diff --git a/src/cnetz/cnetz.h b/src/cnetz/cnetz.h
index 483bb3a..2033907 100644
--- a/src/cnetz/cnetz.h
+++ b/src/cnetz/cnetz.h
@@ -127,8 +127,8 @@ struct cnetz {
 	double			frame_last_phase;	/* master's bit phase of last frame sync */
 
 	/* audio offset removal */
-	double			offset_factor;		/* filer alpha of high-pass filter */
-	double			offset_y_last;		/* last stored sample */
+	double			offset_last;		/* last sample value of last frame */
+	int			offset_range;		/* range of samples to ramp the offset */
 
 	/* measurements */
 	int			measure_speed;		/* measure clock speed */
diff --git a/src/cnetz/dsp.c b/src/cnetz/dsp.c
index 8880fef..02ae6b9 100644
--- a/src/cnetz/dsp.c
+++ b/src/cnetz/dsp.c
@@ -46,7 +46,6 @@
 #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) */
 
 #ifdef TEST_SCRAMBLE
 jitter_t scrambler_test_jb;
@@ -103,7 +102,6 @@ int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], en
 {
 	int rc = 0;
 	double size;
-	double RC, dt;
 
 	PDEBUG_CHAN(DDSP, DEBUG_DEBUG, "Init FSK for 'Sender'.\n");
 
@@ -150,7 +148,7 @@ int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], en
 	}
 
 	/* reinit the sample rate to shrink/expand audio */
-	init_samplerate(&cnetz->sender.srstate, 8000.0, (double)cnetz->sender.samplerate / 1.1, 3300.0); /* 66 <-> 60 */
+	init_samplerate(&cnetz->sender.srstate, 8000.0, (double)cnetz->sender.samplerate / (1.1 / (1.0 + clock_speed[0] / 1000000.0)), 3300.0); /* 66 <-> 60 */
 
 	rc = fsk_fm_init(&cnetz->fsk_demod, cnetz, cnetz->sender.samplerate, (double)BITRATE / (1.0 + clock_speed[0] / 1000000.0), demod);
 	if (rc < 0)
@@ -170,10 +168,8 @@ int dsp_init_sender(cnetz_t *cnetz, int measure_speed, double clock_speed[2], en
 	 * shall not exceed according to ITU G.162 */
 	init_compandor(&cnetz->cstate, 8000, 5.0, 22.5);
 
-	/* use this filter to compensate level changes between two subsequent audio chunks */
-	RC = 1.0 / (CUT_OFF_OFFSET * 2.0 *3.14);
-	dt = 1.0 / cnetz->sender.samplerate;
-	cnetz->offset_factor = RC / (RC + dt);
+	/* use duration of one bit to ramp level of last frame to current frame */
+	cnetz->offset_range = ceil(cnetz->fsk_bitduration);
 
 #ifdef TEST_SCRAMBLE
 	rc = jitter_create(&scrambler_test_jb, cnetz->sender.samplerate / 5);
@@ -821,7 +817,8 @@ void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count)
 {
 	sample_t *spl;
 	int pos, i;
-	double x, y, x_last, y_last, factor;
+	int range;
+	double offset;
 
 	/* check if we still have a transaction
 	 * this might not be true, if we just released transaction, but still
@@ -830,21 +827,13 @@ void unshrink_speech(cnetz_t *cnetz, sample_t *speech_buffer, int count)
 	if (!cnetz->trans_list)
 		return;
 
-	/* fix offset between speech blocks by using high pass filter */
-	/* use first sample as previous sample, so we don't have a level jump between two subsequent audio chunks */
-	x_last = speech_buffer[0];
-	y_last = cnetz->offset_y_last;
-	factor = cnetz->offset_factor;
-	for (i = 0; i < count; i++) {
-		/* change level */
-		x = speech_buffer[i];
-		/* high-pass to remove low level frequencies, caused by level jump between audio chunks */
-		y = factor * (y_last + x - x_last);
-		x_last = x;
-		y_last = y;
-		speech_buffer[i] = y;
+	/* ramp from level of last frame to level of current frame */
+	range = cnetz->offset_range;
+	offset = speech_buffer[0] - cnetz->offset_last;
+	for (i = 0; i < range; i++) {
+		speech_buffer[i] -= offset * (1.0 - (double)i / (double)range);
 	}
-	cnetz->offset_y_last = y_last;
+	cnetz->offset_last = speech_buffer[count - 1];
 
 	/* 4. de-emphasis is done by cnetz code, not by common code */
 	/* de-emphasis is only used when scrambler is off, see FTZ 171 TR 60 Clause 4 */
diff --git a/src/cnetz/fsk_demod.c b/src/cnetz/fsk_demod.c
index 5ffc8fb..a1acd70 100644
--- a/src/cnetz/fsk_demod.c
+++ b/src/cnetz/fsk_demod.c
@@ -121,7 +121,10 @@
  * if debug is set to 1, debugging will start at program start
  */
 //#define DEBUG_DECODER
-//static int debug = 0;
+
+#ifdef DEBUG_DECODER
+static int debug = 0;
+#endif
 
 #include <stdio.h>
 #include <stdint.h>
@@ -182,7 +185,7 @@ int fsk_fm_init(fsk_fm_demod_t *fsk, cnetz_t *cnetz, int samplerate, double bitr
 
