diff --git a/dlls/dsound/dsound_convert.c b/dlls/dsound/dsound_convert.c
index c9ce42614c2..d3d686a2cbd 100644
--- a/dlls/dsound/dsound_convert.c
+++ b/dlls/dsound/dsound_convert.c
@@ -125,13 +125,6 @@ static inline unsigned char f_to_8(float value)
     return lrintf((value + 1.f) * 0x80);
 }
 
-static void put8(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
-{
-    BYTE* buf = dsb->device->tmp_buffer;
-    buf += pos + channel;
-    *buf = f_to_8(value);
-}
-
 static inline SHORT f_to_16(float value)
 {
     if(value <= -1.f)
@@ -141,13 +134,6 @@ static inline SHORT f_to_16(float value)
     return le16(lrintf(value * 0x8000));
 }
 
-static void put16(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
-{
-    BYTE* buf = dsb->device->tmp_buffer;
-    SHORT *sbuf = (SHORT*)(buf + pos + 2 * channel);
-    *sbuf = f_to_16(value);
-}
-
 static LONG f_to_24(float value)
 {
     if(value <= -1.f)
@@ -157,17 +143,6 @@ static LONG f_to_24(float value)
     return lrintf(value * 0x80000000U);
 }
 
-static void put24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
-{
-    BYTE* buf = dsb->device->tmp_buffer;
-    LONG t;
-    buf += pos + 3 * channel;
-    t = f_to_24(value);
-    buf[0] = (t >> 8) & 0xFF;
-    buf[1] = (t >> 16) & 0xFF;
-    buf[2] = (t >> 24) & 0xFF;
-}
-
 static inline LONG f_to_32(float value)
 {
     if(value <= -1.f)
@@ -177,79 +152,26 @@ static inline LONG f_to_32(float value)
     return le32(lrintf(value * 0x80000000U));
 }
 
-static void put32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
+void putieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
 {
-    BYTE* buf = dsb->device->tmp_buffer;
-    LONG *sbuf = (LONG*)(buf + pos + 4 * channel);
-    *sbuf = f_to_32(value);
+    BYTE *buf = (BYTE *)dsb->device->tmp_buffer;
+    float *fbuf = (float*)(buf + pos + sizeof(float) * channel);
+    *fbuf = value;
 }
 
-const bitsputfunc putbpp[4] = {put8, put16, put24, put32};
-
 void put_mono2stereo(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
 {
     dsb->put_aux(dsb, pos, 0, value);
     dsb->put_aux(dsb, pos, 1, value);
 }
 
-static void mix8(signed char *src, float *dst, unsigned len)
+void mixieee32(float *src, float *dst, unsigned samples)
 {
-    TRACE("%p - %p %d\n", src, dst, len);
-    while (len--)
-        /* 8-bit WAV is unsigned, it's here converted to signed, normalize function will convert it back again */
-        *(dst++) += ((signed char)((BYTE)*(src++) - (BYTE)0x80)) / (float)0x80;
-}
-
-static void mix16(SHORT *src, float *dst, unsigned len)
-{
-    TRACE("%p - %p %d\n", src, dst, len);
-    len /= 2;
-    while (len--)
-    {
-        *dst += le16(*src) / (float)0x8000U;
-        ++dst; ++src;
-    }
-}
-
-static void mix24(BYTE *src, float *dst, unsigned len)
-{
-    TRACE("%p - %p %d\n", src, dst, len);
-    len /= 3;
-    while (len--)
-    {
-        DWORD field;
-        field = ((DWORD)src[2] << 16) + ((DWORD)src[1] << 8) + (DWORD)src[0];
-        if (src[2] & 0x80)
-            field |= 0xFF000000U;
-        *(dst++) += field / (float)0x800000U;
-        src += 3;
-    }
-}
-
-static void mix32(INT *src, float *dst, unsigned len)
-{
-    TRACE("%p - %p %d\n", src, dst, len);
-    len /= 4;
-    while (len--)
-        *(dst++) += le32(*(src++)) / (float)0x80000000U;
-}
-
-static void mixieee32(float *src, float *dst, unsigned len)
-{
-    TRACE("%p - %p %d\n", src, dst, len);
-    len /= 4;
-    while (len--)
+    TRACE("%p - %p %d\n", src, dst, samples);
+    while (samples--)
         *(dst++) += *(src++);
 }
 
