulse frequency */
-#define TONE_DETECT_CNT	7	/* 70 milliseconds to detect continuous tone */
-#define TONE_LOST_CNT	14	/* we use twice of the detect time, so we bridge a loss of "TONE_DETECT_CNT duration" */
-#define	TONE_STDDEV_TH	0.2	/* threshold of bad quality (standard deviation) to reject tone */
+#define MAX_DEVIATION		4000.0
+#define MAX_MODULATION		3000.0
+#define SPEECH_DEVIATION	2800.0	/* deviation of speech at 1 kHz */
+#define TX_PEAK_FSK		(4000.0 / 2000.0 * 1000.0 / SPEECH_DEVIATION)
+#define TX_PEAK_METER		(2000.0 / 2900.0 * 1000.0 / SPEECH_DEVIATION) /* FIXME: what is the metering pulse deviation??? we use half of the 4kHz deviation, so we can still use -6dB of the speech level */
+#define DAMPEN_METER		0.5	/* use -6dB to dampen speech while sending metering pulse (according to FTZ 1727 Pfl 32 Clause 3.2.6.6.5) */
+#define MAX_DISPLAY		1.4	/* something above speech level */
+#define BIT_RATE		100.0
+#define BIT_ADJUST		0.5	/* full adjustment on bit change */
+#define F0			2070.0
+#define F1			1950.0
+#define METERING_HZ		2900	/* metering pulse frequency */
+#define TONE_DETECT_CNT		7	/* 70 milliseconds to detect continuous tone */
+#define TONE_LOST_CNT		14	/* we use twice of the detect time, so we bridge a loss of "TONE_DETECT_CNT duration" */
+#define	TONE_STDDEV_TH		0.2	/* threshold of bad quality (standard deviation) to reject tone */
 
 /* carrier loss detection */
 #define MUTE_TIME	0.1	/* time to mute after loosing signal */
@@ -91,7 +91,7 @@ int dsp_init_sender(bnetz_t *bnetz, double squelch_db)
 	squelch_init(&bnetz->squelch, bnetz->sender.kanal, squelch_db, MUTE_TIME, LOSS_TIME);
 
 	/* set modulation parameters */
-	sender_set_fm(&bnetz->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
+	sender_set_fm(&bnetz->sender, MAX_DEVIATION, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
 
 	PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 2000 Hz)\n", TX_PEAK_FSK, 4.0);
 
diff --git a/src/cnetz/cnetz.c b/src/cnetz/cnetz.c
index 3c2ac45..990d48d 100644
--- a/src/cnetz/cnetz.c
+++ b/src/cnetz/cnetz.c
@@ -247,7 +247,7 @@ int cnetz_init(void)
 }
 
 /* Create transceiver instance and link to a list. */
-int cnetz_create(const char *kanal, enum cnetz_chan_type chan_type, const char *audiodev, int use_sdr, enum demod_type demod, int samplerate, double rx_gain, int challenge_valid, uint64_t challenge, int response_valid, uint64_t response, int warteschlange, int metering, double dbm0_deviation, int ms_power, int measure_speed, double clock_speed[2], int polarity, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback)
+int cnetz_create(const char *kanal, enum cnetz_chan_type chan_type, const char *audiodev, int use_sdr, enum demod_type demod, int samplerate, double rx_gain, int challenge_valid, uint64_t challenge, int response_valid, uint64_t response, int warteschlange, int metering, double speech_deviation, int ms_power, int measure_speed, double clock_speed[2], int polarity, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback)
 {
 	sender_t *sender;
 	cnetz_t *cnetz;
@@ -321,7 +321,7 @@ int cnetz_create(const char *kanal, enum cnetz_chan_type chan_type, const char *
 #endif
 
