1 /* DirectSound format conversion and mixing routines
3 * Copyright 2007 Maarten Lankhorst
4 * Copyright 2011 Owen Rudge for CodeWeavers
6 * This library is free software; you can redistribute it and/or
7 * modify it under the terms of the GNU Lesser General Public
8 * License as published by the Free Software Foundation; either
9 * version 2.1 of the License, or (at your option) any later version.
11 * This library is distributed in the hope that it will be useful,
12 * but WITHOUT ANY WARRANTY; without even the implied warranty of
13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
14 * Lesser General Public License for more details.
16 * You should have received a copy of the GNU Lesser General Public
17 * License along with this library; if not, write to the Free Software
18 * Foundation, Inc., 51 Franklin St, Fifth Floor, Boston, MA 02110-1301, USA
21 /* 8 bits is unsigned, the rest is signed.
22 * First I tried to reuse existing stuff from alsa-lib, after that
23 * didn't work, I gave up and just went for individual hacks.
25 * 24 bit is expensive to do, due to unaligned access.
26 * In dlls/winex11.drv/dib_convert.c convert_888_to_0888_asis there is a way
27 * around it, but I'm happy current code works, maybe something for later.
29 * The ^ 0x80 flips the signed bit, this is the conversion from
30 * signed (-128.. 0.. 127) to unsigned (0...255)
31 * This is only temporary: All 8 bit data should be converted to signed.
32 * then when fed to the sound card, it should be converted to unsigned again.
34 * Sound is LITTLE endian
42 #define WIN32_NO_STATUS
44 #define COM_NO_WINDOWS_H
46 #define NONAMELESSSTRUCT
47 #define NONAMELESSUNION
52 #include <wine/debug.h>
54 #include "dsound_private.h"
56 WINE_DEFAULT_DEBUG_CHANNEL(dsound
);
58 #ifdef WORDS_BIGENDIAN
59 #define le16(x) RtlUshortByteSwap((x))
60 #define le32(x) RtlUlongByteSwap((x))
66 /* This is an inlined version of lrintf. */
69 #include <xmmintrin.h>
84 #elif defined(_M_AMD64)
85 return _mm_cvtss_si32(_mm_load_ss(&f
));
90 static float get8(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
92 const BYTE
* buf
= dsb
->buffer
->memory
;
94 return (buf
[0] - 0x80) / (float)0x80;
97 static float get16(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
99 const BYTE
* buf
= dsb
->buffer
->memory
;
100 const SHORT
*sbuf
= (const SHORT
*)(buf
+ pos
+ 2 * channel
);
101 SHORT sample
= (SHORT
)le16(*sbuf
);
102 return sample
/ (float)0x8000;
105 static float get24(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
108 const BYTE
* buf
= dsb
->buffer
->memory
;
109 buf
+= pos
+ 3 * channel
;
110 /* The next expression deliberately has an overflow for buf[2] >= 0x80,
111 this is how negative values are made.
113 sample
= (buf
[0] << 8) | (buf
[1] << 16) | (buf
[2] << 24);
114 return sample
/ (float)0x80000000U
;
117 static float get32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
119 const BYTE
* buf
= dsb
->buffer
->memory
;
120 const LONG
*sbuf
= (const LONG
*)(buf
+ pos
+ 4 * channel
);
121 LONG sample
= le32(*sbuf
);
122 return sample
/ (float)0x80000000U
;
125 static float getieee32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
127 const BYTE
* buf
= dsb
->buffer
->memory
;
128 const float *sbuf
= (const float*)(buf
+ pos
+ 4 * channel
);
129 /* The value will be clipped later, when put into some non-float buffer */
133 const bitsgetfunc getbpp
[5] = {get8
, get16
, get24
, get32
, getieee32
};
135 float get_mono(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
137 DWORD channels
= dsb
->pwfx
->nChannels
;
140 /* XXX: does Windows include LFE into the mix? */
141 for (c
= 0; c
< channels
; c
++)
142 val
+= dsb
->get_aux(dsb
, pos
, c
);
147 static inline unsigned char f_to_8(float value
)
151 if(value
>= 1.f
* 0x7f / 0x80)
153 return lrintf((value
+ 1.f
) * 0x80);
156 static inline SHORT
f_to_16(float value
)
160 if(value
>= 1.f
* 0x7FFF / 0x8000)
162 return le16(lrintf(value
* 0x8000));
165 static LONG
f_to_24(float value
)
169 if(value
>= 1.f
* 0x7FFFFF / 0x800000)
171 return lrintf(value
* 0x80000000U
);
174 static inline LONG
f_to_32(float value
)
178 if(value
>= 1.f
* 0x7FFFFFFF / 0x80000000U
) /* this rounds to 1.f */
180 return le32(lrintf(value
* 0x80000000U
));
183 void putieee32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
, float value
)
185 BYTE
*buf
= (BYTE
*)dsb
->device
->tmp_buffer
;
186 float *fbuf
= (float*)(buf
+ pos
+ sizeof(float) * channel
);
190 void put_mono2stereo(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
, float value
)
192 dsb
->put_aux(dsb
, pos
, 0, value
);
193 dsb
->put_aux(dsb
, pos
, 1, value
);
196 void mixieee32(float *src
, float *dst
, unsigned samples
)
198 TRACE("%p - %p %d\n", src
, dst
, samples
);
200 *(dst
++) += *(src
++);
203 static void norm8(float *src
, unsigned char *dst
, unsigned len
)
205 TRACE("%p - %p %d\n", src
, dst
, len
);
214 static void norm16(float *src
, SHORT
*dst
, unsigned len
)
216 TRACE("%p - %p %d\n", src
, dst
, len
);
220 *dst
= f_to_16(*src
);
226 static void norm24(float *src
, BYTE
*dst
, unsigned len
)
228 TRACE("%p - %p %d\n", src
, dst
, len
);
232 LONG t
= f_to_24(*src
);
233 dst
[0] = (t
>> 8) & 0xFF;
234 dst
[1] = (t
>> 16) & 0xFF;
241 static void norm32(float *src
, INT
*dst
, unsigned len
)
243 TRACE("%p - %p %d\n", src
, dst
, len
);
247 *dst
= f_to_32(*src
);
253 static void normieee32(float *src
, float *dst
, unsigned len
)
255 TRACE("%p - %p %d\n", src
, dst
, len
);
270 const normfunc normfunctions
[5] = {