[PATCH 2/4] dsound: Replace convert() functions with get() and put()
Alexander E. Patrakov
patrakov at gmail.com
Tue May 1 10:35:31 CDT 2012
This patch simplifies the sample depth conversions. This way, only 2N
conversion functions have to be written, instead of N^2. The
intermediate format is chosen to be a floating-point format, and the
formulas guarantee that, for all bit depths except 32, if one converts
the sample to float and back, he gets the original value. This property
might be important for future things like DTS passthrough. Conversely, I
don't know why it can be important for 32-bit samples, as nobody would
hear the difference for true analog signals, and no encoded digital data
formats pretend that they are a sequence of 32-bit PCM samples.
Note that Andrew Eikum has patches with further simplifications on top
of this in Bug 29585.
--
Alexander E. Patrakov
-------------- next part --------------
>From 4666ad666c70ea9c2f08927fb0d90bbc3af4bbde Mon Sep 17 00:00:00 2001
From: "Alexander E. Patrakov" <patrakov at gmail.com>
Date: Mon, 7 Nov 2011 10:25:49 -0600
Subject: [PATCH 2/7] dsound: Replace convert() functions with get() and put()
---
dlls/dsound/dsound_convert.c | 400 ++++++++----------------------------------
dlls/dsound/dsound_private.h | 12 +-
dlls/dsound/mixer.c | 77 +++++----
3 files changed, 123 insertions(+), 366 deletions(-)
diff --git a/dlls/dsound/dsound_convert.c b/dlls/dsound/dsound_convert.c
index dd982c6..b670c04 100644
--- a/dlls/dsound/dsound_convert.c
+++ b/dlls/dsound/dsound_convert.c
@@ -37,6 +37,7 @@
#include "config.h"
#include <stdarg.h>
+#include <math.h>
#define NONAMELESSSTRUCT
#define NONAMELESSUNION
@@ -58,359 +59,104 @@ WINE_DEFAULT_DEBUG_CHANNEL(dsound);
#define le32(x) (x)
#endif
-static inline void src_advance(const void **src, UINT stride, INT *count, UINT *freqAcc, UINT adj)
+static float get8(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
- *freqAcc += adj;
- if (*freqAcc >= (1 << DSOUND_FREQSHIFT))
- {
- ULONG adv = (*freqAcc >> DSOUND_FREQSHIFT);
- *freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
- *(const char **)src += adv * stride;
- }
- *count -= 1;
-}
-
-static void convert_8_to_8 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- *(BYTE *)dst = *(const BYTE *)src;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_8_to_16 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- WORD dest = *(const BYTE *)src, *dest16 = dst;
- *dest16 = le16(dest * 257 - 32768);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_8_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- BYTE dest = *(const BYTE *)src;
- BYTE *dest24 = dst;
- dest24[0] = dest;
- dest24[1] = dest;
- dest24[2] = dest - 0x80;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_8_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- DWORD dest = *(const BYTE *)src, *dest32 = dst;
- *dest32 = le32(dest * 16843009 - 2147483648U);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_16_to_8 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- BYTE *dst8 = dst;
- *dst8 = (le16(*(const WORD *)src)) / 256;
- *dst8 -= 0x80;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_16_to_16 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- *(WORD *)dst = *(const WORD *)src;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_16_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- WORD dest = le16(*(const WORD *)src);
- BYTE *dest24 = dst;
-
- dest24[0] = dest / 256;
- dest24[1] = dest;
- dest24[2] = dest / 256;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_16_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- DWORD dest = *(const WORD *)src, *dest32 = dst;
- *dest32 = dest * 65537;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_24_to_8 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- BYTE *dst8 = dst;
- *dst8 = ((const BYTE *)src)[2];
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_24_to_16 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- WORD *dest16 = dst;
- const BYTE *source = src;
- *dest16 = le16(source[2] * 256 + source[1]);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ const BYTE* buf = dsb->buffer->memory;
+ buf += pos + channel;
+ return (buf[0] - 0x80) / (float)0x80;
}
-static void convert_24_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static float get16(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
- while (count > 0)
- {
- BYTE *dest24 = dst;
- const BYTE *src24 = src;
-
- dest24[0] = src24[0];
- dest24[1] = src24[1];
- dest24[2] = src24[2];
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ const BYTE* buf = dsb->buffer->memory;
+ const SHORT *sbuf = (const SHORT*)(buf + pos + 2 * channel);
+ SHORT sample = (SHORT)le16(*sbuf);
+ return sample / (float)0x8000;
}
-static void convert_24_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static float get24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
- while (count > 0)
- {
- DWORD *dest32 = dst;
- const BYTE *source = src;
- *dest32 = le32(source[2] * 16777217 + source[1] * 65536 + source[0] * 256);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ LONG sample;
+ const BYTE* buf = dsb->buffer->memory;
+ buf += pos + 3 * channel;
+ /* The next expression deliberately has an overflow for buf[2] >= 0x80,
+ this is how negative values are made.