 	/* init audio processing */
-	rc = dsp_init_sender(cnetz, measure_speed, clock_speed, demod, dbm0_deviation);
+	rc = dsp_init_sender(cnetz, measure_speed, clock_speed, demod, speech_deviation);
 	if (rc < 0) {
 		PDEBUG(DCNETZ, DEBUG_ERROR, "Failed to init signal processing!\n");
 		goto error;
diff --git a/src/cnetz/cnetz.h b/src/cnetz/cnetz.h
index 25e1f03..20d04c0 100644
--- a/src/cnetz/cnetz.h
+++ b/src/cnetz/cnetz.h
@@ -138,7 +138,7 @@ int cnetz_channel_by_short_name(const char *short_name);
 const char *chan_type_short_name(enum cnetz_chan_type chan_type);
 const char *chan_type_long_name(enum cnetz_chan_type chan_type);
 int cnetz_init(void);
-int cnetz_create(const char *kanal, enum cnetz_chan_type chan_type, const char *audiodev, int use_sdr, enum demod_type demod, int samplerate, double rx_gain, int challenge_valid, uint64_t challenge, int response_valid, uint64_t response, int warteschlange, int metering, double dbm0_deviation, int ms_power, int measure_speed, double clock_speed[2], int polarity, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback);
+int cnetz_create(const char *kanal, enum cnetz_chan_type chan_type, const char *audiodev, int use_sdr, enum demod_type demod, int samplerate, double rx_gain, int challenge_valid, uint64_t challenge, int response_valid, uint64_t response, int warteschlange, int metering, double speech_deviation, int ms_power, int measure_speed, double clock_speed[2], int polarity, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback);
 void cnetz_destroy(sender_t *sender);
 void cnetz_go_idle(cnetz_t *cnetz);
 void cnetz_sync_frame(cnetz_t *cnetz, double sync, int ts);
diff --git a/src/cnetz/dsp.c b/src/cnetz/dsp.c
index ec39303..91f967c 100644
--- a/src/cnetz/dsp.c
+++ b/src/cnetz/dsp.c
@@ -42,8 +42,8 @@
 
 #define MAX_DEVIATION	4000.0
 #define MAX_MODULATION	3000.0
-#define FSK_DEVIATION	(2500.0 / dbm0_deviation) /* no emphasis */
-#define MAX_DISPLAY	1.4	/* something above dBm0, no emphasis */
+#define FSK_DEVIATION	(2500.0 / speech_deviation) /* no emphasis */
+#define MAX_DISPLAY	1.4	/* something above speech level, no emphasis */
 #define BITRATE		5280.0	/* bits per second */
 #define BLOCK_BITS	198	/* duration of one time slot including pause at beginning and end */
 #define CUT_OFF_OFFSET	300.0	/* cut off frequency for offset filter (level correction between subsequent audio chunks) */
@@ -80,7 +80,7 @@ static void dsp_init_ramp(cnetz_t *cnetz)
 }
 
 /* Init transceiver instance. */
-int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], enum demod_type demod, double dbm0_deviation)
+int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], enum demod_type demod, double speech_deviation)
 {
 	int rc = 0;
 	double size;
@@ -89,7 +89,7 @@ int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], en
 	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init FSK for 'Sender'.\n");
 
 	/* set modulation parameters */
-	sender_set_fm(&cnetz->sender, MAX_DEVIATION, MAX_MODULATION, dbm0_deviation, MAX_DISPLAY);
+	sender_set_fm(&cnetz->sender, MAX_DEVIATION, MAX_MODULATION, speech_deviation, MAX_DISPLAY);
 
 	if (measure_speed) {
 		cnetz->measure_speed = measure_speed;
diff --git a/src/cnetz/dsp.h b/src/cnetz/dsp.h
index 34e8f38..f5e3506 100644
--- a/src/cnetz/dsp.h
+++ b/src/cnetz/dsp.h
@@ -1,6 +1,6 @@
 
 void dsp_init(void);
-int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], enum demod_type demod, double dbm0_deviation);
+int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], enum demod_type demod, double speech_deviation);
 void dsp_cleanup_sender(cnetz_t *cnetz);
 void calc_clock_speed(cnetz_t *cnetz, double samples, int tx, int result);
 void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count);
diff --git a/src/cnetz/main.c b/src/cnetz/main.c
index 153d1f5..518151a 100644
--- a/src/cnetz/main.c
+++ b/src/cnetz/main.c
@@ -73,7 +73,7 @@ int8_t	futln_sperre_start = -1; /* no blocking */
 int8_t	futln_sperre_end = -1; /* no range */
 enum demod_type demod = FSK_DEMOD_AUTO;
 int metering = 20;
-double dbm0_deviation = 4000.0; /* best results with all my equipment */
+double speech_deviation = 4000.0; /* best results with all my equipment */
 
