[PATCH] dlls/dsound: New resampler engine. (resent from different address)
Krzysztof Nikiel
knik00 at gmail.com
Mon Nov 29 05:13:39 CST 2010
Bug thread:
http://bugs.winehq.org/show_bug.cgi?id=14717
Cleaner and faster code. Sound quality much better.
------
dlls/dsound/Makefile.in | 2 +-
dlls/dsound/buffer.c | 62 +----
dlls/dsound/dsound.c | 2 -
dlls/dsound/dsound_convert.c | 435 ----------------------------------
dlls/dsound/dsound_main.c | 5 +-
dlls/dsound/dsound_private.h | 35 +--
dlls/dsound/mixer.c | 539 +++---------------------------------------
dlls/dsound/primary.c | 22 --
dlls/dsound/resample.c | 428 +++++++++++++++++++++++++++++++++
9 files changed, 494 insertions(+), 1036 deletions(-)
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diff --git a/dlls/dsound/Makefile.in b/dlls/dsound/Makefile.in
index 5addc95..ef61d2c 100644
--- a/dlls/dsound/Makefile.in
+++ b/dlls/dsound/Makefile.in
@@ -6,13 +6,13 @@ C_SRCS = \
buffer.c \
capture.c \
dsound.c \
- dsound_convert.c \
dsound_main.c \
duplex.c \
mixer.c \
primary.c \
propset.c \
regsvr.c \
+ resample.c \
sound3d.c
RC_SRCS = version.rc
diff --git a/dlls/dsound/buffer.c b/dlls/dsound/buffer.c
index 0e9096a..c596710 100644
--- a/dlls/dsound/buffer.c
+++ b/dlls/dsound/buffer.c
@@ -292,10 +292,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetFrequency(
oldFreq = This->freq;
This->freq = freq;
if (freq != oldFreq) {
- This->freqAdjust = ((DWORD64)This->freq << DSOUND_FREQSHIFT) / This->device->pwfx->nSamplesPerSec;
- This->nAvgBytesPerSec = freq * This->pwfx->nBlockAlign;
DSOUND_RecalcFormat(This);
- DSOUND_MixToTemporary(This, 0, This->buflen, FALSE);
}
RtlReleaseResource(&This->lock);
@@ -316,7 +313,6 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Play(
This->playflags = flags;
if (This->state == STATE_STOPPED && !This->hwbuf) {
- This->leadin = TRUE;
This->state = STATE_STARTING;
} else if (This->state == STATE_STOPPING)
This->state = STATE_PLAYING;
@@ -393,7 +389,6 @@ static ULONG WINAPI IDirectSoundBufferImpl_Release(LPDIRECTSOUNDBUFFER8 iface)
}
}
- HeapFree(GetProcessHeap(), 0, This->tmp_buffer);
HeapFree(GetProcessHeap(), 0, This->notifies);
HeapFree(GetProcessHeap(), 0, This->pwfx);
HeapFree(GetProcessHeap(), 0, This);
@@ -418,7 +413,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_GetCurrentPosition(
return hres;
}
} else {
- DWORD pos = This->sec_mixpos;
+ DWORD pos = This->inpos;
/* sanity */
if (pos >= This->buflen){
@@ -569,7 +564,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Lock(
} else {
if (writecursor+writebytes <= This->buflen) {
*(LPBYTE*)lplpaudioptr1 = This->buffer->memory+writecursor;
- if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING)
+ if (This->inpos >= writecursor && This->inpos < writecursor + writebytes && This->state == STATE_PLAYING)
WARN("Overwriting mixing position, case 1\n");
*audiobytes1 = writebytes;
if (lplpaudioptr2)
@@ -583,13 +578,13 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Lock(
DWORD remainder = writebytes + writecursor - This->buflen;
*(LPBYTE*)lplpaudioptr1 = This->buffer->memory+writecursor;
*audiobytes1 = This->buflen-writecursor;
- if (This->sec_mixpos >= writecursor && This->sec_mixpos < writecursor + writebytes && This->state == STATE_PLAYING)
+ if (This->inpos >= writecursor && This->inpos < writecursor + writebytes && This->state == STATE_PLAYING)
WARN("Overwriting mixing position, case 2\n");
if (lplpaudioptr2)
*(LPBYTE*)lplpaudioptr2 = This->buffer->memory;
if (audiobytes2)
*audiobytes2 = writebytes-(This->buflen-writecursor);
- if (audiobytes2 && This->sec_mixpos < remainder && This->state == STATE_PLAYING)
+ if (audiobytes2 && This->inpos < remainder && This->state == STATE_PLAYING)
WARN("Overwriting mixing position, case 3\n");
TRACE("Locked %p(%i bytes) and %p(%i bytes) writecursor=%d\n", *(LPBYTE*)lplpaudioptr1, *audiobytes1, lplpaudioptr2 ? *(LPBYTE*)lplpaudioptr2 : NULL, audiobytes2 ? *audiobytes2: 0, writecursor);
}
@@ -612,25 +607,23 @@ static HRESULT WINAPI IDirectSoundBufferImpl_SetCurrentPosition(
/* **** */
RtlAcquireResourceExclusive(&This->lock, TRUE);
- oldpos = This->sec_mixpos;
+ oldpos = This->inpos;
/* start mixing from this new location instead */
newpos %= This->buflen;
newpos -= newpos%This->pwfx->nBlockAlign;
- This->sec_mixpos = newpos;
+ This->inpos = newpos;
+ This->infrac = 0;
/* at this point, do not attempt to reset buffers, mess with primary mix position,
or anything like that to reduce latancy. The data already prebuffered cannot be changed */
/* position HW buffer if applicable, else just start mixing from new location instead */
if (This->hwbuf) {
- hres = IDsDriverBuffer_SetPosition(This->hwbuf, This->buf_mixpos);
+ hres = IDsDriverBuffer_SetPosition(This->hwbuf, DSOUND_secpos_to_bufpos(This, newpos));
if (hres != DS_OK)
WARN("IDsDriverBuffer_SetPosition failed\n");
}
- else if (oldpos != newpos)
- /* FIXME: Perhaps add a call to DSOUND_MixToTemporary here? Not sure it's needed */
- This->buf_mixpos = DSOUND_secpos_to_bufpos(This, newpos, 0, NULL);
RtlReleaseResource(&This->lock);
/* **** */
@@ -702,7 +695,7 @@ static HRESULT WINAPI IDirectSoundBufferImpl_GetPan(
static HRESULT WINAPI IDirectSoundBufferImpl_Unlock(
LPDIRECTSOUNDBUFFER8 iface,LPVOID p1,DWORD x1,LPVOID p2,DWORD x2
) {
- IDirectSoundBufferImpl *This = (IDirectSoundBufferImpl *)iface, *iter;
+ IDirectSoundBufferImpl *This = (IDirectSoundBufferImpl *)iface;
HRESULT hres = DS_OK;
TRACE("(%p,%p,%d,%p,%d)\n", This,p1,x1,p2,x2);
@@ -719,29 +712,6 @@ static HRESULT WINAPI IDirectSoundBufferImpl_Unlock(
RtlReleaseResource(&This->lock);
/* **** */
- if (!p2)
- x2 = 0;
-
- if (!This->hwbuf && (x1 || x2))
- {
- RtlAcquireResourceShared(&This->device->buffer_list_lock, TRUE);
- LIST_FOR_EACH_ENTRY(iter, &This->buffer->buffers, IDirectSoundBufferImpl, entry )
- {
- RtlAcquireResourceShared(&iter->lock, TRUE);
- if (x1)
- {
- if(x1 + (DWORD_PTR)p1 - (DWORD_PTR)iter->buffer->memory > iter->buflen)
- hres = DSERR_INVALIDPARAM;
- else
- DSOUND_MixToTemporary(iter, (DWORD_PTR)p1 - (DWORD_PTR)iter->buffer->memory, x1, FALSE);
- }
- if (x2)
- DSOUND_MixToTemporary(iter, 0, x2, FALSE);
- RtlReleaseResource(&iter->lock);
- }
- RtlReleaseResource(&This->device->buffer_list_lock);
- }
-
return hres;
}
@@ -1078,16 +1048,9 @@ HRESULT IDirectSoundBufferImpl_Create(
list_add_head(&dsb->buffer->buffers, &dsb->entry);
FillMemory(dsb->buffer->memory, dsb->buflen, dsbd->lpwfxFormat->wBitsPerSample == 8 ? 128 : 0);
- /* It's not necessary to initialize values to zero since */
- /* we allocated this structure with HEAP_ZERO_MEMORY... */
- dsb->buf_mixpos = dsb->sec_mixpos = 0;
