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Fixed bug #292

Browse source 
I might be on crack here.

It looks like SDL_ConvertMono() in src/audio/SDL_audiocvt.c adds the left and
right channels of a stereo stream together, and clamps the new mono channel if
it would overflow.

Shouldn't it be dividing by 2 to average the two sample points instead of
clamping? Otherwise the mono sample point's volume doubles in the conversion.
This would also make the conversion faster, as it replaces two branches per
sample frame with a bitwise shift.

--ryan.

--HG--
branch : SDL-1.2
extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/branches/SDL-1.2%402118
This commit is contained in:
Sam Lantinga 2006-09-24 15:45:37 +00:00
parent 654f1950a3
commit 64fd559c0d
1 changed files with 18 additions and 53 deletions
Download patch file Download diff file Expand all files Collapse all files

71
src/audio/SDL_audiocvt.c
View file

@ -44,11 +44,7 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
dst = cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 255 ) {
*dst = 255;
} else {
*dst = (Uint8)sample;
}
*dst = (Uint8)(sample / 2);
src += 2;
dst += 1;
}
@ -62,14 +58,7 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
dst = (Sint8 *)cvt->buf;
for ( i=cvt->len_cvt/2; i; --i ) {
sample = src[0] + src[1];
if ( sample > 127 ) {
*dst = 127;
} else
if ( sample < -128 ) {
*dst = -128;
} else {
*dst = (Sint8)sample;
}
*dst = (Sint8)(sample / 2);
src += 2;
dst += 1;
}
@ -85,14 +74,10 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[0]<<8)|src[1])+
(Uint16)((src[2]<<8)|src[3]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
sample /= 2;
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
src += 4;
dst += 2;
}
@ -100,14 +85,10 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Uint16)((src[1]<<8)|src[0])+
(Uint16)((src[3]<<8)|src[2]);
if ( sample > 65535 ) {
dst[0] = 0xFF;
dst[1] = 0xFF;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
sample /= 2;
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
src += 4;
dst += 2;
}
@ -124,18 +105,10 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[0]<<8)|src[1])+
(Sint16)((src[2]<<8)|src[3]);
if ( sample > 32767 ) {
dst[0] = 0x7F;
dst[1] = 0xFF;
} else
if ( sample < -32768 ) {
dst[0] = 0x80;
dst[1] = 0x00;
} else {
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
}
sample /= 2;
dst[1] = (sample&0xFF);
sample >>= 8;
dst[0] = (sample&0xFF);
src += 4;
dst += 2;
}
@ -143,18 +116,10 @@ void SDLCALL SDL_ConvertMono(SDL_AudioCVT *cvt, Uint16 format)
for ( i=cvt->len_cvt/4; i; --i ) {
sample = (Sint16)((src[1]<<8)|src[0])+
(Sint16)((src[3]<<8)|src[2]);
if ( sample > 32767 ) {
dst[1] = 0x7F;
dst[0] = 0xFF;
} else
if ( sample < -32768 ) {
dst[1] = 0x80;
dst[0] = 0x00;
} else {
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
}
sample /= 2;
dst[0] = (sample&0xFF);
sample >>= 8;
dst[1] = (sample&0xFF);
src += 4;
dst += 2;
}