 void print_help(const char *arg0)
 {
@@ -118,7 +118,7 @@ void print_help(const char *arg0)
 	printf(" -V --voice-deviation <2400..4000 Hz>\n");
 	printf("        It is unclear what the actual voice deviation is. Please decrease, if\n");
 	printf("        mobile's microphone is too loud and speaker is too quiet.\n");
-	printf("        (default = %.0f)\n", dbm0_deviation);
+	printf("        (default = %.0f)\n", speech_deviation);
 	printf(" -S --sysinfo fuz-nat=<nat>\n");
 	printf("        Set country ID of base station. All IDs were used inside Germany only.\n");
 	printf("        (default = %d)\n", fuz_nat);
@@ -327,7 +327,7 @@ static int handle_options(int short_option, int argi, char **argv)
 		metering = atoi(argv[argi]);
 		break;
 	case 'V':
-		dbm0_deviation = atoi_limit(argv[argi], 2400, 4000);
+		speech_deviation = atoi_limit(argv[argi], 2400, 4000);
 		break;
 	case 'S':
 		p = strchr(argv[argi], '=');
@@ -621,7 +621,7 @@ int main(int argc, char *argv[])
 
 	/* create transceiver instance */
 	for (i = 0; i < num_kanal; i++) {
-		rc = cnetz_create(kanal[i], chan_type[i], audiodev[i], use_sdr, demod, samplerate, rx_gain, challenge_valid, challenge, response_valid, response, warteschlange, metering, dbm0_deviation, ms_power, (i == 0) ? measure_speed : 0, clock_speed, polarity, do_pre_emphasis, do_de_emphasis, write_rx_wave, write_tx_wave, read_rx_wave, read_tx_wave, loopback);
+		rc = cnetz_create(kanal[i], chan_type[i], audiodev[i], use_sdr, demod, samplerate, rx_gain, challenge_valid, challenge, response_valid, response, warteschlange, metering, speech_deviation, ms_power, (i == 0) ? measure_speed : 0, clock_speed, polarity, do_pre_emphasis, do_de_emphasis, write_rx_wave, write_tx_wave, read_rx_wave, read_tx_wave, loopback);
 		if (rc < 0) {
 			fprintf(stderr, "Failed to create \"Sender\" instance. Quitting!\n");
 			goto fail;
diff --git a/src/imts/dsp.c b/src/imts/dsp.c
index 52b388f..c8a0029 100644
--- a/src/imts/dsp.c
+++ b/src/imts/dsp.c
@@ -38,22 +38,22 @@
 #define PI		3.1415927
 
 /* signaling */
-#define MAX_DEVIATION	7000.0				/* signaling tone plus some extra to calibrate */
-#define MAX_MODULATION	3000.0				/* FIXME */
-#define DBM0_DEVIATION	2500.0				/* deviation of dBm0 (with emphasis) */
-#define TX_PEAK_TONE	(5000.0 / DBM0_DEVIATION)	/* signaling tone level (5khz, no emphasis) */
-#define RX_MIN_AMPL	0.25				/* FIXME: Minimum level to detect tone */
+#define MAX_DEVIATION		7000.0				/* signaling tone plus some extra to calibrate */
+#define MAX_MODULATION		3000.0				/* FIXME */
+#define SPEECH_DEVIATION	2500.0				/* deviation of speech (with emphasis) */
+#define TX_PEAK_TONE		(5000.0 / SPEECH_DEVIATION)	/* signaling tone level (5khz, no emphasis) */
+#define RX_MIN_AMPL		0.25				/* FIXME: Minimum level to detect tone */
 /* Note that 75 is half of the distance between two tones (2000 and 2150 Hz)
  * An error of more than 50 causes too much toggeling between two tones,
  * less would take too long to detect the tone and maybe not detect it, if
  * it is too far off the expected frequency.
  */
-#define RX_MIN_FREQ	50.0				/* minimum frequency error to detect tone */
-#define MAX_DISPLAY	(MAX_DEVIATION / DBM0_DEVIATION)/* as much as MAX_DEVIATION */
+#define RX_MIN_FREQ		50.0				/* minimum frequency error to detect tone */
+#define MAX_DISPLAY		(MAX_DEVIATION / SPEECH_DEVIATION)/* as much as MAX_DEVIATION */
 /* Note that FILTER_BW / SUSTAIN and QUAL_TIME sum up and should not exeed minimum tone length */
-#define RX_FILTER_BW	100.0				/* amplitude filter (causes delay) */
-#define RX_SUSTAIN	0.010				/* how long a tone must sustain until detected (causes delay) */
-#define RX_QUAL_TIME	0.005				/* how long a quality measurement lasts after detecting a tone */
+#define RX_FILTER_BW		100.0				/* amplitude filter (causes delay) */
+#define RX_SUSTAIN		0.010				/* how long a tone must sustain until detected (causes delay) */
+#define RX_QUAL_TIME		0.005				/* how long a quality measurement lasts after detecting a tone */
 