dsb->state = STATE_STOPPED;
- dsb->freqAdjust = ((DWORD64)dsb->freq << DSOUND_FREQSHIFT) / device->pwfx->nSamplesPerSec;
- dsb->nAvgBytesPerSec = dsb->freq *
- dsbd->lpwfxFormat->nBlockAlign;
-
- /* calculate fragment size and write lead */
+ /* calculate new format values */
DSOUND_RecalcFormat(dsb);
if (dsb->dsbd.dwFlags & DSBCAPS_CTRL3D) {
@@ -1212,14 +1175,12 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
list_add_head(&dsb->buffer->buffers, &dsb->entry);
dsb->ref = 0;
dsb->state = STATE_STOPPED;
- dsb->buf_mixpos = dsb->sec_mixpos = 0;
+ dsb->inpos = dsb->infrac = 0;
dsb->device = device;
dsb->ds3db = NULL;
dsb->iks = NULL; /* FIXME? */
dsb->secondary = NULL;
- dsb->tmp_buffer = NULL;
DSOUND_RecalcFormat(dsb);
- DSOUND_MixToTemporary(dsb, 0, dsb->buflen, FALSE);
RtlInitializeResource(&dsb->lock);
@@ -1227,7 +1188,6 @@ HRESULT IDirectSoundBufferImpl_Duplicate(
hres = DirectSoundDevice_AddBuffer(device, dsb);
if (hres != DS_OK) {
RtlDeleteResource(&dsb->lock);
- HeapFree(GetProcessHeap(),0,dsb->tmp_buffer);
list_remove(&dsb->entry);
dsb->buffer->ref--;
HeapFree(GetProcessHeap(),0,dsb->pwfx);
diff --git a/dlls/dsound/dsound.c b/dlls/dsound/dsound.c
index 83636c2..1241db6 100644
--- a/dlls/dsound/dsound.c
+++ b/dlls/dsound/dsound.c
@@ -1276,8 +1276,6 @@ ULONG DirectSoundDevice_Release(DirectSoundDevice * device)
DSOUND_renderer[device->drvdesc.dnDevNode] = NULL;
- HeapFree(GetProcessHeap(), 0, device->tmp_buffer);
- HeapFree(GetProcessHeap(), 0, device->mix_buffer);
if (device->drvdesc.dwFlags & DSDDESC_USESYSTEMMEMORY)
HeapFree(GetProcessHeap(), 0, device->buffer);
RtlDeleteResource(&device->buffer_list_lock);
diff --git a/dlls/dsound/dsound_convert.c b/dlls/dsound/dsound_convert.c
deleted file mode 100644
index 0a6e474..0000000
--- a/dlls/dsound/dsound_convert.c
+++ /dev/null
@@ -1,435 +0,0 @@
-/* DirectSound format conversion and mixing routines
- *
- * Copyright 2007 Maarten Lankhorst
- *
- * This library is free software; you can redistribute it and/or
- * modify it under the terms of the GNU Lesser General Public
- * License as published by the Free Software Foundation; either
- * version 2.1 of the License, or (at your option) any later version.
- *
- * This library is distributed in the hope that it will be useful,
- * but WITHOUT ANY WARRANTY; without even the implied warranty of
- * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
- * Lesser General Public License for more details.
- *
- * You should have received a copy of the GNU Lesser General Public
- * License along with this library; if not, write to the Free Software
- * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
- */
-
-/* 8 bits is unsigned, the rest is signed.
- * First I tried to reuse existing stuff from alsa-lib, after that
- * didn't work, I gave up and just went for individual hacks.
- *
- * 24 bit is expensive to do, due to unaligned access.
- * In dlls/winex11.drv/dib_convert.c convert_888_to_0888_asis there is a way
- * around it, but I'm happy current code works, maybe something for later.
- *
- * The ^ 0x80 flips the signed bit, this is the conversion from
- * signed (-128.. 0.. 127) to unsigned (0...255)
- * This is only temporary: All 8 bit data should be converted to signed.
- * then when fed to the sound card, it should be converted to unsigned again.
- *
- * Sound is LITTLE endian
- */
-
-#include "config.h"
-
-#include <stdarg.h>
-
-#define NONAMELESSSTRUCT
-#define NONAMELESSUNION
-#include "windef.h"
-#include "winbase.h"
-#include "mmsystem.h"
-#include "winternl.h"
-#include "wine/debug.h"
-#include "dsound.h"
-#include "dsdriver.h"
-#include "dsound_private.h"
-
-WINE_DEFAULT_DEBUG_CHANNEL(dsound);
-
-#ifdef WORDS_BIGENDIAN
-#define le16(x) RtlUshortByteSwap((x))
-#define le32(x) RtlUlongByteSwap((x))
-#else
-#define le16(x) (x)
-#define le32(x) (x)
-#endif
-
-static inline void src_advance(const void **src, UINT stride, INT *count, UINT *freqAcc, UINT adj)
-{
- *freqAcc += adj;
- if (*freqAcc >= (1 << DSOUND_FREQSHIFT))
- {
- ULONG adv = (*freqAcc >> DSOUND_FREQSHIFT);
- *freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
- *(const char **)src += adv * stride;
- *count -= adv;
- }
-}
-
-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);
- }
-}
-
-static void convert_24_to_24 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- 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);
- }
-}
-
-static void convert_24_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- 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);
- }
-}
-
-static void convert_32_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 = (le32(*(const DWORD *)src) / 16777216);
- *dst8 -= 0x80;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-static void convert_32_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;
- *dest16 = le16(le32(*(const DWORD *)src) / 65536);
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-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);
- }
-}
-
-static void convert_32_to_32 (const void *src, void *dst, UINT src_stride,
- UINT dst_stride, INT count, UINT freqAcc, UINT adj)
-{
- while (count > 0)
- {
- DWORD *dest = dst;
- *dest = *(const DWORD *)src;
-
- dst = (char *)dst + dst_stride;
- src_advance(&src, src_stride, &count, &freqAcc, adj);
- }
-}
-
-const bitsconvertfunc convertbpp[4][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 },
-};
-
-static void mix8(signed char *src, INT *dst, unsigned len)
-{
- 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);
-}
-
-static void mix16(SHORT *src, INT *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- len /= 2;
- while (len--)
- {
- *dst += le16(*src);
- ++dst; ++src;
- }
-}
-
-static void mix24(BYTE *src, INT *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;
- ++src;
- }
-}
-
-static void mix32(INT *src, LONGLONG *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- len /= 4;
- while (len--)
- *(dst++) += le32(*(src++));
-}
-
-const mixfunc mixfunctions[4] = {
- (mixfunc)mix8,
- (mixfunc)mix16,
- (mixfunc)mix24,
- (mixfunc)mix32
-};
-
-static void norm8(INT *src, signed char *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- while (len--)
- {
- *dst = (*src) + 0x80;
- if (*src < -0x80)
- *dst = 0;
- else if (*src > 0x7f)
- *dst = 0xff;
- ++dst;
- ++src;
- }
-}
-
-static void norm16(INT *src, SHORT *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- len /= 2;
- while (len--)
- {
- *dst = le16(*src);
- if (*src <= -0x8000)
- *dst = le16(0x8000);
- else if (*src > 0x7fff)
- *dst = le16(0x7fff);
- ++dst;
- ++src;
- }
-}
-
-static void norm24(INT *src, BYTE *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- len /= 3;
