ReceiveTempSMS
/
Sweden-Number
1
0
Fork 0
Code Issues Pull Requests Releases Wiki Activity

Use the same 8/16bit conversion routines as in pcmconverter.c. 

Reorder the two ifs in the first part of cp_fields to simplify the
code.
This commit is contained in:
Francois Gouget 2003-01-16 00:20:07 +00:00 committed by Alexandre Julliard
parent 336d8fe298
commit 8e9dcb6655
1 changed files with 28 additions and 43 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

71
dlls/dsound/mixer.c
View File

@ -117,61 +117,46 @@ void DSOUND_CheckEvent(IDirectSoundBufferImpl *dsb, int len)
}
}
/* WAV format info can be found at: */
/* */
/* http://www.cwi.nl/ftp/audio/AudioFormats.part2 */
/* ftp://ftp.cwi.nl/pub/audio/RIFF-format */
/* */
/* Import points to remember: */
/* */
/* 8-bit WAV is unsigned */
/* 16-bit WAV is signed */
static inline INT16 cvtU8toS16(BYTE byte)
/* WAV format info can be found at:
*
* http://www.cwi.nl/ftp/audio/AudioFormats.part2
* ftp://ftp.cwi.nl/pub/audio/RIFF-format
*
* Import points to remember:
* 8-bit WAV is unsigned
* 16-bit WAV is signed
*/
/* Use the same formulas as pcmconverter.c */
static inline INT16 cvtU8toS16(BYTE b)
{
INT16 s = (byte - 128) << 8;
return s;
return (short)((b+(b << 8))-32768);
}
static inline BYTE cvtS16toU8(INT16 word)
static inline BYTE cvtS16toU8(INT16 s)
{
BYTE b = (word + 32768) >> 8;
return b;
return (s >> 8) ^ (unsigned char)0x80;
}
static inline void cp_fields(const IDirectSoundBufferImpl *dsb, BYTE *ibuf, BYTE *obuf )
{
INT fl = 0, fr = 0;
if (dsb->wfx.nChannels == 2) {
if (dsb->wfx.wBitsPerSample == 8) {
INT fl,fr;
if (dsb->wfx.wBitsPerSample == 8) {
if (dsound->wfx.wBitsPerSample == 8 &&
dsound->wfx.nChannels == dsb->wfx.nChannels) {
/* avoid needless 8->16->8 conversion */
if ( (dsound->wfx.wBitsPerSample == 8) && (dsound->wfx.nChannels == 2) ) {
*obuf=*ibuf;
*obuf=*ibuf;
if (dsb->wfx.nChannels==2)
*(obuf+1)=*(ibuf+1);
return;
}
fl = cvtU8toS16(*ibuf);
fr = cvtU8toS16(*(ibuf + 1));
} else if (dsb->wfx.wBitsPerSample == 16) {
fl = *((INT16 *)ibuf);
fr = *(((INT16 *)ibuf) + 1);
}
} else if (dsb->wfx.nChannels == 1) {
if (dsb->wfx.wBitsPerSample == 8) {
/* avoid needless 8->16->8 conversion */
if ( (dsound->wfx.wBitsPerSample == 8) && (dsound->wfx.nChannels == 1) ) {
*obuf=*ibuf;
return;
}
fl = cvtU8toS16(*ibuf);
fr = fl;
} else if (dsb->wfx.wBitsPerSample == 16) {
fl = *((INT16 *)ibuf);
fr = fl;
return;
}
fl = cvtU8toS16(*ibuf);
fr = (dsb->wfx.nChannels==2 ? cvtU8toS16(*(ibuf + 1)) : fl);
} else {
fl = *((INT16 *)ibuf);
fr = (dsb->wfx.nChannels==2 ? *(((INT16 *)ibuf) + 1) : fl);
}
if (dsound->wfx.nChannels == 2) {
if (dsound->wfx.wBitsPerSample == 8) {
*obuf = cvtS16toU8(fl);