 /* carrier loss detection */
 #define MUTE_TIME	0.1	/* time to mute after loosing signal */
@@ -131,7 +131,7 @@ int dsp_init_transceiver(imts_t *imts, double squelch_db, int ptt)
 	squelch_init(&imts->squelch, imts->sender.kanal, squelch_db, MUTE_TIME, LOSS_TIME);
 
 	/* set modulation parameters */
-	sender_set_fm(&imts->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
+	sender_set_fm(&imts->sender, MAX_DEVIATION, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
 
 	/* init FM demodulator for tone detection */
 	if (imts->mode == MODE_IMTS) {
diff --git a/src/jolly/dsp.c b/src/jolly/dsp.c
index 94231fd..dfd694b 100644
--- a/src/jolly/dsp.c
+++ b/src/jolly/dsp.c
@@ -37,15 +37,15 @@
 #define db2level(db)	pow(10, (double)db / 20.0)
 
 /* transceiver parameters */
-#define MAX_DEVIATION	5000.0	/* deviation of signal */
-#define MAX_MODULATION  4000.0	/* frequency spectrum of signal */
-#define DBM0_DEVIATION	3000.0	/* deviation of dBm0 at 1 kHz (generally used with 25 kHz channel spacing) */
-#define	MAX_DISPLAY	1.0	/* maximum level to display */
-#define TX_INFO_TONE	1.0	/* Level of tone relative to dBm0 (each component) */
-#define TX_ACK_TONE	0.1	/* Level of tone relative to dBm0 */
-#define INFO_TONE_F1	640.0
-#define INFO_TONE_F2	670.0
-#define ACK_TONE	1000.0
+#define MAX_DEVIATION		5000.0	/* deviation of signal */
+#define MAX_MODULATION		4000.0	/* frequency spectrum of signal */
+#define SPEECH_DEVIATION	3000.0	/* deviation of speech at 1 kHz (generally used with 25 kHz channel spacing) */
+#define	MAX_DISPLAY		1.0	/* maximum level to display */
+#define TX_INFO_TONE		1.0	/* Level of tone relative to speech level (each component) */
+#define TX_ACK_TONE		0.1	/* Level of tone relative to speech level */
+#define INFO_TONE_F1		640.0
+#define INFO_TONE_F2		670.0
+#define ACK_TONE		1000.0
 
 /* Squelch */
 #define MUTE_TIME	0.1	/* Time until muting */
@@ -83,9 +83,9 @@ int dsp_init_sender(jolly_t *jolly, int nbfm, double squelch_db, int repeater)
 
 	/* set modulation parameters (NBFM uses half channel spacing, so we use half deviation) */
 	if (nbfm)
-		sender_set_fm(&jolly->sender, MAX_DEVIATION / 2.0, MAX_MODULATION, DBM0_DEVIATION / 2.0, MAX_DISPLAY);
+		sender_set_fm(&jolly->sender, MAX_DEVIATION / 2.0, MAX_MODULATION, SPEECH_DEVIATION / 2.0, MAX_DISPLAY);
 	else
-		sender_set_fm(&jolly->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
+		sender_set_fm(&jolly->sender, MAX_DEVIATION, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
 
 	/* init dtmf audio processing.
 	 * each frequency may be +6 dB deviation, which means a total deviation of +12 dB is allowed for detection.
diff --git a/src/libmobile/sender.c b/src/libmobile/sender.c
index fe2951f..31d1dd0 100644
--- a/src/libmobile/sender.c
+++ b/src/libmobile/sender.c
@@ -281,15 +281,16 @@ void sender_destroy(sender_t *sender)
 	jitter_destroy(&sender->dejitter);
 }
 
-void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double dBm0_deviation, double max_display)
+/* set frequency modulation and parameters */
+void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double speech_deviation, double max_display)
 {
 	sender->max_deviation = max_deviation;
 	sender->max_modulation = max_modulation;
-	sender->dBm0_deviation = dBm0_deviation;
+	sender->speech_deviation = speech_deviation;
 	sender->max_display = max_display;
 