- while (len--)
- {
- if (*src <= -0x800000)
- {
- dst[0] = 0;
- dst[1] = 0;
- dst[2] = 0x80;
- }
- else if (*src > 0x7fffff)
- {
- dst[0] = 0xff;
- dst[1] = 0xff;
- dst[2] = 0x7f;
- }
- else
- {
- dst[0] = *src;
- dst[1] = *src >> 8;
- dst[2] = *src >> 16;
- }
- ++dst;
- ++src;
- }
-}
-
-static void norm32(LONGLONG *src, INT *dst, unsigned len)
-{
- TRACE("%p - %p %d\n", src, dst, len);
- len /= 4;
- while (len--)
- {
- *dst = le32(*src);
- if (*src <= -(LONGLONG)0x80000000)
- *dst = le32(0x80000000);
- else if (*src > 0x7fffffff)
- *dst = le32(0x7fffffff);
- ++dst;
- ++src;
- }
-}
-
-const normfunc normfunctions[4] = {
- (normfunc)norm8,
- (normfunc)norm16,
- (normfunc)norm24,
- (normfunc)norm32,
-};
diff --git a/dlls/dsound/dsound_main.c b/dlls/dsound/dsound_main.c
index 8e264c7..c81d2aa 100644
--- a/dlls/dsound/dsound_main.c
+++ b/dlls/dsound/dsound_main.c
@@ -95,7 +95,7 @@ int ds_snd_queue_max = 10;
int ds_snd_queue_min = 6;
int ds_snd_shadow_maxsize = 2;
int ds_hw_accel = DS_HW_ACCEL_FULL;
-int ds_default_sample_rate = 44100;
+int ds_default_sample_rate = 48000;
int ds_default_bits_per_sample = 16;
static int ds_default_playback;
static int ds_default_capture;
@@ -672,9 +672,12 @@ BOOL WINAPI DllMain(HINSTANCE hInstDLL, DWORD fdwReason, LPVOID lpvReserved)
DisableThreadLibraryCalls(hInstDLL);
/* Increase refcount on dsound by 1 */
GetModuleHandleExW(GET_MODULE_HANDLE_EX_FLAG_FROM_ADDRESS, (LPCWSTR)hInstDLL, &hInstDLL);
+ /* Create resampler stuff */
+ DSOUND_CreateFIR();
break;
case DLL_PROCESS_DETACH:
TRACE("DLL_PROCESS_DETACH\n");
+ DSOUND_DeleteFIR();
break;
default:
TRACE("UNKNOWN REASON\n");
diff --git a/dlls/dsound/dsound_private.h b/dlls/dsound/dsound_private.h
index 1b25ddd..4ab8049 100644
--- a/dlls/dsound/dsound_private.h
+++ b/dlls/dsound/dsound_private.h
@@ -63,13 +63,10 @@ typedef struct SecondaryBufferImpl SecondaryBufferImpl;
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[4][4];
-typedef void (*mixfunc)(const void *, void *, unsigned);
-extern const mixfunc mixfunctions[4];
-typedef void (*normfunc)(const void *, void *, unsigned);
-extern const normfunc normfunctions[4];
+/* resample.c */
+DWORD DSOUND_PullBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen);
+void DSOUND_CreateFIR(void);
+void DSOUND_DeleteFIR(void);
/*****************************************************************************
* IDirectSoundDevice implementation structure
@@ -98,14 +95,9 @@ struct DirectSoundDevice
PrimaryBufferImpl* primary;
DSBUFFERDESC dsbd;
DWORD speaker_config;
- LPBYTE tmp_buffer, mix_buffer;
- DWORD tmp_buffer_len, mix_buffer_len;
DSVOLUMEPAN volpan;
- mixfunc mixfunction;
- normfunc normfunction;
-
/* DirectSound3DListener fields */
IDirectSound3DListenerImpl* listener;
DS3DLISTENER ds3dl;
@@ -171,17 +163,17 @@ struct IDirectSoundBufferImpl
PIDSDRIVERBUFFER hwbuf;
PWAVEFORMATEX pwfx;
BufferMemory* buffer;
- LPBYTE tmp_buffer;
- DWORD playflags,state,leadin;
+ DWORD playflags,state;
DWORD writelead,buflen;
DWORD nAvgBytesPerSec;
- DWORD freq, tmp_buffer_len, max_buffer_len;
+ DWORD freq;
DSVOLUMEPAN volpan;
DSBUFFERDESC dsbd;
- /* used for frequency conversion (PerfectPitch) */
- ULONG freqneeded, freqAdjust, freqAcc, freqAccNext, resampleinmixer;
- /* used for mixing */
- DWORD primary_mixpos, buf_mixpos, sec_mixpos;
+
+ /* resampler fields */
+ DWORD outfreq, inpos, infrac;
+ DOUBLE outfreq_1;
+
/* IDirectSoundNotifyImpl fields */
IDirectSoundNotifyImpl* notify;
@@ -197,7 +189,6 @@ struct IDirectSoundBufferImpl
/* IKsPropertySet fields */
IKsBufferPropertySetImpl* iks;
- bitsconvertfunc convert;
struct list entry;
};
@@ -385,13 +376,11 @@ HRESULT DSOUND_ReopenDevice(DirectSoundDevice *device, BOOL forcewave);
HRESULT DSOUND_FullDuplexCreate(REFIID riid, LPDIRECTSOUNDFULLDUPLEX* ppDSFD);
/* mixer.c */
-DWORD DSOUND_bufpos_to_mixpos(const DirectSoundDevice* device, DWORD pos);
void DSOUND_CheckEvent(const IDirectSoundBufferImpl *dsb, DWORD playpos, int len);
void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan);
void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan);
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb);
-void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen, BOOL inmixer);
-DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot);
+DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos);
void CALLBACK DSOUND_timer(UINT timerID, UINT msg, DWORD_PTR dwUser, DWORD_PTR dw1, DWORD_PTR dw2);
void CALLBACK DSOUND_callback(HWAVEOUT hwo, UINT msg, DWORD_PTR dwUser, DWORD_PTR dw1, DWORD_PTR dw2);
diff --git a/dlls/dsound/mixer.c b/dlls/dsound/mixer.c
index 185df6e..6b58eab 100644
--- a/dlls/dsound/mixer.c
+++ b/dlls/dsound/mixer.c
@@ -41,36 +41,40 @@ WINE_DEFAULT_DEBUG_CHANNEL(dsound);
void DSOUND_RecalcVolPan(PDSVOLUMEPAN volpan)
{
double temp;
- TRACE("(%p)\n",volpan);
- TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
+ /* 0.01dB units: vol=100*20*log(amp); amp = 10^(vol/2000) */
+ temp = pow(10.0, volpan->lVolume / 2e3);
+
/* the AmpFactors are expressed in 16.16 fixed point */
- volpan->dwVolAmpFactor = (ULONG) (pow(2.0, volpan->lVolume / 600.0) * 0xffff);
- /* FIXME: dwPan{Left|Right}AmpFactor */
+ volpan->dwVolAmpFactor = (DWORD)(temp*(DOUBLE)0x10000);
+ volpan->dwTotalLeftAmpFactor = volpan->dwVolAmpFactor;
+ volpan->dwTotalRightAmpFactor = volpan->dwVolAmpFactor;
- /* FIXME: use calculated vol and pan ampfactors */
- temp = (double) (volpan->lVolume - (volpan->lPan > 0 ? volpan->lPan : 0));
- volpan->dwTotalLeftAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
- temp = (double) (volpan->lVolume + (volpan->lPan < 0 ? volpan->lPan : 0));
- volpan->dwTotalRightAmpFactor = (ULONG) (pow(2.0, temp / 600.0) * 0xffff);
+ if (volpan->lPan > 0) /* left channel attenuated */
+ volpan->dwTotalLeftAmpFactor = (DWORD)(temp*(DOUBLE)0x10000
+ * pow(10.0, -volpan->lPan / 2e3));
- TRACE("left = %x, right = %x\n", volpan->dwTotalLeftAmpFactor, volpan->dwTotalRightAmpFactor);
+ if (volpan->lPan < 0) /* right channel attenuated */
+ volpan->dwTotalRightAmpFactor = (DWORD)(temp*(DOUBLE)0x10000
+ * pow(10.0, volpan->lPan / 2e3));
}
void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
{
double left,right;
- TRACE("(%p)\n",volpan);
- TRACE("left=%x, right=%x\n",volpan->dwTotalLeftAmpFactor,volpan->dwTotalRightAmpFactor);
- if (volpan->dwTotalLeftAmpFactor==0)
- left=-10000;
+#define LOG10(x) log(x)/log(10)
+
+ if (volpan->dwTotalLeftAmpFactor == 0)
+ left = -10000;
else
- left=600 * log(((double)volpan->dwTotalLeftAmpFactor) / 0xffff) / log(2);
- if (volpan->dwTotalRightAmpFactor==0)
- right=-10000;