 #ifdef DEBUG_DECODER
 	char debug_filename[256];
-	sprintf(debug_filename, "/tmp/debug_decoder_channel_%d.txt", cnetz->sender.kanal);
+	sprintf(debug_filename, "/tmp/debug_decoder_channel_%s.txt", cnetz->sender.kanal);
 	fsk->debug_fp = fopen(debug_filename, "w");
 	if (!fsk->debug_fp) {
 		fprintf(stderr, "Failed to open decoder debug file '%s'!\n", debug_filename);
@@ -414,17 +417,6 @@ static inline void find_change_slope(fsk_fm_demod_t *fsk)
 	sample_t threshold;
 	int i;
 	
-#ifdef DEBUG_DECODER
-	/* show deviation of middle sample in windows (in a range of bandwidth) */
-	if (debug) {
-		fprintf(fsk->debug_fp, "%s",
-			debug_amplitude(
-				fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len]
-			)
-		);
-	}
-#endif
-
 	/* get level range (level_min and level_max) and also
 	 * get maximum slope (change_max) and where it was
 	 * (change_at) and what direction it went (change_positive)
@@ -492,10 +484,6 @@ static inline void find_change_slope(fsk_fm_demod_t *fsk)
 	}
 	fsk->next_bit -= fsk->bits_per_sample;
 
-#ifdef DEBUG_DECODER
-	if (debug)
-		fprintf(fsk->debug_fp, "\n");
-#endif
 }
 
 /* find bit change by looking at zero crossing */
@@ -507,12 +495,6 @@ static inline void find_change_level(fsk_fm_demod_t *fsk)
 	/* get bit in the middle of the buffer */
 	s = fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len];
 
-#ifdef DEBUG_DECODER
-	/* show deviation */
-	if (debug)
-		fprintf(fsk->debug_fp, "%s", debug_amplitude(s));
-#endif
-
 	/* just sample first bit in distributed mode */
 	if (fsk->cnetz->dsp_mode == DSP_MODE_SPK_V && fsk->bit_count == 0) {
 		if (fmod(fsk->bit_time, BITS_PER_SPK_BLOCK) < 1.5)
@@ -569,10 +551,6 @@ static inline void find_change_level(fsk_fm_demod_t *fsk)
 	fsk->next_bit -= fsk->bits_per_sample;
 
 done:
-#ifdef DEBUG_DECODER
-	if (debug)
-		fprintf(fsk->debug_fp, "\n");
-#endif
 	return;
 }
 
@@ -580,7 +558,7 @@ done:
 void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
 {
 	int i;
-	double t;
+	double t = 0.0;
 
 	/* process signaling block, sample by sample */
 	for (i = 0; i < length; i++) {
@@ -590,6 +568,14 @@ void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
 			if (fsk->bit_buffer_pos == fsk->bit_buffer_len)
 				fsk->bit_buffer_pos = 0;
 
+#ifdef DEBUG_DECODER
+			/* show deviation of center sample in window */
+			if (debug)
+				fprintf(fsk->debug_fp, "%s", debug_amplitude(fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len]));
+			if (debug && fmod(fsk->bit_time - fsk->bits_per_sample, 1.0) > fmod(fsk->bit_time, 1.0))
+				fprintf(fsk->debug_fp, " -bitchange-");
+#endif
+
 			/* for each sample process buffer */
 			if (fsk->cnetz->dsp_mode != DSP_MODE_SPK_V) {
 				if (fsk->demod_type == FSK_DEMOD_SLOPE)
@@ -611,7 +597,7 @@ void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
 					fsk->next_bit = 1.0 - fsk->bits_per_sample;
 #ifdef DEBUG_DECODER
 					if (debug && fsk->bit_count)
-						fprintf(fsk->debug_fp, "---- SPK(V) BLOCK START ----\n");
+						fprintf(fsk->debug_fp, "\n---- SPK(V) BLOCK START ----");
 #endif
 					fsk->bit_count = 0;
 				} else
@@ -622,13 +608,21 @@ void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
 						find_change_level(fsk);
 				} else
 				if (t >= 5.5 && t < 65.5) {
+#ifdef DEBUG_DECODER
+					if (debug && !fsk->speech_count)
+						fprintf(fsk->debug_fp, " (start recording speech)");
+#endif
 					/* get audio for the duration of 60 bits */
 					/* prevent overflow, if speech_size != 0 and SPK_V
 					 * has been restarted. */
 					if (fsk->speech_count < fsk->speech_size)
-						fsk->speech_buffer[fsk->speech_count++] = samples[i];
+						fsk->speech_buffer[fsk->speech_count++] = fsk->bit_buffer_spl[(fsk->bit_buffer_pos + fsk->bit_buffer_half) % fsk->bit_buffer_len];
 				} else
 				if (t >= 65.5) {
+#ifdef DEBUG_DECODER
+					if (debug && fsk->speech_count)
+						fprintf(fsk->debug_fp, " (stop recording speech)");
+#endif
 					if (fsk->speech_count) {
 						unshrink_speech(fsk->cnetz, fsk->speech_buffer, fsk->speech_count);
 						fsk->speech_count = 0;
@@ -641,6 +635,10 @@ void fsk_fm_demod(fsk_fm_demod_t *fsk, sample_t *samples, int length)
 		if (fsk->bit_time >= BITS_PER_SUPERFRAME) {
 			fsk->bit_time -= BITS_PER_SUPERFRAME;
 		}
+#ifdef DEBUG_DECODER
+		if (debug && samples)
+				fprintf(fsk->debug_fp, "\n");
+#endif
 		/* another clock is used to measure actual super frame time */
 		fsk->bit_time_uncorrected += fsk->bits_per_sample;
 		if (fsk->bit_time_uncorrected >= BITS_PER_SUPERFRAME) {