 	PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Maxium deviation: %.1f kHz, Maximum modulation: %.1f kHz\n", max_deviation / 1000.0, max_modulation / 1000.0);
-	PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Deviation at dBm0 (audio level): %.1f kHz\n", dBm0_deviation / 1000.0);
+	PDEBUG_CHAN(DSENDER, DEBUG_DEBUG, "Deviation at speech level: %.1f kHz\n", speech_deviation / 1000.0);
 }
 
 static void gain_samples(sample_t *samples, int length, double gain)
@@ -361,8 +362,8 @@ cant_recover:
 			/* do pre emphasis towards radio */
 			if (inst->pre_emphasis)
 				pre_emphasis(&inst->estate, samples[i], count);
-			/* normal level to frequency deviation of dBm0 */
-			gain_samples(samples[i], count, inst->dBm0_deviation);
+			/* normal level to frequency deviation of speech level */
+			gain_samples(samples[i], count, inst->speech_deviation);
 			/* set paging signal */
 			paging_signal[i] = inst->paging_signal;
 			on[i] = inst->paging_on;
@@ -417,8 +418,8 @@ cant_recover:
 
 		/* loop through all channels */
 		for (i = 0, inst = sender; inst; i++, inst = inst->slave) {
-			/* frequency deviation of dBm0 to normal level */
-			gain_samples(samples[i], count, 1.0 / inst->dBm0_deviation);
+			/* frequency deviation of speech level to normal level */
+			gain_samples(samples[i], count, 1.0 / inst->speech_deviation);
 			/* rx gain */
 			if (inst->rx_gain != 1.0)
 				gain_samples(samples[i], count, inst->rx_gain);
diff --git a/src/libmobile/sender.h b/src/libmobile/sender.h
index 746a1d0..a2721f1 100644
--- a/src/libmobile/sender.h
+++ b/src/libmobile/sender.h
@@ -36,7 +36,7 @@ typedef struct sender {
 	/* fm levels */
 	double			max_deviation;		/* max frequency deviation */
 	double			max_modulation;		/* max frequency modulated */
-	double			dBm0_deviation;		/* deviation of 1000 Hz reference tone at dBm0 */
+	double			speech_deviation;	/* deviation of 1000 Hz reference tone at speech level */
 	double			max_display;		/* level of displaying wave form */
 
 	/* audio */
@@ -92,7 +92,7 @@ extern int cant_recover;
 
 int sender_create(sender_t *sender, const char *kanal, double sendefrequenz, double empfangsfrequenz, const char *audiodev, int use_sdr, int samplerate, double rx_gain, int pre_emphasis, int de_emphasis, const char *write_rx_wave, const char *write_tx_wave, const char *read_rx_wave, const char *read_tx_wave, int loopback, enum paging_signal paging_signal);
 void sender_destroy(sender_t *sender);
-void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double dBm0_deviation, double max_display);
+void sender_set_fm(sender_t *sender, double max_deviation, double max_modulation, double speech_deviation, double max_display);
 int sender_open_audio(int latspl);
 int sender_start_audio(void);
 void process_sender_audio(sender_t *sender, int *quit, int latspl);
diff --git a/src/libsendevolumenregler/sendevolumenregler.c b/src/libsendevolumenregler/sendevolumenregler.c
index 30b3285..6cc2196 100644
--- a/src/libsendevolumenregler/sendevolumenregler.c
+++ b/src/libsendevolumenregler/sendevolumenregler.c
@@ -38,22 +38,22 @@
 /*
  * Init function
  *
- * abwaerts_dbs = how many dB per second to lower the amplification, if input signal is above dBm0
- * aufwaerts_dbs = how many dB per second to raise the amplification, if input signal is below dBm0
- * maximum_db = limit of the output level above dBm0
+ * abwaerts_dbs = how many dB per second to lower the amplification, if input signal is above db0
+ * aufwaerts_dbs = how many dB per second to raise the amplification, if input signal is below db0
+ * maximum_db = limit of the output level above db0
  * minimum_db = below this input level, the amplification is not raised, so it stays constant.
- * dbm0_level = target level to be treated as 0 dB
+ * db0_level = target level to be treated as 0 dB
  *
  * Hopefully this is correct
  *
  */
-void init_sendevolumenregler(sendevolumenregler_t *state, double samplerate, double abwaerts_dbs, double aufwaerts_dbs, double maximum_db, double minimum_db, double dbm0_level)
+void init_sendevolumenregler(sendevolumenregler_t *state, double samplerate, double abwaerts_dbs, double aufwaerts_dbs, double maximum_db, double minimum_db, double db0_level)
 {
 	memset(state, 0, sizeof(*state));
 