+ left = 2e3 * LOG10((DOUBLE)volpan->dwTotalLeftAmpFactor / 0x10000);
+
+ if (volpan->dwTotalRightAmpFactor == 0)
+ right =- 10000;
else
- right=600 * log(((double)volpan->dwTotalRightAmpFactor) / 0xffff) / log(2);
+ right= 2e3 * LOG10((DOUBLE)volpan->dwTotalRightAmpFactor / 0x10000);
+
if (left<right)
{
volpan->lVolume=right;
@@ -90,127 +94,35 @@ void DSOUND_AmpFactorToVolPan(PDSVOLUMEPAN volpan)
TRACE("Vol=%d Pan=%d\n", volpan->lVolume, volpan->lPan);
}
-/** Convert a primary buffer position to a pointer position for device->mix_buffer
- * device: DirectSoundDevice for which to calculate
- * pos: Primary buffer position to converts
- * Returns: Offset for mix_buffer
- */
-DWORD DSOUND_bufpos_to_mixpos(const DirectSoundDevice* device, DWORD pos)
-{
- DWORD ret = pos * 32 / device->pwfx->wBitsPerSample;
- if (device->pwfx->wBitsPerSample == 32)
- ret *= 2;
- return ret;
-}
-
-/* NOTE: Not all secpos have to always be mapped to a bufpos, other way around is always the case
- * DWORD64 is used here because a single DWORD wouldn't be big enough to fit the freqAcc for big buffers
- */
/** This function converts a 'native' sample pointer to a resampled pointer that fits for primary
- * secmixpos is used to decide which freqAcc is needed
- * overshot tells what the 'actual' secpos is now (optional)
*/
-DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos, DWORD secmixpos, DWORD* overshot)
+DWORD DSOUND_secpos_to_bufpos(const IDirectSoundBufferImpl *dsb, DWORD secpos)
{
- DWORD64 framelen = secpos / dsb->pwfx->nBlockAlign;
- DWORD64 freqAdjust = dsb->freqAdjust;
- DWORD64 acc, freqAcc;
-
- if (secpos < secmixpos)
- freqAcc = dsb->freqAccNext;
- else freqAcc = dsb->freqAcc;
- acc = (framelen << DSOUND_FREQSHIFT) + (freqAdjust - 1 - freqAcc);
- acc /= freqAdjust;
- if (overshot)
- {
- DWORD64 oshot = acc * freqAdjust + freqAcc;
- assert(oshot >= framelen << DSOUND_FREQSHIFT);
- oshot -= framelen << DSOUND_FREQSHIFT;
- *overshot = (DWORD)oshot;
- assert(*overshot < dsb->freqAdjust);
- }
- return (DWORD)acc * dsb->device->pwfx->nBlockAlign;
-}
+ DWORD insample = secpos / dsb->pwfx->nBlockAlign;
+ DWORD outsample = insample * dsb->outfreq / dsb->freq;
+ DWORD bufpos = outsample * dsb->device->pwfx->nBlockAlign;
-/** Convert a resampled pointer that fits for primary to a 'native' sample pointer
- * freqAccNext is used here rather than freqAcc: In case the app wants to fill up to
- * the play position it won't overwrite it
- */
-static DWORD DSOUND_bufpos_to_secpos(const IDirectSoundBufferImpl *dsb, DWORD bufpos)
-{
- DWORD oAdv = dsb->device->pwfx->nBlockAlign, iAdv = dsb->pwfx->nBlockAlign, pos;
- DWORD64 framelen;
- DWORD64 acc;
-
- framelen = bufpos/oAdv;
- acc = framelen * (DWORD64)dsb->freqAdjust + (DWORD64)dsb->freqAccNext;
- acc = acc >> DSOUND_FREQSHIFT;
- pos = (DWORD)acc * iAdv;
- if (pos >= dsb->buflen)
- /* Because of differences between freqAcc and freqAccNext, this might happen */
- pos = dsb->buflen - iAdv;
- TRACE("Converted %d/%d to %d/%d\n", bufpos, dsb->tmp_buffer_len, pos, dsb->buflen);
- return pos;
-}
+ bufpos %= dsb->device->buflen;
-/**
- * Move freqAccNext to freqAcc, and find new values for buffer length and freqAccNext
- */
-static void DSOUND_RecalcFreqAcc(IDirectSoundBufferImpl *dsb)
-{
- if (!dsb->freqneeded) return;
- dsb->freqAcc = dsb->freqAccNext;
- dsb->tmp_buffer_len = DSOUND_secpos_to_bufpos(dsb, dsb->buflen, 0, &dsb->freqAccNext);
- TRACE("New freqadjust: %04x, new buflen: %d\n", dsb->freqAccNext, dsb->tmp_buffer_len);
+ return bufpos;
}
/**
- * Recalculate the size for temporary buffer, and new writelead
* Should be called when one of the following things occur:
* - Primary buffer format is changed
* - This buffer format (frequency) is changed
- *
- * After this, DSOUND_MixToTemporary(dsb, 0, dsb->buflen) should
- * be called to refill the temporary buffer with data.
*/
void DSOUND_RecalcFormat(IDirectSoundBufferImpl *dsb)
{
- BOOL needremix = TRUE, needresample = (dsb->freq != dsb->device->pwfx->nSamplesPerSec);
- DWORD bAlign = dsb->pwfx->nBlockAlign, pAlign = dsb->device->pwfx->nBlockAlign;
+ dsb->nAvgBytesPerSec = dsb->freq * dsb->pwfx->nBlockAlign;
- TRACE("(%p)\n",dsb);
+ dsb->outfreq = dsb->device->pwfx->nSamplesPerSec;
+ dsb->outfreq_1 = 1.0 / dsb->outfreq;
/* calculate the 10ms write lead */
dsb->writelead = (dsb->freq / 100) * dsb->pwfx->nBlockAlign;
- if ((dsb->pwfx->wBitsPerSample == dsb->device->pwfx->wBitsPerSample) &&
- (dsb->pwfx->nChannels == dsb->device->pwfx->nChannels) && !needresample)
- needremix = FALSE;
- HeapFree(GetProcessHeap(), 0, dsb->tmp_buffer);
- dsb->tmp_buffer = NULL;
- dsb->max_buffer_len = dsb->freqAcc = dsb->freqAccNext = 0;
- dsb->freqneeded = needresample;
-
- dsb->convert = convertbpp[dsb->pwfx->wBitsPerSample/8 - 1][dsb->device->pwfx->wBitsPerSample/8 - 1];
-
- dsb->resampleinmixer = FALSE;
-
- if (needremix)
- {
- if (needresample)
- DSOUND_RecalcFreqAcc(dsb);
- else
- dsb->tmp_buffer_len = dsb->buflen / bAlign * pAlign;
- dsb->max_buffer_len = dsb->tmp_buffer_len;
- if ((dsb->max_buffer_len <= dsb->device->buflen || dsb->max_buffer_len < ds_snd_shadow_maxsize * 1024 * 1024) && ds_snd_shadow_maxsize >= 0)
- dsb->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, dsb->max_buffer_len);
- if (dsb->tmp_buffer)
- FillMemory(dsb->tmp_buffer, dsb->tmp_buffer_len, dsb->device->pwfx->wBitsPerSample == 8 ? 128 : 0);
- else
- dsb->resampleinmixer = TRUE;
- }
- else dsb->max_buffer_len = dsb->tmp_buffer_len = dsb->buflen;
- dsb->buf_mixpos = DSOUND_secpos_to_bufpos(dsb, dsb->sec_mixpos, 0, NULL);
+ dsb->inpos = dsb->infrac = 0; /* reset resampler pointer */
}
/**
@@ -267,34 +179,6 @@ 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,
- 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)) {
- dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
- if (device->pwfx->nChannels == 2)
- dsb->convert(ibuf + istep, obuf + ostep, istride, ostride, count, freqAcc, adj);
- }
-
- if (device->pwfx->nChannels == 1 && dsb->pwfx->nChannels == 2)
- {
- dsb->convert(ibuf, obuf, istride, ostride, count, freqAcc, adj);
- }
-
- 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);
- }
-}
/**
* Calculate the distance between two buffer offsets, taking wraparound
@@ -311,335 +195,6 @@ static inline DWORD DSOUND_BufPtrDiff(DWORD buflen, DWORD ptr1, DWORD ptr2)
return buflen + ptr1 - ptr2;
}
}
-/**
- * Mix at most the given amount of data into the allocated temporary buffer
- * of the given secondary buffer, starting from the dsb's first currently
- * unsampled frame (writepos), translating frequency (pitch), stereo/mono
- * and bits-per-sample so that it is ideal for the primary buffer.
