From a3835b001dbdfd639b34f6530817a3ae9a775bb8 Mon Sep 17 00:00:00 2001
From: Andreas Eversberg <jolly@eversberg.eu>
Date: Sun, 30 Oct 2022 11:32:07 +0100
Subject: [PATCH] AMPS: Minor fixes at DTX handling

---
 src/amps/amps.c | 7 -------
 src/amps/dsp.c  | 8 ++++++--
 2 files changed, 6 insertions(+), 9 deletions(-)

diff --git a/src/amps/amps.c b/src/amps/amps.c
index 79332d4..dafff22 100644
--- a/src/amps/amps.c
+++ b/src/amps/amps.c
@@ -819,13 +819,6 @@ void amps_rx_sat(amps_t *amps, int tone, double quality)
 		amps_set_dsp_mode(amps, DSP_MODE_AUDIO_RX_FRAME_TX, 0);
 	}
 
-	/* no SAT timeout handling during alerting */
-	if (trans->state == TRANS_CALL_MT_ALERT
-	 || trans->state == TRANS_CALL_MT_ALERT_SEND
-	 || trans->state == TRANS_CALL_MT_ALERT_CONFIRM
-	 || trans->state == TRANS_CALL_MT_ANSWER_WAIT)
-		return;
-
 	if (tone) {
 		timer_stop(&trans->timer);
 	} else {
diff --git a/src/amps/dsp.c b/src/amps/dsp.c
index 26bc633..a18abee 100644
--- a/src/amps/dsp.c
+++ b/src/amps/dsp.c
@@ -125,7 +125,7 @@
 #define SIG_LOST_COUNT		4	/* number of measures to loose Signaling Tone */
 #define CUT_OFF_HIGHPASS	300.0   /* cut off frequency for high pass filter to remove dc level from sound card / sample */
 #define BEST_QUALITY		0.68	/* Best possible RX quality */
-#define COMFORT_NOISE		0.02	/* audio level of comfort noise (relative to ISDN level) */
+#define COMFORT_NOISE		0.02	/* audio level of comfort noise (relative to speech level) */
 
 static sample_t ramp_up[256], ramp_down[256];
 
@@ -253,6 +253,7 @@ int dsp_init_sender(amps_t *amps, int tolerant)
 		PDEBUG(DDSP, DEBUG_ERROR, "No memory!\n");
 		return -ENOMEM;
 	}
+	PDEBUG(DDSP, DEBUG_DEBUG, "Sat detection interval is %d ms.\n", amps->sat_samples * 1000 / amps->sender.samplerate);
 	amps->sat_filter_spl = spl;
 
 	/* count SAT tones */
@@ -840,7 +841,7 @@ static void sat_decode(amps_t *amps, sample_t *samples, int length)
 static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
 {
 	transaction_t *trans = amps->trans_list;
-	sample_t *spl;
+	sample_t *spl, s;
 	int max, pos;
 	int i;
 
@@ -849,7 +850,10 @@ static void sender_receive_audio(amps_t *amps, sample_t *samples, int length)
 	spl = amps->sat_filter_spl;
 	pos = amps->sat_filter_pos;
 	for (i = 0; i < length; i++) {
+		/* unmute: use buffer, to delay audio, so we do not miss that chunk when SAT is detected */
+		s = spl[pos];
 		spl[pos++] = samples[i];
+		samples[i] = s;
 		if (pos == max) {
 			pos = 0;
 			sat_decode(amps, spl, max);