 	state->peak = 1.0;
 	state->envelope = 1.0;
-	state->dbm0_level = dbm0_level;
+	state->db0_level = db0_level;
 	state->step_down = pow(db2level(abwaerts_dbs), 1.0 / samplerate);
 	state->step_up = pow(db2level(aufwaerts_dbs), 1.0 / samplerate);
 	state->maximum_level = db2level(maximum_db);
@@ -72,10 +72,10 @@ void init_sendevolumenregler(sendevolumenregler_t *state, double samplerate, dou
  */
 void sendevolumenregler(sendevolumenregler_t *state, sample_t *samples, int num)
 {
-	double value, peak, envelope, step_up, step_down, maximum_level, minimum_level, dbm0_level;
+	double value, peak, envelope, step_up, step_down, maximum_level, minimum_level, db0_level;
 	int i;
 
-	dbm0_level = state->dbm0_level;
+	db0_level = state->db0_level;
 	step_up = state->step_up;
 	step_down = state->step_down;
 	maximum_level = state->maximum_level;
@@ -84,8 +84,8 @@ void sendevolumenregler(sendevolumenregler_t *state, sample_t *samples, int num)
 	envelope = state->envelope;
 
 	for (i = 0; i < num; i++) {
-		/* normalize sample value to dbm0_level level */
-		value = *samples / dbm0_level;
+		/* normalize sample value to db0_level level */
+		value = *samples / db0_level;
 
 		/* 'peak' is the level that raises directly with the value
 		 * level, but falls as specified by step_up. */
@@ -106,7 +106,7 @@ void sendevolumenregler(sendevolumenregler_t *state, sample_t *samples, int num)
 		if (peak / envelope > maximum_level)
 			envelope = peak / maximum_level;
 
-		*samples++ = value / envelope * dbm0_level;
+		*samples++ = value / envelope * db0_level;
 	}
 
 	state->envelope = envelope;
diff --git a/src/libsendevolumenregler/sendevolumenregler.h b/src/libsendevolumenregler/sendevolumenregler.h
index 2bc916b..5d21049 100644
--- a/src/libsendevolumenregler/sendevolumenregler.h
+++ b/src/libsendevolumenregler/sendevolumenregler.h
@@ -6,9 +6,9 @@ typedef struct sendevolumenregler {
 	double	step_down;
 	double	minimum_level;
 	double	maximum_level;
-	double	dbm0_level;
+	double	db0_level;
 } sendevolumenregler_t;
 
-void init_sendevolumenregler(sendevolumenregler_t *state, double samplerate, double abwaerts_dbs, double aufwaerts_dbs, double maximum_db, double minimum_db, double dbm0_level);
+void init_sendevolumenregler(sendevolumenregler_t *state, double samplerate, double abwaerts_dbs, double aufwaerts_dbs, double maximum_db, double minimum_db, double db0_level);
 void sendevolumenregler(sendevolumenregler_t *state, sample_t *samples, int num);
 
diff --git a/src/nmt/dsp.c b/src/nmt/dsp.c
index d8e1077..d942c03 100644
--- a/src/nmt/dsp.c
+++ b/src/nmt/dsp.c
@@ -35,11 +35,11 @@
 
 /* Notes on TX_PEAK_FSK level:
  *
- * This deviation is -2.2db below the dBm0 deviation.
+ * This deviation is -2.2db below the speech deviation.
  *
  * At 1800 Hz the deviation shall be 4.2 kHz, so with emphasis the deviation
  * at 1000 Hz would be theoretically 2.333 kHz. This is factor 0.777 below
- * 3 kHz deviation we want at dBm0.
+ * 3 kHz deviation we want at speech.
  */
 