- * Doesn't perform any mixing - this is a straight copy/convert operation.
- *
- * dsb = the secondary buffer
- * writepos = Starting position of changed buffer
- * len = number of bytes to resample from writepos
- *
- * NOTE: writepos + len <= buflen. When called by mixer, MixOne makes sure of this.
- */
-void DSOUND_MixToTemporary(const IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD len, BOOL inmixer)
-{
- INT size;
- BYTE *ibp, *obp, *obp_begin;
- INT iAdvance = dsb->pwfx->nBlockAlign;
- INT oAdvance = dsb->device->pwfx->nBlockAlign;
- DWORD freqAcc, target_writepos = 0, overshot, maxlen;
-
- /* We resample only when needed */
- if ((dsb->tmp_buffer && inmixer) || (!dsb->tmp_buffer && !inmixer) || dsb->resampleinmixer != inmixer)
- return;
-
- assert(writepos + len <= dsb->buflen);
- if (inmixer && writepos + len < dsb->buflen)
- len += dsb->pwfx->nBlockAlign;
-
- maxlen = DSOUND_secpos_to_bufpos(dsb, len, 0, NULL);
-
- ibp = dsb->buffer->memory + writepos;
- if (!inmixer)
- obp_begin = dsb->tmp_buffer;
- else if (dsb->device->tmp_buffer_len < maxlen || !dsb->device->tmp_buffer)
- {
- dsb->device->tmp_buffer_len = maxlen;
- if (dsb->device->tmp_buffer)
- dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, maxlen);
- else
- dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, maxlen);
- obp_begin = dsb->device->tmp_buffer;
- }
- else
- obp_begin = dsb->device->tmp_buffer;
-
- TRACE("(%p, %p)\n", dsb, ibp);
- size = len / iAdvance;
-
- /* Check for same sample rate */
- if (dsb->freq == dsb->device->pwfx->nSamplesPerSec) {
- TRACE("(%p) Same sample rate %d = primary %d\n", dsb,
- dsb->freq, dsb->device->pwfx->nSamplesPerSec);
- obp = obp_begin;
- if (!inmixer)
- obp += writepos/iAdvance*oAdvance;
-
- cp_fields(dsb, ibp, obp, iAdvance, oAdvance, size, 0, 1 << DSOUND_FREQSHIFT);
- return;
- }
-
- /* Mix in different sample rates */
- TRACE("(%p) Adjusting frequency: %d -> %d\n", dsb, dsb->freq, dsb->device->pwfx->nSamplesPerSec);
-
- target_writepos = DSOUND_secpos_to_bufpos(dsb, writepos, dsb->sec_mixpos, &freqAcc);
- overshot = freqAcc >> DSOUND_FREQSHIFT;
- if (overshot)
- {
- if (overshot >= size)
- return;
- size -= overshot;
- writepos += overshot * iAdvance;
- if (writepos >= dsb->buflen)
- return;
- ibp = dsb->buffer->memory + writepos;
- freqAcc &= (1 << DSOUND_FREQSHIFT) - 1;
- TRACE("Overshot: %d, freqAcc: %04x\n", overshot, freqAcc);
- }
-
- if (!inmixer)
- obp = obp_begin + target_writepos;
- else obp = obp_begin;
-
- /* 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);
-}
-
-/** 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)
-{
- INT i;
- BYTE *bpc;
- INT16 *bps, *mems;
- DWORD vLeft, vRight;
- INT nChannels = dsb->device->pwfx->nChannels;
- LPBYTE mem = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos;
-
- if (dsb->resampleinmixer)
- mem = dsb->device->tmp_buffer;
-
- TRACE("(%p,%d)\n",dsb,len);
- 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 */
-
- if (nChannels != 1 && nChannels != 2)
- {
- FIXME("There is no support for %d channels\n", nChannels);
- return NULL;
- }
-
- 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;
- }
-
- if (dsb->device->tmp_buffer_len < len || !dsb->device->tmp_buffer)
- {
- /* If we just resampled in DSOUND_MixToTemporary, we shouldn't need to resize here */
- assert(!dsb->resampleinmixer);
- dsb->device->tmp_buffer_len = len;
- if (dsb->device->tmp_buffer)
- dsb->device->tmp_buffer = HeapReAlloc(GetProcessHeap(), 0, dsb->device->tmp_buffer, len);
- else
- dsb->device->tmp_buffer = HeapAlloc(GetProcessHeap(), 0, len);
- }
-
- 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;
- }
- 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;
-}
-
-/**
- * Mix (at most) the given number of bytes into the given position of the
- * device buffer, from the secondary buffer "dsb" (starting at the current
- * mix position for that buffer).
- *
- * Returns the number of bytes actually mixed into the device buffer. This
- * will match fraglen unless the end of the secondary buffer is reached
- * (and it is not looping).
- *
- * dsb = the secondary buffer to mix from
- * writepos = position (offset) in device buffer to write at
- * fraglen = number of bytes to mix
- */
-static DWORD DSOUND_MixInBuffer(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD fraglen)
-{
- INT len = fraglen, ilen;
- BYTE *ibuf = (dsb->tmp_buffer ? dsb->tmp_buffer : dsb->buffer->memory) + dsb->buf_mixpos, *volbuf;
- DWORD oldpos, mixbufpos;
-
- TRACE("buf_mixpos=%d/%d sec_mixpos=%d/%d\n", dsb->buf_mixpos, dsb->tmp_buffer_len, dsb->sec_mixpos, dsb->buflen);
- TRACE("(%p,%d,%d)\n",dsb,writepos,fraglen);
-
- assert(dsb->buf_mixpos + len <= dsb->tmp_buffer_len);
-
- if (len % dsb->device->pwfx->nBlockAlign) {
- INT nBlockAlign = dsb->device->pwfx->nBlockAlign;
- ERR("length not a multiple of block size, len = %d, block size = %d\n", len, nBlockAlign);
- len -= len % nBlockAlign; /* data alignment */
- }
-
- /* Resample buffer to temporary buffer specifically allocated for this purpose, if needed */
- DSOUND_MixToTemporary(dsb, dsb->sec_mixpos, DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos+len) - dsb->sec_mixpos, TRUE);
- if (dsb->resampleinmixer)
- ibuf = dsb->device->tmp_buffer;
-
- /* Apply volume if needed */
- volbuf = DSOUND_MixerVol(dsb, len);
- if (volbuf)
- ibuf = volbuf;
-
- mixbufpos = DSOUND_bufpos_to_mixpos(dsb->device, writepos);
- /* Now mix the temporary buffer into the devices main buffer */
- if ((writepos + len) <= dsb->device->buflen)
- dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, len);
- else
- {
- DWORD todo = dsb->device->buflen - writepos;
- dsb->device->mixfunction(ibuf, dsb->device->mix_buffer + mixbufpos, todo);
- dsb->device->mixfunction(ibuf + todo, dsb->device->mix_buffer, len - todo);
- }
-
- oldpos = dsb->sec_mixpos;
- dsb->buf_mixpos += len;
-
- if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
- if (dsb->buf_mixpos > dsb->tmp_buffer_len)
- ERR("Mixpos (%u) past buflen (%u), capping...\n", dsb->buf_mixpos, dsb->tmp_buffer_len);
- if (dsb->playflags & DSBPLAY_LOOPING) {
- dsb->buf_mixpos -= dsb->tmp_buffer_len;
- } else if (dsb->buf_mixpos >= dsb->tmp_buffer_len) {
- dsb->buf_mixpos = dsb->sec_mixpos = 0;
- dsb->state = STATE_STOPPED;
- }
- DSOUND_RecalcFreqAcc(dsb);
- }
-
- dsb->sec_mixpos = DSOUND_bufpos_to_secpos(dsb, dsb->buf_mixpos);
- ilen = DSOUND_BufPtrDiff(dsb->buflen, dsb->sec_mixpos, oldpos);
- /* check for notification positions */
- if (dsb->dsbd.dwFlags & DSBCAPS_CTRLPOSITIONNOTIFY &&
- dsb->state != STATE_STARTING) {
- DSOUND_CheckEvent(dsb, oldpos, ilen);
- }
-
- /* increase mix position */
- dsb->primary_mixpos += len;
- if (dsb->primary_mixpos >= dsb->device->buflen)
- dsb->primary_mixpos -= dsb->device->buflen;
- return len;
-}
-
-/**
- * Mix some frames from the given secondary buffer "dsb" into the device
- * primary buffer.