+ */
+ sample = (buf[0] << 8) | (buf[1] << 16) | (buf[2] << 24);
+ return sample / (float)0x80000000U;
}
-static void convert_32_to_8 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static float get32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
- while (count > 0)
- {
- BYTE *dst8 = dst;
- *dst8 = (le32(*(const DWORD *)src) / 16777216);
- *dst8 -= 0x80;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ const BYTE* buf = dsb->buffer->memory;
+ const LONG *sbuf = (const LONG*)(buf + pos + 4 * channel);
+ LONG sample = le32(*sbuf);
+ return sample / (float)0x80000000U;
}
-static void convert_32_to_16 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static float getieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
- while (count > 0)
- {
- WORD *dest16 = dst;
- *dest16 = le16(le32(*(const DWORD *)src) / 65536);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ const BYTE* buf = dsb->buffer->memory;
+ const float *sbuf = (const float*)(buf + pos + 4 * channel);
+ /* The value will be clipped later, when put into some non-float buffer */
+ return *sbuf;
}
-static void convert_32_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- DWORD dest = le32(*(const DWORD *)src);
- BYTE *dest24 = dst;
-
- dest24[0] = dest / 256;
- dest24[1] = dest / 65536;
- dest24[2] = dest / 16777216;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
+const bitsgetfunc getbpp[5] = {get8, get16, get24, get32, getieee32};
-static void convert_32_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static void put8(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
- while (count > 0)
- {
- DWORD *dest = dst;
- *dest = *(const DWORD *)src;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ BYTE* buf = dsb->device->tmp_buffer;
+ buf += pos + channel;
+ if(value <= -1.f)
+ *buf = 0;
+ else if(value >= 1.f * 0x7F / 0x80)
+ *buf = 0xFF;
+ else
+ *buf = lrintf((value + 1.f) * 0x80);
}
-static void convert_ieee_32_to_8 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static void put16(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
- while (count > 0)
- {
- DWORD src_le = le32(*(DWORD *) src);
- float v = *((float *) &src_le);
- INT8 d = 0;
-
- if (v < -1.0f)
- d = -128;
- else if (v > 1.0f)
- d = 127;
- else
- d = v * 127.5f - 0.5f;
-
- *(BYTE *) dst = d ^ 0x80;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ BYTE* buf = dsb->device->tmp_buffer;
+ SHORT *sbuf = (SHORT*)(buf + pos + 2 * channel);
+ if(value <= -1.f)
+ *sbuf = 0x8000;
+ else if(value >= 1.f * 0x7FFF / 0x8000)
+ *sbuf = 0x7FFF;
+ else
+ *sbuf = le16(lrintf(value * 0x8000));
}
-static void convert_ieee_32_to_16 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static void put24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
- while (count > 0)
- {
- DWORD src_le = le32(*(DWORD *) src);
- float v = *((float *) &src_le);
-
- INT16 *d = (INT16 *) dst;
-
- if (v < -1.0f)
- *d = -32768;
- else if (v > 1.0f)
- *d = 32767;
- else
- *d = v * 32767.5f - 0.5f;
-
- *d = le16(*d);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_ieee_32_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- DWORD src_le = le32(*(DWORD *) src);
- float v = *((float *) &src_le);
- BYTE *dest24 = dst;
-
- if (v < -1.0f)
- {
- dest24[0] = 0;
- dest24[1] = 0;
- dest24[2] = 0x80;
- }
- else if (v > 1.0f)
- {
- dest24[0] = 0xff;
- dest24[1] = 0xff;
- dest24[2] = 0x7f;
- }
- else if (v < 0.0f)
- {
- dest24[0] = v * 8388608.0f;
- dest24[1] = v * 32768.0f;
- dest24[2] = v * 128.0f;
- }
- else if (v >= 0.0f)
- {
- dest24[0] = v * 8388608.0f;
- dest24[1] = v * 32768.0f;
- dest24[2] = v * 127.0f;
- }
