diff --git a/Core/HW/SasAudio.cpp b/Core/HW/SasAudio.cpp
index 8c9bd5aeee..13a8aebf26 100644
--- a/Core/HW/SasAudio.cpp
+++ b/Core/HW/SasAudio.cpp
@@ -546,14 +546,12 @@ void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
 
 	int voicesPlayingCount = 0;
 
-	int sendVolumeSum = 0;  // Figure out if we actually need to run the effect. This can cut off echo trails though so TODO: Let's remove after debugging.
 	for (int v = 0; v < PSP_SAS_VOICES_MAX; v++) {
 		SasVoice &voice = voices[v];
 		if (!voice.playing || voice.paused)
 			continue;
 		voicesPlayingCount++;
 		MixVoice(voice);
-		sendVolumeSum += voice.effectLeft + voice.effectRight;
 	}
 
 	// Then mix the send buffer in with the rest.
@@ -564,7 +562,7 @@ void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
 	if (outputMode == PSP_SAS_OUTPUTMODE_MIXED) {
 		// Okay, apply effects processing to the Send buffer.
 		// TODO: Is this only done in PSP_SAS_OUTPUTMODE_MIXED?
-		if (sendVolumeSum && waveformEffect.type != PSP_SAS_EFFECT_TYPE_OFF) {
+		if (waveformEffect.type != PSP_SAS_EFFECT_TYPE_OFF) {
 			ApplyWaveformEffect();
 			// TODO: Mix send when it has proper values, probably based on dry/wet?
 			if (inp) {
diff --git a/Core/HW/SasReverb.cpp b/Core/HW/SasReverb.cpp
index b008d1a709..2f3c7ab7d2 100644
--- a/Core/HW/SasReverb.cpp
+++ b/Core/HW/SasReverb.cpp
@@ -226,6 +226,7 @@ void SasReverb::ProcessReverb(int16_t *output, const int16_t *input, size_t inpu
 
 	// This runs at 22khz.
 	// Very unoptimized, straight from the description. Can probably be reformulated into something way more efficient.
+	// For example, we could just do the whole thing in floating point. Should actually be faster.
 	for (int i = 0; i < inputSize; i++) {
 		int16_t LeftInput = input[i * 2];
 		int16_t RightInput = input[i * 2 + 1];
@@ -246,9 +247,9 @@ void SasReverb::ProcessReverb(int16_t *output, const int16_t *input, size_t inpu
 		b[d.mRAPF1] = clamp_s16(Rout - (d.vAPF1*b[(d.mRAPF1 - d.dAPF1) * 4] >> 15));
 		Rout = b[(d.mRAPF1 - d.dAPF1)*4] + (b[d.mRAPF1*4] * d.vAPF1 >> 15);
 		// ___Late Reverb APF2(All Pass Filter 2, with input from APF1)________________
-		b[d.mLAPF2] = clamp_s16(Lout - d.vAPF2*b[(d.mLAPF2 - d.dAPF2)*4]);
+		b[d.mLAPF2] = clamp_s16(Lout - (d.vAPF2*b[(d.mLAPF2 - d.dAPF2)*4] >> 15));
 		Lout = b[(d.mLAPF2 - d.dAPF2)*4] + (b[d.mLAPF2*4] * d.vAPF2 >> 15);
-		b[d.mRAPF2] = clamp_s16(Rout - d.vAPF2*b[(d.mRAPF2 - d.dAPF2)*4]);
+		b[d.mRAPF2] = clamp_s16(Rout - (d.vAPF2*b[(d.mRAPF2 - d.dAPF2)*4] >> 15));
 		Rout = b[(d.mRAPF2 - d.dAPF2)*4] + (b[d.mRAPF2*4] * d.vAPF2 >> 15);
 		// ___Output to Mixer(Output volume multiplied with input from APF2)___________
 		output[i*2] = clamp_s16(Lout*volLeft >> 15);