-const mixfunc mixfunctions[5] = {
-    (mixfunc)mix8,
-    (mixfunc)mix16,
-    (mixfunc)mix24,
-    (mixfunc)mix32,
-    (mixfunc)mixieee32
-};
-
 static void norm8(float *src, unsigned char *dst, unsigned len)
 {
     TRACE("%p - %p %d\n", src, dst, len);
diff --git a/dlls/dsound/dsound_private.h b/dlls/dsound/dsound_private.h
index 26e59ffcbea..1d43c5a1264 100644
--- a/dlls/dsound/dsound_private.h
+++ b/dlls/dsound/dsound_private.h
@@ -55,9 +55,8 @@ typedef struct DirectSoundCaptureDevice      DirectSoundCaptureDevice;
 typedef float (*bitsgetfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD);
 typedef void (*bitsputfunc)(const IDirectSoundBufferImpl *, DWORD, DWORD, float);
 extern const bitsgetfunc getbpp[5] DECLSPEC_HIDDEN;
-extern const bitsputfunc putbpp[4] DECLSPEC_HIDDEN;
-typedef void (*mixfunc)(const void *, void *, unsigned);
-extern const mixfunc mixfunctions[5] DECLSPEC_HIDDEN;
+void putieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value) DECLSPEC_HIDDEN;
+void mixieee32(float *src, float *dst, unsigned samples) DECLSPEC_HIDDEN;
 typedef void (*normfunc)(const void *, void *, unsigned);
 extern const normfunc normfunctions[5] DECLSPEC_HIDDEN;
 
@@ -94,13 +93,11 @@ struct DirectSoundDevice
     CRITICAL_SECTION            mixlock;
     IDirectSoundBufferImpl     *primary;
     DWORD                       speaker_config;
-    LPBYTE                      tmp_buffer;
-    float *mix_buffer;
+    float *mix_buffer, *tmp_buffer;
     DWORD                       tmp_buffer_len, mix_buffer_len;
 
     DSVOLUMEPAN                 volpan;
 
-    mixfunc mixfunction;
     normfunc normfunction;
 
     /* DirectSound3DListener fields */
diff --git a/dlls/dsound/mixer.c b/dlls/dsound/mixer.c
index 9d4b35bce01..06e5773d2e0 100644
--- a/dlls/dsound/mixer.c
+++ b/dlls/dsound/mixer.c
@@ -141,7 +141,7 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
 	dsb->freqAcc = 0;
 
 	dsb->get_aux = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
-	dsb->put_aux = putbpp[dsb->device->pwfx->wBitsPerSample/8 - 1];
+	dsb->put_aux = putieee32;
 
 	dsb->get = dsb->get_aux;
 	dsb->put = dsb->put_aux;
@@ -234,10 +234,10 @@ static inline float get_current_sample(const IDirectSoundBufferImpl *dsb,
     return dsb->get(dsb, mixpos % dsb->buflen, channel);
 }
 
-static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb,
-        UINT ostride, UINT count)
+static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb, UINT count)
 {
     UINT istride = dsb->pwfx->nBlockAlign;
+    UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
     DWORD channel, i;
     for (i = 0; i < count; i++)
         for (channel = 0; channel < dsb->mix_channels; channel++)
@@ -246,11 +246,11 @@ static UINT cp_fields_noresample(IDirectSoundBufferImpl *dsb,
     return count;
 }
 
-static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb,
-        UINT ostride, UINT count, float *freqAcc)
+static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb, UINT count, float *freqAcc)
 {
     UINT i, channel;
     UINT istride = dsb->pwfx->nBlockAlign;
+    UINT ostride = dsb->device->pwfx->nChannels * sizeof(float);
 
     float freqAdjust = dsb->freqAdjust;
     float freqAcc_start = *freqAcc;
@@ -314,15 +314,14 @@ static UINT cp_fields_resample(IDirectSoundBufferImpl *dsb,
     return max_ipos;
 }
 
-static void cp_fields(IDirectSoundBufferImpl *dsb,
-        UINT ostride, UINT count, float *freqAcc)
+static void cp_fields(IDirectSoundBufferImpl *dsb, UINT count, float *freqAcc)
 {
     DWORD ipos, adv;
 
     if (dsb->freqAdjust == 1.0)
-        adv = cp_fields_noresample(dsb, ostride, count); /* *freqAcc is unmodified */
+        adv = cp_fields_noresample(dsb, count); /* *freqAcc is unmodified */
     else
-        adv = cp_fields_resample(dsb, ostride, count, freqAcc);
+        adv = cp_fields_resample(dsb, count, freqAcc);
 
     ipos = dsb->sec_mixpos + adv * dsb->pwfx->nBlockAlign;
     if (ipos >= dsb->buflen) {
@@ -365,89 +364,53 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
  *
  * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
  */
-static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD tmp_len)
+static void DSOUND_MixToTemporary(IDirectSoundBufferImpl *dsb, DWORD frames)
 {
-	INT	oAdvance = dsb->device->pwfx->nBlockAlign;
-	INT	size = tmp_len / oAdvance;
+	UINT size_bytes = frames * sizeof(float) * dsb->device->pwfx->nChannels;
 
