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Sweden-Number/dlls/dsound/dsound_convert.c

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/* DirectSound format conversion and mixing routines
*
* Copyright 2007 Maarten Lankhorst
* Copyright 2011 Owen Rudge for CodeWeavers
*
* 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 <stdarg.h>
#include <math.h>
#include "windef.h"
#include "winbase.h"
#include "mmsystem.h"
#include "wine/debug.h"
#include "dsound.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 float get8(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
const BYTE* buf = dsb->buffer->memory;
buf += pos + channel;
return (buf[0] - 0x80) / (float)0x80;
}
static float get16(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
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 float get24(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
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 float get32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
const BYTE* buf = dsb->buffer->memory;
const LONG *sbuf = (const LONG*)(buf + pos + 4 * channel);
LONG sample = le32(*sbuf);
return sample / (float)0x80000000U;
}
static float getieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
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;
}
const bitsgetfunc getbpp[5] = {get8, get16, get24, get32, getieee32};
float get_mono(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel)
{
DWORD channels = dsb->pwfx->nChannels;
DWORD c;
float val = 0;
/* XXX: does Windows include LFE into the mix? */
for (c = 0; c < channels; c++)
val += dsb->get_aux(dsb, pos, c);
val /= channels;
return val;
}
static inline unsigned char f_to_8(float value)
{
if(value <= -1.f)
return 0;
if(value >= 1.f * 0x7f / 0x80)
return 0xFF;
return lrintf((value + 1.f) * 0x80);
}
static inline SHORT f_to_16(float value)
{
if(value <= -1.f)
return 0x8000;
if(value >= 1.f * 0x7FFF / 0x8000)
return 0x7FFF;
return le16(lrintf(value * 0x8000));
}
static LONG f_to_24(float value)
{
if(value <= -1.f)
return 0x80000000;
if(value >= 1.f * 0x7FFFFF / 0x800000)
return 0x7FFFFF00;
return lrintf(value * 0x80000000U);
}
static inline LONG f_to_32(float value)
{
if(value <= -1.f)
return 0x80000000;
if(value >= 1.f * 0x7FFFFFFF / 0x80000000U) /* this rounds to 1.f */
return 0x7FFFFFFF;
return le32(lrintf(value * 0x80000000U));
}
void putieee32(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
BYTE *buf = (BYTE *)dsb->device->tmp_buffer;
float *fbuf = (float*)(buf + pos + sizeof(float) * channel);
*fbuf = value;
}
void putieee32_sum(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
BYTE *buf = (BYTE *)dsb->device->tmp_buffer;
float *fbuf = (float*)(buf + pos + sizeof(float) * channel);
*fbuf += value;
}
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);
}
void put_mono2quad(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
dsb->put_aux(dsb, pos, 0, value);
dsb->put_aux(dsb, pos, 1, value);
dsb->put_aux(dsb, pos, 2, value);
dsb->put_aux(dsb, pos, 3, value);
}
void put_stereo2quad(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
if (channel == 0) { /* Left */
dsb->put_aux(dsb, pos, 0, value); /* Front left */
dsb->put_aux(dsb, pos, 2, value); /* Back left */
} else if (channel == 1) { /* Right */
dsb->put_aux(dsb, pos, 1, value); /* Front right */
dsb->put_aux(dsb, pos, 3, value); /* Back right */
}
}
void put_mono2surround51(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
dsb->put_aux(dsb, pos, 0, value);
dsb->put_aux(dsb, pos, 1, value);
dsb->put_aux(dsb, pos, 2, value);
dsb->put_aux(dsb, pos, 3, value);
dsb->put_aux(dsb, pos, 4, value);
dsb->put_aux(dsb, pos, 5, value);
}