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ BYTE* buf = dsb->device->tmp_buffer;
+ LONG t;
+ buf += pos + 3 * channel;
+ if(value <= -1.f)
+ t = 0x80000000;
+ else if(value >= 1.f * 0x7FFFFF / 0x800000)
+ t = 0x7FFFFF00;
+ else
+ t = lrintf(value * 0x80000000U);
+ buf[0] = (t >> 8) & 0xFF;
+ buf[1] = (t >> 16) & 0xFF;
+ buf[2] = (t >> 24) & 0xFF;
}
-static void convert_ieee_32_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
+static void put32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
- while (count > 0)
- {
- DWORD src_le = le32(*(DWORD *) src);
- float v = *((float *) &src_le);
- INT32 *d = (INT32 *) dst;
-
- if (v < -1.0f)
- *d = -2147483647 - 1; /* silence warning */
- else if (v > 1.0f)
- *d = 2147483647;
- else
- *d = v * 2147483647.5f - 0.5f;
-
- *d = le32(*d);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
+ BYTE* buf = dsb->device->tmp_buffer;
+ LONG *sbuf = (LONG*)(buf + pos + 4 * channel);
+ if(value <= -1.f)
+ *sbuf = 0x80000000;
+ else if(value >= 1.f * 0x7FFFFFFF / 0x80000000U) /* this rounds to 1.f */
+ *sbuf = 0x7FFFFFFF;
+ else
+ *sbuf = le32(lrintf(value * 0x80000000U));
}
-const bitsconvertfunc convertbpp[5][4] = {
- { convert_8_to_8, convert_8_to_16, convert_8_to_24, convert_8_to_32 },
- { convert_16_to_8, convert_16_to_16, convert_16_to_24, convert_16_to_32 },
- { convert_24_to_8, convert_24_to_16, convert_24_to_24, convert_24_to_32 },
- { convert_32_to_8, convert_32_to_16, convert_32_to_24, convert_32_to_32 },
- { convert_ieee_32_to_8, convert_ieee_32_to_16, convert_ieee_32_to_24, convert_ieee_32_to_32 },
-};
+const bitsputfunc putbpp[4] = {put8, put16, put24, put32};
static void mix8(signed char *src, INT *dst, unsigned len)
{
@@ -442,7 +188,7 @@ static void mix24(BYTE *src, INT *dst, unsigned len)
if (src[2] & 0x80)
field |= 0xFF000000U;
*(dst++) += field;
- ++src;
+ src += 3;
}
}
@@ -516,7 +262,7 @@ static void norm24(INT *src, BYTE *dst, unsigned len)
dst[1] = *src >> 8;
dst[2] = *src >> 16;
}
- ++dst;
+ dst += 3;
++src;
}
}
diff --git a/dlls/dsound/dsound_private.h b/dlls/dsound/dsound_private.h
index a7209ca..9f30bff 100644
--- a/dlls/dsound/dsound_private.h
+++ b/dlls/dsound/dsound_private.h
@@ -52,8 +52,10 @@ typedef struct DirectSoundDevice DirectSoundDevice;
typedef struct DirectSoundCaptureDevice DirectSoundCaptureDevice;
/* dsound_convert.h */
-typedef void (*bitsconvertfunc)(const void *, void *, UINT, UINT, INT, UINT, UINT);
-extern const bitsconvertfunc convertbpp[5][4] DECLSPEC_HIDDEN;
+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[4] DECLSPEC_HIDDEN;
typedef void (*normfunc)(const void *, void *, unsigned);
@@ -190,8 +192,10 @@ struct IDirectSoundBufferImpl
DS3DBUFFER ds3db_ds3db;
LONG ds3db_lVolume;
BOOL ds3db_need_recalc;
- /* IKsPropertySet fields */
- bitsconvertfunc convert;
+ /* Used for bit depth conversion */
+ bitsgetfunc get;
+ bitsputfunc put;
+
struct list entry;
};
diff --git a/dlls/dsound/mixer.c b/dlls/dsound/mixer.c
index 1cb7e05..95f8e9f 100644
--- a/dlls/dsound/mixer.c
+++ b/dlls/dsound/mixer.c
@@ -200,10 +200,8 @@ void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
dsb->freqAcc = dsb->freqAccNext = 0;
dsb->freqneeded = needresample;
- if (ieee)
- dsb->convert = convertbpp[4][dsb->device->pwfx->wBitsPerSample/8 - 1];
- else
- dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];
+ dsb->get = ieee ? getbpp[4] : getbpp[dsb->pwfx->wBitsPerSample/8 - 1];
+ dsb->put = putbpp[dsb->device->pwfx->wBitsPerSample/8 - 1];
if (needremix)
{
@@ -274,37 +272,47 @@ void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len
* Copy a single frame from the given input buffer to the given output buffer.
