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 NONAMELESSSTRUCT
43 #define NONAMELESSUNION
48 #include "wine/debug.h"
50 #include "dsound_private.h"
52 WINE_DEFAULT_DEBUG_CHANNEL(dsound
);
54 #ifdef WORDS_BIGENDIAN
55 #define le16(x) RtlUshortByteSwap((x))
56 #define le32(x) RtlUlongByteSwap((x))
62 static float get8(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
64 const BYTE
* buf
= dsb
->buffer
->memory
;
66 return (buf
[0] - 0x80) / (float)0x80;
69 static float get16(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
71 const BYTE
* buf
= dsb
->buffer
->memory
;
72 const SHORT
*sbuf
= (const SHORT
*)(buf
+ pos
+ 2 * channel
);
73 SHORT sample
= (SHORT
)le16(*sbuf
);
74 return sample
/ (float)0x8000;
77 static float get24(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
80 const BYTE
* buf
= dsb
->buffer
->memory
;
81 buf
+= pos
+ 3 * channel
;
82 /* The next expression deliberately has an overflow for buf[2] >= 0x80,
83 this is how negative values are made.
85 sample
= (buf
[0] << 8) | (buf
[1] << 16) | (buf
[2] << 24);
86 return sample
/ (float)0x80000000U
;
89 static float get32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
91 const BYTE
* buf
= dsb
->buffer
->memory
;
92 const LONG
*sbuf
= (const LONG
*)(buf
+ pos
+ 4 * channel
);
93 LONG sample
= le32(*sbuf
);
94 return sample
/ (float)0x80000000U
;
97 static float getieee32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
99 const BYTE
* buf
= dsb
->buffer
->memory
;
100 const float *sbuf
= (const float*)(buf
+ pos
+ 4 * channel
);
101 /* The value will be clipped later, when put into some non-float buffer */
105 const bitsgetfunc getbpp
[5] = {get8
, get16
, get24
, get32
, getieee32
};
107 float get_mono(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
)
109 DWORD channels
= dsb
->pwfx
->nChannels
;
112 /* XXX: does Windows include LFE into the mix? */
113 for (c
= 0; c
< channels
; c
++)
114 val
+= dsb
->get_aux(dsb
, pos
, c
);
119 static inline unsigned char f_to_8(float value
)
123 if(value
>= 1.f
* 0x7f / 0x80)
125 return lrintf((value
+ 1.f
) * 0x80);
128 static inline SHORT
f_to_16(float value
)
132 if(value
>= 1.f
* 0x7FFF / 0x8000)
134 return le16(lrintf(value
* 0x8000));
137 static LONG
f_to_24(float value
)
141 if(value
>= 1.f
* 0x7FFFFF / 0x800000)
143 return lrintf(value
* 0x80000000U
);
146 static inline LONG
f_to_32(float value
)
150 if(value
>= 1.f
* 0x7FFFFFFF / 0x80000000U
) /* this rounds to 1.f */
152 return le32(lrintf(value
* 0x80000000U
));
155 void putieee32(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
, float value
)
157 BYTE
*buf
= (BYTE
*)dsb
->device
->tmp_buffer
;
158 float *fbuf
= (float*)(buf
+ pos
+ sizeof(float) * channel
);
162 void put_mono2stereo(const IDirectSoundBufferImpl
*dsb
, DWORD pos
, DWORD channel
, float value
)
164 dsb
->put_aux(dsb
, pos
, 0, value
);
165 dsb
->put_aux(dsb
, pos
, 1, value
);
168 void mixieee32(float *src
, float *dst
, unsigned samples
)
170 TRACE("%p - %p %d\n", src
, dst
, samples
);
172 *(dst
++) += *(src
++);
175 static void norm8(float *src
, unsigned char *dst
, unsigned len
)
177 TRACE("%p - %p %d\n", src
, dst
, len
);
186 static void norm16(float *src
, SHORT
*dst
, unsigned len
)
188 TRACE("%p - %p %d\n", src
, dst
, len
);
192 *dst
= f_to_16(*src
);
198 static void norm24(float *src
, BYTE
*dst
, unsigned len
)
200 TRACE("%p - %p %d\n", src
, dst
, len
);
204 LONG t
= f_to_24(*src
);
205 dst
[0] = (t
>> 8) & 0xFF;
206 dst
[1] = (t
>> 16) & 0xFF;
213 static void norm32(float *src
, INT
*dst
, unsigned len
)
215 TRACE("%p - %p %d\n", src
, dst
, len
);
219 *dst
= f_to_32(*src
);
225 static void normieee32(float *src
, float *dst
, unsigned len
)
227 TRACE("%p - %p %d\n", src
, dst
, len
);
242 const normfunc normfunctions
[5] = {