void put_stereo2surround51(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
if (channel == 0) { /* Left */
dsb->put_aux(dsb, pos, 0, value); /* Front left */
dsb->put_aux(dsb, pos, 4, value); /* Back left */
dsb->put_aux(dsb, pos, 2, 0.0f); /* Mute front centre */
dsb->put_aux(dsb, pos, 3, 0.0f); /* Mute LFE */
} else if (channel == 1) { /* Right */
dsb->put_aux(dsb, pos, 1, value); /* Front right */
dsb->put_aux(dsb, pos, 5, value); /* Back right */
}
}
void put_surround512stereo(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
/* based on analyzing a recording of a dsound downmix */
switch(channel){
case 4: /* surround left */
value *= 0.24f;
dsb->put_aux(dsb, pos, 0, value);
break;
case 0: /* front left */
value *= 1.0f;
dsb->put_aux(dsb, pos, 0, value);
break;
case 5: /* surround right */
value *= 0.24f;
dsb->put_aux(dsb, pos, 1, value);
break;
case 1: /* front right */
value *= 1.0f;
dsb->put_aux(dsb, pos, 1, value);
break;
case 2: /* centre */
value *= 0.7;
dsb->put_aux(dsb, pos, 0, value);
dsb->put_aux(dsb, pos, 1, value);
break;
case 3:
/* LFE is totally ignored in dsound when downmixing to 2 channels */
break;
}
}
void put_surround712stereo(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
/* based on analyzing a recording of a dsound downmix */
switch(channel){
case 6: /* back left */
value *= 0.24f;
dsb->put_aux(dsb, pos, 0, value);
break;
case 4: /* surround left */
value *= 0.24f;
dsb->put_aux(dsb, pos, 0, value);
break;
case 0: /* front left */
value *= 1.0f;
dsb->put_aux(dsb, pos, 0, value);
break;
case 7: /* back right */
value *= 0.24f;
dsb->put_aux(dsb, pos, 1, value);
break;
case 5: /* surround right */
value *= 0.24f;
dsb->put_aux(dsb, pos, 1, value);
break;
case 1: /* front right */
value *= 1.0f;
dsb->put_aux(dsb, pos, 1, value);
break;
case 2: /* centre */
value *= 0.7;
dsb->put_aux(dsb, pos, 0, value);
dsb->put_aux(dsb, pos, 1, value);
break;
case 3:
/* LFE is totally ignored in dsound when downmixing to 2 channels */
break;
}
}
void put_quad2stereo(const IDirectSoundBufferImpl *dsb, DWORD pos, DWORD channel, float value)
{
/* based on pulseaudio's downmix algorithm */
switch(channel){
case 2: /* back left */
value *= 0.1f; /* (1/9) / (sum of left volumes) */
dsb->put_aux(dsb, pos, 0, value);
break;
case 0: /* front left */
value *= 0.9f; /* 1 / (sum of left volumes) */
dsb->put_aux(dsb, pos, 0, value);
break;
case 3: /* back right */
value *= 0.1f; /* (1/9) / (sum of right volumes) */
dsb->put_aux(dsb, pos, 1, value);
break;
case 1: /* front right */
value *= 0.9f; /* 1 / (sum of right volumes) */
dsb->put_aux(dsb, pos, 1, value);
break;
}
}
void mixieee32(float *src, float *dst, unsigned samples)
{
TRACE("%p - %p %d\n", src, dst, samples);
while (samples--)
*(dst++) += *(src++);
}
static void norm8(float *src, unsigned char *dst, unsigned samples)
{
TRACE("%p - %p %d\n", src, dst, samples);
while (samples--)
{
*dst = f_to_8(*src);
++dst;
++src;
}
}
static void norm16(float *src, SHORT *dst, unsigned samples)
{
TRACE("%p - %p %d\n", src, dst, samples);
while (samples--)
{
*dst = f_to_16(*src);
++dst;
++src;
}
}
static void norm24(float *src, BYTE *dst, unsigned samples)
{
TRACE("%p - %p %d\n", src, dst, samples);
while (samples--)
{
LONG t = f_to_24(*src);
dst[0] = (t >> 8) & 0xFF;
dst[1] = (t >> 16) & 0xFF;
dst[2] = t >> 24;
dst += 3;
++src;
}
}
static void norm32(float *src, INT *dst, unsigned samples)
{
TRACE("%p - %p %d\n", src, dst, samples);
while (samples--)
{
*dst = f_to_32(*src);
++dst;
++src;
}
}
const normfunc normfunctions[4] = {
(normfunc)norm8,
(normfunc)norm16,
(normfunc)norm24,
(normfunc)norm32,
};
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