* Translate 8 <-> 16 bits and mono <-> stereo
*/
-static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf, BYTE *obuf,
+static inline void cp_fields(const IDirectSoundBufferImpl *dsb, const BYTE *ibuf,
UINT istride, UINT ostride, UINT count, UINT freqAcc, UINT adj)
{
DirectSoundDevice *device = dsb->device;
- INT istep = dsb->pwfx->wBitsPerSample / 8, ostep = device->pwfx->wBitsPerSample / 8;
-
- if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
- (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6) ||
- (device->pwfx->nChannels == 8 && dsb->pwfx->nChannels == 2) ||
- (device->pwfx->nChannels == 6 && dsb->pwfx->nChannels == 2)) {
- dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
- if (device->pwfx->nChannels == 2 || dsb->pwfx->nChannels == 2)
- dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
- return;
- }
-
- if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
- {
- dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
- return;
+ float value;
+ ULONG adv;
+ DWORD ipos = dsb->sec_mixpos, opos = 0;
+
+ while (count-- > 0) {
+ if (device->pwfx->nChannels == dsb->pwfx->nChannels ||
+ (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 6) ||
+ (device->pwfx->nChannels == 8 && dsb->pwfx->nChannels == 2) ||
+ (device->pwfx->nChannels == 6 && dsb->pwfx->nChannels == 2)) {
+ value = dsb->get(dsb, ipos, 0);
+ dsb->put(dsb, opos, 0, value);
+ if (device->pwfx->nChannels == 2 || dsb->pwfx->nChannels == 2) {
+ value = dsb->get(dsb, ipos, 1);
+ dsb->put(dsb, opos, 1, value);
+ }
+ }
+
+ if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
+ {
+ float val = (dsb->get(dsb, ipos, 0) + dsb->get(dsb, ipos, 1)) / 2.;
+
+ dsb->put(dsb, opos, 0, val);
+ }
+
+ if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
+ {
+ value = dsb->get(dsb, ipos, 0);
+ dsb->put(dsb, opos, 0, value);
+ dsb->put(dsb, opos, 1, value);
+ }
+
+ freqAcc += adj;
+ adv = (freqAcc >> DSOUND_FREQSHIFT);
+ freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
+ ipos += adv * istride;
+ opos += ostride;
}
-
- if (device->pwfx->nChannels == 2 && dsb->pwfx->nChannels == 1)
- {
- dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
- dsb->convert(ibuf, obuf + ostep, istride, ostride, count, freqAcc, adj);
- return;
- }
-
- WARN("Unable to remap channels: device=%u, buffer=%u\n", device->pwfx->nChannels,
- dsb->pwfx->nChannels);
}
/**
@@ -338,7 +346,7 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_len)
{
INT size;
- BYTE *ibp, *obp;
+ BYTE *ibp;
INT iAdvance = dsb->pwfx->nBlockAlign;
INT oAdvance = dsb->device->pwfx->nBlockAlign;
DWORD freqAcc;
@@ -352,7 +360,6 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
else
dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, tmp_len);
}
- obp = dsb->device->tmp_buffer;
TRACE("(%p, %p)\n", dsb, ibp);
size = tmp_len / oAdvance;
@@ -362,7 +369,7 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
dsb->freq, dsb->device->pwfx->nSamplesPerSec);
- cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
+ cp_fields(dsb, ibp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
return;
}
@@ -372,7 +379,7 @@ static void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD tmp_l
DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, dsb->sec_mixpos, &freqAcc);
/* FIXME: Small problem here when we're overwriting buf_mixpos, it then STILL uses old freqAcc, not sure if it matters or not */
- cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
+ cp_fields(dsb, ibp, iAdvance, oAdvance, size, freqAcc, dsb->freqAdjust);
}
/** Apply volume to the given soundbuffer from (primary) position writepos and length len
--
1.7.8.6
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