 /* Notes on TX_PEAK_SUPER (supervisory signal) level:
@@ -54,14 +54,14 @@
 /* signaling */
 #define MAX_DEVIATION		4700.0
 #define MAX_MODULATION		4055.0
-#define DBM0_DEVIATION		3000.0	/* deviation of dBm0 at 1 kHz */
-#define TX_PEAK_FSK		(4200.0 / 1800.0 * 1000.0 / DBM0_DEVIATION)
-#define TX_PEAK_SUPER		(300.0 / 4015.0 * 1000.0 / DBM0_DEVIATION)
+#define SPEECH_DEVIATION	3000.0	/* deviation of speech at 1 kHz */
+#define TX_PEAK_FSK		(4200.0 / 1800.0 * 1000.0 / SPEECH_DEVIATION)
+#define TX_PEAK_SUPER		(300.0 / 4015.0 * 1000.0 / SPEECH_DEVIATION)
 #define BIT_RATE		1200.0
 #define BIT_ADJUST		0.1	/* how much do we adjust bit clock on frequency change */
 #define F0			1800.0
 #define F1			1200.0
-#define MAX_DISPLAY		1.4	/* something above dBm0 */
+#define MAX_DISPLAY		1.4	/* something above speech level */
 #define DIALTONE_HZ		425.0	/* dial tone frequency */
 #define TX_PEAK_DIALTONE	1.0	/* dial tone peak FIXME: Not found in the specs! */
 #define SUPER_DURATION		0.25	/* duration of supervisory signal measurement */
@@ -114,7 +114,7 @@ int dsp_init_sender(nmt_t *nmt, double deviation_factor)
 	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
 
 	/* set modulation parameters */
-	sender_set_fm(&nmt->sender, MAX_DEVIATION * deviation_factor, MAX_MODULATION * deviation_factor, DBM0_DEVIATION * deviation_factor, MAX_DISPLAY);
+	sender_set_fm(&nmt->sender, MAX_DEVIATION * deviation_factor, MAX_MODULATION * deviation_factor, SPEECH_DEVIATION * deviation_factor, MAX_DISPLAY);
 
 	PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f (%.3f KHz deviation @ 1500 Hz)\n", TX_PEAK_FSK * deviation_factor, 3.5 * deviation_factor);
 	PDEBUG(DDSP, DEBUG_DEBUG, "Using Supervisory level of %.3f (%.3f KHz deviation @ 4015 Hz)\n", TX_PEAK_SUPER * deviation_factor, 0.3 * deviation_factor);
diff --git a/src/r2000/dsp.c b/src/r2000/dsp.c
index 771089c..4d28685 100644
--- a/src/r2000/dsp.c
+++ b/src/r2000/dsp.c
@@ -47,9 +47,9 @@
 /* signaling */
 #define MAX_DEVIATION		2500.0
 #define MAX_MODULATION		2550.0
-#define DBM0_DEVIATION		1500.0	/* deviation of dBm0 at 1 kHz */
-#define TX_PEAK_FSK		(1425.0 / 1500.0 * 1000.0 / DBM0_DEVIATION) /* with emphasis */
-#define TX_PEAK_SUPER		(300.0 / DBM0_DEVIATION) /* no emphasis */
+#define SPEECH_DEVIATION	1500.0	/* deviation of speech at 1 kHz */
+#define TX_PEAK_FSK		(1425.0 / 1500.0 * 1000.0 / SPEECH_DEVIATION) /* with emphasis */
+#define TX_PEAK_SUPER		(300.0 / SPEECH_DEVIATION) /* no emphasis */
 #define FSK_BIT_RATE		1200.0
 #define FSK_BIT_ADJUST		0.1	/* how much do we adjust bit clock on frequency change */
 #define FSK_F0			1800.0
@@ -59,7 +59,7 @@
 #define SUPER_F0		136.0
 #define SUPER_F1		164.0
 #define SUPER_CUTOFF_H		400.0	/* filter to remove spectrum of supervisory signal */
-#define MAX_DISPLAY		1.4	/* something above dBm0 */
+#define MAX_DISPLAY		1.4	/* something above speech level */
 
 /* global init for FSK */
 void dsp_init(void)
@@ -80,7 +80,7 @@ int dsp_init_sender(r2000_t *r2000)
 	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init DSP for Transceiver.\n");
 
 	/* set modulation parameters */
-	sender_set_fm(&r2000->sender, MAX_DEVIATION, MAX_MODULATION, DBM0_DEVIATION, MAX_DISPLAY);
+	sender_set_fm(&r2000->sender, MAX_DEVIATION, MAX_MODULATION, SPEECH_DEVIATION, MAX_DISPLAY);
 
 	PDEBUG(DDSP, DEBUG_DEBUG, "Using FSK level of %.3f\n", TX_PEAK_FSK);