- *
- * dsb = the secondary buffer
- * playpos = the current play position in the device buffer (primary buffer)
- * writepos = the current safe-to-write position in the device buffer
- * mixlen = the maximum number of bytes in the primary buffer to mix, from the
- * current writepos.
- *
- * Returns: the number of bytes beyond the writepos that were mixed.
- */
-static DWORD DSOUND_MixOne(IDirectSoundBufferImpl *dsb, DWORD writepos, DWORD mixlen)
-{
- /* The buffer's primary_mixpos may be before or after the device
- * buffer's mixpos, but both must be ahead of writepos. */
- DWORD primary_done;
-
- TRACE("(%p,%d,%d)\n",dsb,writepos,mixlen);
- TRACE("writepos=%d, buf_mixpos=%d, primary_mixpos=%d, mixlen=%d\n", writepos, dsb->buf_mixpos, dsb->primary_mixpos, mixlen);
- TRACE("looping=%d, leadin=%d, buflen=%d\n", dsb->playflags, dsb->leadin, dsb->tmp_buffer_len);
-
- /* If leading in, only mix about 20 ms, and 'skip' mixing the rest, for more fluid pointer advancement */
- if (dsb->leadin && dsb->state == STATE_STARTING)
- {
- if (mixlen > 2 * dsb->device->fraglen)
- {
- dsb->primary_mixpos += mixlen - 2 * dsb->device->fraglen;
- dsb->primary_mixpos %= dsb->device->buflen;
- }
- }
- dsb->leadin = FALSE;
-
- /* calculate how much pre-buffering has already been done for this buffer */
- primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
-
- /* sanity */
- if(mixlen < primary_done)
- {
- /* Should *NEVER* happen */
- ERR("Fatal error. Under/Overflow? primary_done=%d, mixpos=%d/%d (%d/%d), primary_mixpos=%d, writepos=%d, mixlen=%d\n", primary_done,dsb->buf_mixpos,dsb->tmp_buffer_len,dsb->sec_mixpos, dsb->buflen, dsb->primary_mixpos, writepos, mixlen);
- dsb->primary_mixpos = writepos + mixlen;
- dsb->primary_mixpos %= dsb->device->buflen;
- return mixlen;
- }
-
- /* take into account already mixed data */
- mixlen -= primary_done;
-
- TRACE("primary_done=%d, mixlen (primary) = %i\n", primary_done, mixlen);
-
- if (!mixlen)
- return primary_done;
-
- /* First try to mix to the end of the buffer if possible
- * Theoretically it would allow for better optimization
- */
- if (mixlen + dsb->buf_mixpos >= dsb->tmp_buffer_len)
- {
- DWORD newmixed, mixfirst = dsb->tmp_buffer_len - dsb->buf_mixpos;
- newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
- mixlen -= newmixed;
-
- if (dsb->playflags & DSBPLAY_LOOPING)
- while (newmixed && mixlen)
- {
- mixfirst = (dsb->tmp_buffer_len < mixlen ? dsb->tmp_buffer_len : mixlen);
- newmixed = DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixfirst);
- mixlen -= newmixed;
- }
- }
- else DSOUND_MixInBuffer(dsb, dsb->primary_mixpos, mixlen);
-
- /* re-calculate the primary done */
- primary_done = DSOUND_BufPtrDiff(dsb->device->buflen, dsb->primary_mixpos, writepos);
-
- TRACE("new primary_mixpos=%d, total mixed data=%d\n", dsb->primary_mixpos, primary_done);
-
- /* Report back the total prebuffered amount for this buffer */
- return primary_done;
-}
/**
* For a DirectSoundDevice, go through all the currently playing buffers and
@@ -685,17 +240,15 @@ static DWORD DSOUND_MixToPrimary(const DirectSoundDevice *device, DWORD writepos
DSOUND_CheckEvent(dsb, 0, 0);
} else if (dsb->state != STATE_STOPPED) {
- /* if recovering, reset the mix position */
- if ((dsb->state == STATE_STARTING) || recover) {
- dsb->primary_mixpos = writepos;
- }
-
/* if the buffer was starting, it must be playing now */
if (dsb->state == STATE_STARTING)
dsb->state = STATE_PLAYING;
/* mix next buffer into the main buffer */
- len = DSOUND_MixOne(dsb, writepos, mixlen);
+ len = DSOUND_PullBuffer(dsb, writepos, mixlen);
+ if (len != mixlen)
+ ERR("Only %d/%d bytes from buffer %p\n", len, mixlen, dsb);
+
if (!minlen) minlen = len;
@@ -795,7 +348,7 @@ static void DSOUND_PerformMix(DirectSoundDevice *device)
if (device->priolevel != DSSCL_WRITEPRIMARY) {
BOOL recover = FALSE, all_stopped = FALSE;
- DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2, mixplaypos, mixplaypos2;
+ DWORD playpos, writepos, writelead, maxq, frag, prebuff_max, prebuff_left, size1, size2;
LPVOID buf1, buf2;
BOOL lock = (device->hwbuf && !(device->drvdesc.dwFlags & DSDDESC_DONTNEEDPRIMARYLOCK));
BOOL mustlock = FALSE;
@@ -814,9 +367,6 @@ static void DSOUND_PerformMix(DirectSoundDevice *device)
playpos,writepos,device->playpos,device->mixpos,device->buflen);
assert(device->playpos < device->buflen);
- mixplaypos = DSOUND_bufpos_to_mixpos(device, device->playpos);
- mixplaypos2 = DSOUND_bufpos_to_mixpos(device, playpos);
-
/* calc maximum prebuff */
prebuff_max = (device->prebuf * device->fraglen);
if (!device->hwbuf && playpos + prebuff_max >= device->helfrags * device->fraglen)
@@ -839,15 +389,12 @@ static void DSOUND_PerformMix(DirectSoundDevice *device)
/* reset mix position to write position */
device->mixpos = writepos;
- ZeroMemory(device->mix_buffer, device->mix_buffer_len);
- ZeroMemory(device->buffer, device->buflen);
+ FillMemory(device->buffer, device->buflen, nfiller);
} else if (playpos < device->playpos) {
buf1 = device->buffer + device->playpos;
buf2 = device->buffer;
size1 = device->buflen - device->playpos;
size2 = playpos;
- FillMemory(device->mix_buffer + mixplaypos, device->mix_buffer_len - mixplaypos, 0);
- FillMemory(device->mix_buffer, mixplaypos2, 0);
if (lock)
IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
FillMemory(buf1, size1, nfiller);
@@ -861,7 +408,6 @@ static void DSOUND_PerformMix(DirectSoundDevice *device)
buf2 = NULL;
size1 = playpos - device->playpos;
size2 = 0;
- FillMemory(device->mix_buffer + mixplaypos, mixplaypos2 - mixplaypos, 0);
if (lock)
IDsDriverBuffer_Lock(device->hwbuf, &buf1, &size1, &buf2, &size2, device->playpos, size1+size2, 0);
FillMemory(buf1, size1, nfiller);
@@ -891,15 +437,6 @@ static void DSOUND_PerformMix(DirectSoundDevice *device)
/* do the mixing */
frag = DSOUND_MixToPrimary(device, writepos, maxq, mustlock, recover, &all_stopped);
