util/spkmodem-decode: automatic tone calibration

current logic is hardcoded, as in the original spkmodem-recv.

with this change, small differences are observed and averaged,
then the detection thresholds are adjusted accordingly.

the existing macros serve as a baseline, but real signals
differ. with this change, we therefore account for possible
drift in timings, which can change in real-time; the old
code could possibly get out of sync beccause of that, which
may have resulted in corrupt characters on the screen. this
change therefore should make the output a bit more stable.

the detection window is continually adjusted, so that the
output timings don't drift.

the tolerances are automatically adjusted based on base
timings (see new define in patch)

Signed-off-by: Leah Rowe <leah@libreboot.org>

1 files changed, 51 insertions, 6 deletions

diff --git a/util/spkmodem_decode/spkmodem-decode.c b/util/spkmodem_decode/spkmodem-decode.c
index 3c4b94c1..f08b1b52 100644
--- a/util/spkmodem_decode/spkmodem-decode.c
+++ b/util/spkmodem_decode/spkmodem-decode.c
@@ -128,6 +128,9 @@
 #define SEP_TONE_MIN_HZ 1000
 #define SEP_TONE_MAX_HZ 3000
 
+#define SEP_TOLERANCE_PULSES \
+    (((SEP_TONE_MAX_HZ) - (SEP_TONE_MIN_HZ)) / (2 * (FRAME_RATE)))
+
 #define DATA_TONE_MIN_HZ 3000
 #define DATA_TONE_MAX_HZ 12000
 
@@ -180,6 +183,12 @@ struct decoder_state {
 
 	int debug;
 	int swap_bytes;
+
+	/* dynamic separator calibration */
+	int sep_sum;
+	int sep_samples;
+	int sep_min;
+	int sep_max;
 };
 
 static const char *argv0;
@@ -228,6 +237,12 @@ main(int argc, char **argv)
 		break;
 	}
 
+	/*
+	 * Used for separator calibration
+	 */
+	st.sep_min = FREQ_SEP_MIN;
+	st.sep_max = FREQ_SEP_MAX;
+
 	st.ascii_bit = 7;
 
 	st.ringpos = 0;
@@ -254,6 +269,7 @@ host_is_big_endian(void)
 static void
 handle_audio(struct decoder_state *st)
 {
+	int avg;
 	int sample;
 
 	/*
@@ -262,7 +278,30 @@ handle_audio(struct decoder_state *st)
 	 */
 	if (st->sample_count >= (3 * SAMPLES_PER_FRAME))
 		reset_char(st);
+
 	if (!valid_signal(st)) {
+
+		/*
+		 * collect separator tone statistics
+		 * (and auto-adjust tolerances)
+		 */
+		if (st->sep_samples < 50 && st->freq_separator > 0) {
+			st->sep_sum += st->freq_separator;
+			st->sep_samples++;
+
+			if (st->sep_samples == 50) {
+				avg = st->sep_sum / st->sep_samples;
+
+				/* ±3 pulse window */
+				st->sep_min = avg - SEP_TOLERANCE_PULSES;
+				st->sep_max = avg + SEP_TOLERANCE_PULSES;
+
+				if (st->debug)
+					printf("separator calibrated: %dHz\n",
+					    avg * FRAME_RATE);
+			}
+		}
+
 		decode_pulse(st);
 		return;
 	}
@@ -283,8 +322,8 @@ handle_audio(struct decoder_state *st)
 static int
 valid_signal(struct decoder_state *st)
 {
-	return (st->freq_separator > FREQ_SEP_MIN &&
-	    st->freq_separator < FREQ_SEP_MAX &&
+	return (st->freq_separator > st->sep_min &&
+	    st->freq_separator < st->sep_max &&
 	    st->freq_data > FREQ_DATA_MIN &&
 	    st->freq_data < FREQ_DATA_MAX);
 }
@@ -438,24 +477,30 @@ print_stats(struct decoder_state *st)
 
 	int data_hz = st->freq_data * FRAME_RATE;
 	int sep_hz  = st->freq_separator * FRAME_RATE;
+	int sep_hz_min = st->sep_min * FRAME_RATE;
+	int sep_hz_max = st->sep_max * FRAME_RATE;
 
 	if ((pos = ftell(stdin)) == -1) {
-		printf("%d %d %d data=%dHz sep=%dHz\n",
+		printf("%d %d %d data=%dHz sep=%dHz(min %dHz %dHz)\n",
 		    st->freq_data,
 		    st->freq_separator,
 		    FREQ_DATA_THRESHOLD,
 		    data_hz,
-		    sep_hz);
+		    sep_hz,
+		    sep_hz_min,
+		    sep_hz_max);
 		return;
 	}
 
-	printf("%d %d %d @%ld data=%dHz sep=%dHz\n",
+	printf("%d %d %d @%ld data=%dHz sep=%dHz(min %dHz %dHz)\n",
 	    st->freq_data,
 	    st->freq_separator,
 	    FREQ_DATA_THRESHOLD,
 	    pos - SAMPLE_OFFSET,
 	    data_hz,
-	    sep_hz);
+	    sep_hz,
+	    sep_hz_min,
+	    sep_hz_max);
 }
 
 static void