-	if (dsb->device->tmp_buffer_len < tmp_len || !dsb->device->tmp_buffer)
+	if (dsb->device->tmp_buffer_len < size_bytes || !dsb->device->tmp_buffer)
 	{
-		dsb->device->tmp_buffer_len = tmp_len;
+		dsb->device->tmp_buffer_len = size_bytes;
 		if (dsb->device->tmp_buffer)
-			dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, tmp_len);
+			dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, size_bytes);
 		else
-			dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, tmp_len);
+			dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, size_bytes);
 	}
 
-	cp_fields(dsb, oAdvance, size, &dsb->freqAcc);
+	cp_fields(dsb, frames, &dsb->freqAcc);
 }
 
-/** Apply volume to the given soundbuffer from (primary) position writepos and length len
- * Returns: NULL if no volume needs to be applied
- * or else a memory handle that holds 'len' volume adjusted buffer */
-static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
+static void DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT frames)
 {
 	INT	i;
-	BYTE	*bpc;
-	INT16	*bps, *mems;
-	DWORD vLeft, vRight;
-	INT nChannels = dsb->device->pwfx->nChannels;
-	LPBYTE mem = dsb->device->tmp_buffer;
+	float vLeft, vRight;
+	UINT channels = dsb->device->pwfx->nChannels, chan;
 
-	TRACE("(%p,%d)\n",dsb,len);
+	TRACE("(%p,%d)\n",dsb,frames);
 	TRACE("left = %x, right = %x\n", dsb->volpan.dwTotalLeftAmpFactor,
 		dsb->volpan.dwTotalRightAmpFactor);
 
 	if ((!(dsb->dsbd.dwFlags & DSBCAPS_CTRLPAN) || (dsb->volpan.lPan == 0)) &&
 	    (!(dsb->dsbd.dwFlags & DSBCAPS_CTRLVOLUME) || (dsb->volpan.lVolume == 0)) &&
 	     !(dsb->dsbd.dwFlags & DSBCAPS_CTRL3D))
-		return NULL; /* Nothing to do */
+		return; /* Nothing to do */
 
-	if (nChannels != 1 && nChannels != 2)
+	if (channels != 1 && channels != 2)
 	{
-		FIXME("There is no support for %d channels\n", nChannels);
-		return NULL;
+		FIXME("There is no support for %u channels\n", channels);
+		return;
 	}
 
-	if (dsb->device->pwfx->wBitsPerSample != 8 && dsb->device->pwfx->wBitsPerSample != 16)
-	{
-		FIXME("There is no support for %d bpp\n", dsb->device->pwfx->wBitsPerSample);
-		return NULL;
-	}
-
-	assert(dsb->device->tmp_buffer_len >= len && dsb->device->tmp_buffer);
-
-	bpc = dsb->device->tmp_buffer;
-	bps = (INT16 *)bpc;
-	mems = (INT16 *)mem;
-	vLeft = dsb->volpan.dwTotalLeftAmpFactor;
-	if (nChannels > 1)
-		vRight = dsb->volpan.dwTotalRightAmpFactor;
-	else
-		vRight = vLeft;
-
-	switch (dsb->device->pwfx->wBitsPerSample) {
-	case 8:
-		/* 8-bit WAV is unsigned, but we need to operate */
-		/* on signed data for this to work properly */
-		for (i = 0; i < len-1; i+=2) {
-			*(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
-			*(bpc++) = (((*(mem++) - 128) * vRight) >> 16) + 128;
+	vLeft = dsb->volpan.dwTotalLeftAmpFactor / ((float)0xFFFF);
+	vRight = dsb->volpan.dwTotalRightAmpFactor / ((float)0xFFFF);
+	for(i = 0; i < frames; ++i){
+		for(chan = 0; chan < channels; ++chan){
+			if(chan == 0)
+				dsb->device->tmp_buffer[i * channels + chan] *= vLeft;
+			else
+				dsb->device->tmp_buffer[i * channels + chan] *= vRight;
 		}
-		if (len % 2 == 1 && nChannels == 1)
-			*(bpc++) = (((*(mem++) - 128) * vLeft) >> 16) + 128;
-		break;
-	case 16:
-		/* 16-bit WAV is signed -- much better */
-		for (i = 0; i < len-3; i += 4) {
-			*(bps++) = (*(mems++) * vLeft) >> 16;
-			*(bps++) = (*(mems++) * vRight) >> 16;
-		}
-		if (len % 4 == 2 && nChannels == 1)
-			*(bps++) = ((INT)*(mems++) * vLeft) >> 16;
-		break;
 	}
-	return dsb->device->tmp_buffer;
 }
 