- if (frag + writepos > device->buflen)
- {
- DWORD todo = device->buflen - writepos;
- device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, todo);
- device->normfunction(device->mix_buffer, device->buffer, frag - todo);
- }
- else
- device->normfunction(device->mix_buffer + DSOUND_bufpos_to_mixpos(device, writepos), device->buffer + writepos, frag);
-
/* update the mix position, taking wrap-around into account */
device->mixpos = writepos + frag;
device->mixpos %= device->buflen;
diff --git a/dlls/dsound/primary.c b/dlls/dsound/primary.c
index aa8450f..11eb003 100644
--- a/dlls/dsound/primary.c
+++ b/dlls/dsound/primary.c
@@ -200,16 +200,6 @@ static HRESULT DSOUND_PrimaryOpen(DirectSoundDevice *device)
device->prebuf = device->helfrags;
}
- device->mix_buffer_len = DSOUND_bufpos_to_mixpos(device, device->buflen);
- device->mix_buffer = HeapAlloc(GetProcessHeap(), 0, device->mix_buffer_len);
- if (!device->mix_buffer)
- {
- if (device->hwbuf)
- IDsDriverBuffer_Release(device->hwbuf);
- device->hwbuf = NULL;
- return DSERR_OUTOFMEMORY;
- }
-
if (device->state == STATE_PLAYING) device->state = STATE_STARTING;
else if (device->state == STATE_STOPPING) device->state = STATE_STOPPED;
@@ -278,10 +268,7 @@ static HRESULT DSOUND_PrimaryOpen(DirectSoundDevice *device)
TRACE("fraglen=%d, overshot=%d\n", device->fraglen, overshot);
}
- device->mixfunction = mixfunctions[device->pwfx->wBitsPerSample/8 - 1];
- 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);
device->pwplay = device->pwqueue = device->playpos = device->mixpos = 0;
return err;
}
@@ -581,22 +568,13 @@ static HRESULT DSOUND_PrimarySetFormat(DirectSoundDevice *device, LPCWAVEFORMATE
}
}
- device->mix_buffer_len = DSOUND_bufpos_to_mixpos(device, device->buflen);
- device->mix_buffer = HeapReAlloc(GetProcessHeap(), 0, device->mix_buffer, device->mix_buffer_len);
- FillMemory(device->mix_buffer, device->mix_buffer_len, 0);
- device->mixfunction = mixfunctions[device->pwfx->wBitsPerSample/8 - 1];
- device->normfunction = normfunctions[device->pwfx->wBitsPerSample/8 - 1];
-
if (nSamplesPerSec != device->pwfx->nSamplesPerSec || bpp != device->pwfx->wBitsPerSample || chans != device->pwfx->nChannels) {
IDirectSoundBufferImpl** dsb = device->buffers;
for (i = 0; i < device->nrofbuffers; i++, dsb++) {
/* **** */
RtlAcquireResourceExclusive(&(*dsb)->lock, TRUE);
- (*dsb)->freqAdjust = ((DWORD64)(*dsb)->freq << DSOUND_FREQSHIFT) / device->pwfx->nSamplesPerSec;
DSOUND_RecalcFormat((*dsb));
- DSOUND_MixToTemporary((*dsb), 0, (*dsb)->buflen, FALSE);
- (*dsb)->primary_mixpos = 0;
RtlReleaseResource(&(*dsb)->lock);
/* **** */
diff --git a/dlls/dsound/resample.c b/dlls/dsound/resample.c
new file mode 100644
index 0000000..d69292d
--- /dev/null
+++ b/dlls/dsound/resample.c
@@ -0,0 +1,428 @@
+/* DirectSound
+ *
+ * Resample core
+ * Copyright 2010 Krzysztof Nikiel
+ *
+ * Initially based on resample:
+ * http://www-ccrma.stanford.edu/~jos/resample/
+ *
+ *
+ * This library is free software; you can redistribute it and/or
+ * modify it under the terms of the GNU Lesser General Public
+ * License as published by the Free Software Foundation; either
+ * version 2.1 of the License, or (at your option) any later version.
+ *
+ * This library is distributed in the hope that it will be useful,
+ * but WITHOUT ANY WARRANTY; without even the implied warranty of
+ * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
+ * Lesser General Public License for more details.
+ *
+ * You should have received a copy of the GNU Lesser General Public
+ * License along with this library; if not, write to the Free Software
+ * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
+ */
+
+#include <stdarg.h>
+#include <math.h> /* lrint() */
+
+#define NONAMELESSSTRUCT
+#define NONAMELESSUNION
+#include "windef.h"
+#include "winbase.h"
+#include "mmsystem.h"
+#include "winternl.h"
+#include "wine/debug.h"
+#include "dsound.h"
+#include "dsdriver.h"
+#include "dsound_private.h"
+
+WINE_DEFAULT_DEBUG_CHANNEL(dsound);
+
+
+#define CLIPSAMPLE(x,max) if(x>max)x=max;if(x<-max)x=-max;
+#define MAXCHAN 16
+
+typedef double FIRfloat; /* resampling data type */
+
+/* lowpass filter Finite Impulse Response */
+static struct
+{
+ FIRfloat *data;
+ INT width;
+ INT res; /* impulse table resolution */
+} FIR =
+{
+.data = NULL,.width = 0,.res = 50};
+
+
+static inline FIRfloat getsample(LPBYTE buf, INT bps)
+{
+ FIRfloat tmp;
+
+ switch (bps)
+ {
+ case 1:
+ tmp = ((FIRfloat) (*((BYTE *) buf)) - 128.0);
+ tmp *= 256.0;
+ break;
+ case 2:
+ tmp = *((SHORT *) buf);
+ break;
+ case 3:
+ tmp = *((BYTE *) buf) | *((BYTE *) (buf + 1)) << 8 |
+ *((BYTE *) (buf + 2)) << 16;
+ tmp *= (1.0 / 256.0);
+ break;
+ case 4:
+ tmp = *((INT *) buf);
+ tmp *= (1.0 / 65536.0);
+ break;
+ }
+ return tmp;
+}
+
+static inline void putsample(LPBYTE buf, INT bps, FIRfloat smp)
+{
+ INT ismp;
+
+ switch (bps)
+ {
+ case 1:
+ ismp = lrint(smp);
+ ismp /= 0x100;
+ ismp += (INT) * ((BYTE *) buf) - 0x80;
+ CLIPSAMPLE(ismp, 0x7f);
+ *((BYTE *) buf) = ismp + 0x80;
+ break;
+ case 2:
+ ismp = lrint(smp);
+ ismp += *((SHORT *) buf);
+ CLIPSAMPLE(ismp, 0x7fff);
+ *((SHORT *) buf) = ismp;
+ break;
+ case 3:
+ CLIPSAMPLE(smp, (double) 0x7fff);
+ ismp = lrint(smp * 256.0);
+ ismp += *((BYTE *) buf) | *((BYTE *) (buf + 1)) << 8 |
+ *((BYTE *) (buf + 2)) << 16;
+ *((BYTE *) buf) = ismp & 0xff;
+ ismp >>= 8;
+ *((BYTE *) (buf + 1)) = ismp & 0xff;
+ ismp >>= 8;
+ *((BYTE *) (buf + 2)) = ismp & 0xff;
+ break;
+ case 4:
+ CLIPSAMPLE(smp, (double) 0x7fff);
+ ismp = lrint(smp * 65536.0);
+ *((INT *) buf) += ismp;
+ break;
+ }
+}
+
+
+/*
+ * Takes buffer data starting from current position and mixes new samples
+ * with primary buffer, starting at writepos.
+ * At most mixlen bytes can be mixed.
+ * 'writepos' and 'mixlen' should be block aligned.
