diff --git a/Source/Core/Core/Src/DSP/DSPAccelerator.cpp b/Source/Core/Core/Src/DSP/DSPAccelerator.cpp
index 92651205a9..525750fc7d 100644
--- a/Source/Core/Core/Src/DSP/DSPAccelerator.cpp
+++ b/Source/Core/Core/Src/DSP/DSPAccelerator.cpp
@@ -165,14 +165,11 @@ u16 dsp_read_accelerator()
 	// Somehow, YN1 and YN2 must be initialized with their "loop" values,
 	// so yeah, it seems likely that we should raise an exception to let
 	// the DSP program do that, at least if DSP_FORMAT == 0x0A.
-
-	if (Address > EndAddress && g_dsp.ifx_regs[DSP_FORMAT] != 0x00) // PCM audio
-		Address = EndAddress;
-
-	if (Address == EndAddress)
+	if (Address >= EndAddress)
 	{
 		// Set address back to start address.
-		Address = (g_dsp.ifx_regs[DSP_ACSAH] << 16) | g_dsp.ifx_regs[DSP_ACSAL];
+		if (Address == EndAddress)
+			Address = (g_dsp.ifx_regs[DSP_ACSAH] << 16) | g_dsp.ifx_regs[DSP_ACSAL];
 		DSPCore_SetException(EXP_ACCOV);
 	}
 
diff --git a/Source/Core/Core/Src/HW/DSP.cpp b/Source/Core/Core/Src/HW/DSP.cpp
index f2509555ed..e86231db43 100644
--- a/Source/Core/Core/Src/HW/DSP.cpp
+++ b/Source/Core/Core/Src/HW/DSP.cpp
@@ -697,14 +697,12 @@ void UpdateAudioDMA()
 
 void Do_ARAM_DMA()
 {
-	// Fake the DMA taking time to complete. The delay is not accurate, but
-	// seems like a good estimate
-	CoreTiming::ScheduleEvent_Threadsafe(g_arDMA.Cnt.count >> 1, et_GenerateDSPInterrupt, INT_ARAM | (1<<16));
-
 	// Emulating the DMA wait time fixes Knockout Kings 2003 in DSP HLE mode
 	if (!GetDSPEmulator()->IsLLE())
 		g_dspState.DSPControl.DMAState = 1;
 
+	GenerateDSPInterrupt(INT_ARAM, true);
+
 	// Real hardware DMAs in 32byte chunks, but we can get by with 8byte chunks
 	if (g_arDMA.Cnt.dir)
 	{
diff --git a/Source/Core/Core/Src/HW/DSPHLE/UCodes/UCode_AX_Voice.h b/Source/Core/Core/Src/HW/DSPHLE/UCodes/UCode_AX_Voice.h
index 7649a53091..9a058afb83 100644
--- a/Source/Core/Core/Src/HW/DSPHLE/UCodes/UCode_AX_Voice.h
+++ b/Source/Core/Core/Src/HW/DSPHLE/UCodes/UCode_AX_Voice.h
@@ -232,12 +232,12 @@ inline void MixAddVoice(ParamBlockType &pb,
 			temprbuffer[s] += right;
 
 			// Control the behavior when we reach the end of the sample
-			if ((pb.audio_addr.sample_format != AUDIOFORMAT_ADPCM && samplePos >= sampleEnd) ||
-				(pb.audio_addr.sample_format == AUDIOFORMAT_ADPCM && samplePos == sampleEnd))
+			if (samplePos >= sampleEnd)
 			{
 				if (pb.audio_addr.looping == 1)
 				{
-					samplePos = loopPos;
+					if (samplePos == sampleEnd || (pb.audio_addr.sample_format != AUDIOFORMAT_ADPCM))
+						samplePos = loopPos;
 					if ((!pb.is_stream) && (pb.audio_addr.sample_format == AUDIOFORMAT_ADPCM))
 					{
 						pb.adpcm.yn1 = pb.adpcm_loop_info.yn1;