 /**
@@ -466,8 +429,9 @@ static LPBYTE DSOUND_MixerVol(const IDirectSoundBufferImpl *dsb, INT len)
 static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
 {
 	INT len = fraglen;
-	BYTE *ibuf, *volbuf;
+	float *ibuf;
 	DWORD oldpos;
+	UINT frames = fraglen / dsb->device->pwfx->nBlockAlign;
 
 	TRACE("sec_mixpos=%d/%d\n", dsb->sec_mixpos, dsb->buflen);
 	TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
@@ -481,15 +445,13 @@ static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWO
 	/* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
 	oldpos = dsb->sec_mixpos;
 
-	DSOUND_MixToTemporary(dsb, len);
+	DSOUND_MixToTemporary(dsb, frames);
 	ibuf = dsb->device->tmp_buffer;
 
 	/* Apply volume if needed */
-	volbuf = DSOUND_MixerVol(dsb, len);
-	if (volbuf)
-		ibuf = volbuf;
+	DSOUND_MixerVol(dsb, frames);
 
-	dsb->device->mixfunction(ibuf, dsb->device->mix_buffer, len);
+	mixieee32(ibuf, dsb->device->mix_buffer, frames * dsb->device->pwfx->nChannels);
 
 	/* check for notification positions */
 	if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
@@ -694,6 +656,14 @@ static void DSOUND_WaveQueue(DirectSoundDevice *device, BOOL force)
  * Perform mixing for a Direct Sound device. That is, go through all the
  * secondary buffers (the sound bites currently playing) and mix them in
  * to the primary buffer (the device buffer).
+ *
+ * The mixing procedure goes:
+ *
+ * secondary->buffer (secondary format)
+ *   =[Resample]=> device->tmp_buffer (float format)
+ *   =[Volume]=> device->tmp_buffer (float format)
+ *   =[Mix]=> device->mix_buffer (float format)
+ *   =[Reformat]=> device->buffer (device format)
  */
 static void DSOUND_PerformMix(DirectSoundDevice *device)
 {
diff --git a/dlls/dsound/primary.c b/dlls/dsound/primary.c
index 6633124f876..b3b77f9cda4 100644
--- a/dlls/dsound/primary.c
+++ b/dlls/dsound/primary.c
@@ -189,13 +189,10 @@ static HRESULT DSOUND_PrimaryOpen(DirectSoundDevice *device)
     if(device->pwfx->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
             (device->pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
              IsEqualGUID(&((WAVEFORMATEXTENSIBLE*)device->pwfx)->SubFormat,
-                 &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))){
-        device->mixfunction = mixfunctions[4];
+                 &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)))
         device->normfunction = normfunctions[4];
-    }else{
-        device->mixfunction = mixfunctions[device->pwfx->wBitsPerSample/8 - 1];
+    else
         device->normfunction = normfunctions[device->pwfx->wBitsPerSample/8 - 1];
-    }
 
 	FillMemory(device->buffer, device->buflen, (device->pwfx->wBitsPerSample == 8) ? 128 : 0);
 	FillMemory(device->mix_buffer, device->mix_buffer_len, 0);
@@ -512,16 +509,14 @@ opened:
 			WARN("DSOUND_PrimaryOpen(2) failed: %08x\n", err);
 	}
 
+
 	if(device->pwfx->wFormatTag == WAVE_FORMAT_IEEE_FLOAT ||
 			(device->pwfx->wFormatTag == WAVE_FORMAT_EXTENSIBLE &&
 			 IsEqualGUID(&((WAVEFORMATEXTENSIBLE*)device->pwfx)->SubFormat,
-				 &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT))){
-		device->mixfunction = mixfunctions[4];
+				 &KSDATAFORMAT_SUBTYPE_IEEE_FLOAT)))
 		device->normfunction = normfunctions[4];
-	}else{
-		device->mixfunction = mixfunctions[device->pwfx->wBitsPerSample/8 - 1];
+	else
 		device->normfunction = normfunctions[device->pwfx->wBitsPerSample/8 - 1];
-	}
 
 	if (old_fmt->nSamplesPerSec != device->pwfx->nSamplesPerSec ||
 			old_fmt->wBitsPerSample != device->pwfx->wBitsPerSample ||