+ */
+DWORD DSOUND_PullBuffer(IDirectSoundBufferImpl * dsb, DWORD writepos,
+ DWORD mixlen)
+{
+ INT iAdvance, oAdvance, iBPS, oBPS;
+ FIRfloat rsum[MAXCHAN], smp[MAXCHAN];
+ BYTE *bufptr;
+ DWORD outlen;
+ INT iChans, oChans;
+ INT chan;
+ INT firstep;
+ DOUBLE amp[3];
+
+ if (!FIR.data)
+ {
+ ERR("FIR data uninitialized\n");
+ return 0;
+ }
+
+ iAdvance = dsb->pwfx->nBlockAlign;
+ oAdvance = dsb->device->pwfx->nBlockAlign;
+ iBPS = dsb->pwfx->wBitsPerSample >> 3;
+ oBPS = dsb->device->pwfx->wBitsPerSample >> 3;
+ outlen = 0;
+ iChans = dsb->pwfx->nChannels;
+ oChans = dsb->device->pwfx->nChannels;
+
+ if (iChans >= MAXCHAN)
+ iChans = MAXCHAN - 1;
+ if (oChans >= MAXCHAN)
+ oChans = MAXCHAN - 1;
+
+ for (chan = 0; chan < iChans; chan++)
+ smp[chan] = 0.0;
+
+ firstep = FIR.res;
+
+ amp[0] = (DOUBLE) dsb->volpan.dwTotalLeftAmpFactor * (1.0 / 65536.0);
+ if (oChans > 1)
+ {
+ amp[1] = (DOUBLE) dsb->volpan.dwTotalRightAmpFactor * (1.0 / 65536.0);
+ if (oChans > 2)
+ amp[2] = (DOUBLE) dsb->volpan.dwVolAmpFactor * (1.0 / 65536.0);
+ }
+
+ if (dsb->freq > dsb->outfreq) /* downsampling */
+ {
+ /* move transition band below output nuquist */
+ firstep = (FIR.res * dsb->outfreq * 9) / (dsb->freq * 10);
+ /*
+ * If firstep==0 the resample loop will hang.
+ * Maybe it would be a better idea to return some error if downsample
+ * ratio is too high.
+ */
+ if (firstep < 1)
+ firstep = 1;
+
+ /* fix gain */
+ amp[0] *= (double) firstep / FIR.res;
+ amp[1] *= (double) firstep / FIR.res;
+ amp[2] *= (double) firstep / FIR.res;
+ }
+
+ while (1)
+ {
+ double firpos, frac;
+ int firpos_i;
+
+ outlen += oAdvance;
+ if (outlen > mixlen)
+ {
+ outlen -= oAdvance;
+ break;
+ }
+
+ /* just in case: check buffer alignment */
+ if ((dsb->inpos + iAdvance) > dsb->buflen)
+ {
+ TRACE("Buffer not sample aligned (%p,%d,%d)\n",
+ dsb, dsb->buflen, iAdvance);
+ dsb->inpos = 0;
+ }
+
+ /* input pointer */
+ bufptr = dsb->buffer->memory + dsb->inpos;
+
+ if (dsb->freq != dsb->outfreq)
+ {
+ firpos = (double) FIR.res * dsb->infrac * dsb->outfreq_1;
+ frac = firpos - floor(firpos);
+ firpos_i = firpos;
+
+ for (chan = 0; chan < iChans; chan++)
+ rsum[chan] = 0.0;
+ }
+ else
+ {
+ firpos_i = FIR.width - 1;
+ frac = 0.0; /* avoid compiler warning */
+ }
+
+ /* apply the lowpass impulse response to input samples */
+ while (1)
+ {
+ if (firpos_i >= (FIR.width - 1))
+ break;
+
+ /* going backward here */
+ for (chan = iChans - 1; chan >= 0; chan--)
+ {
+ bufptr -= iBPS;
+ if (bufptr < dsb->buffer->memory)
+ bufptr += dsb->buflen;
+
+ smp[chan] = getsample(bufptr, iBPS);
+ }
+
+ if (dsb->freq != dsb->outfreq)
+ {
+ double ifir;
+
+ ifir =
+ (1.0 - frac) * FIR.data[firpos_i] + frac * FIR.data[firpos_i + 1];
+
+ for (chan = 0; chan < iChans; chan++)
+ rsum[chan] += ifir * smp[chan];
+
+ firpos_i += firstep;
+ }
+ else
+ {
+ for (chan = 0; chan < iChans; chan++)
+ rsum[chan] = smp[chan];
+ break;
+ }
+ }
+
+ /* just in case: check buffer alignment */
+ if ((writepos + oAdvance) > dsb->device->buflen)
+ {
+ TRACE("Device buffer not sample aligned (%p,%d,%d)\n",
+ dsb, dsb->device->buflen, oAdvance);
+ writepos = 0;
+ }
+
+ /* output pointer */
+ bufptr = dsb->device->buffer + writepos;
+
+ for (chan = 0; chan < oChans; chan++)
+ {
+ if (chan >= iChans)
+ {
+ if (iChans > 1)
+ break;
+ if (chan > 1)
+ break;
+
+ /* convert mono input to stereo output */
+ rsum[chan] = rsum[0];
+ }
+
+ /* volume control */
+ if (chan < 2)
+ rsum[chan] *= amp[chan];
+ else
+ rsum[chan] *= amp[2];
+
+ putsample(bufptr, oBPS, rsum[chan]);
+
+ bufptr += oBPS;
+ }
+
+ /* advance pointers */
+
+ dsb->infrac += dsb->freq;
+ while (dsb->infrac >= dsb->outfreq)
+ {
+ dsb->infrac -= dsb->outfreq;
+ dsb->inpos += iAdvance;
+ if (dsb->inpos >= dsb->buflen)
+ {
+ if (dsb->playflags & DSBPLAY_LOOPING)
+ dsb->inpos -= dsb->buflen;
+ else
+ {
+ dsb->inpos = dsb->infrac = 0;
+ dsb->state = STATE_STOPPED;
+ }
+ }
+ }
+
+ writepos += oAdvance;
+ if (writepos >= dsb->device->buflen)
+ writepos -= dsb->device->buflen;
+ }
+
+ return outlen;
+}
+
+
+/**
+ * Kaiser windowed filter generator
+ */
+static double Izero(double x)
+{
+ const double IzeroEPSILON = 1e-30; /* Max error acceptable in Izero */
+ double sum, u, halfx, temp;
+ int n;
+
+ sum = u = n = 1;
+ halfx = x / 2.0;
+ do
+ {
+ temp = halfx / (double) n;
+ n += 1;
+ temp *= temp;
+ u *= temp;
+ sum += u;
+ }
+ while (u >= IzeroEPSILON * sum);
+
+ return (sum);
+}
+
+
+/*
+ * Creates new filter table if not already present.
+ * All buffers share single table.
+ */
+void DSOUND_CreateFIR(void)
+{
+ /* passband width, relative to Nyquist frequency */
+ const double passband = 0.9;
+ const static double Beta = 8;
+ const static double width0 = 32;
+ int cnt;
+ double IBeta;
+ double tmp, tmp2;
+ int half;
+ double cutoff;
+ double resamp_ratio;
+
+ if (FIR.data)
+ {
+ ERR("FIR already created?\n");
+ return;
+ }
+
+ resamp_ratio = 1.0 / FIR.res;
+
+ FIR.width = width0 / resamp_ratio;
+ FIR.width |= 1; /* odd symmetry */
+ FIR.data = HeapAlloc(GetProcessHeap(), 0, FIR.width * sizeof(FIRfloat));
+
+ half = FIR.width / 2;
+
+ IBeta = 1.0 / Izero(Beta);
+
+ cutoff = passband * resamp_ratio;
+
+ FIR.data[half] = 1.0;
+
+ for (cnt = 1; cnt <= half; cnt++)
+ {
+ tmp = (double) cnt / half;
+ tmp2 = M_PI * cnt * cutoff;
+
+ /* Kaiser windowed sinc */
+ FIR.data[half + cnt] = FIR.data[half - cnt] =
+ sin(tmp2) / tmp2 * Izero(Beta * sqrt(1.0 - tmp * tmp)) * IBeta;
+ }
+
+ /* normalize the table */
+ tmp = 0.0;
+ for (cnt = 0; cnt < FIR.width; cnt++)
+ tmp += FIR.data[cnt];
+ tmp = 1.0 * FIR.res / tmp;
+ for (cnt = 0; cnt < FIR.width; cnt++)
+ FIR.data[cnt] *= tmp;
+
+ TRACE("FIR created: %d samples\n", FIR.width);
+}
+
+
+/*
+ * Deletes FIR table.
+ */
+void DSOUND_DeleteFIR(void)
+{
+ if (!FIR.data)
+ {
+ ERR("FIR already deleted?\n");
+ return;
+ }
+
+ FIR.width = 0;
+ HeapFree(GetProcessHeap(), 0, FIR.data);
+ FIR.data = NULL;
+
+ TRACE("FIR deleted\n");
+}
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