pandory500/SDL-mirror
2
0
Code Issues Pull requests Projects Releases Packages Wiki Activity Actions

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--
extra : convert_revision : svn%3Ac70aab31-4412-0410-b14c-859654838e24/trunk%402119
This commit is contained in:
Sam Lantinga 2006-09-24 15:56:36 +00:00
parent a42c7f1452
commit d0a80f6b53
1 changed files with 18 additions and 50 deletions
Download patch file Download diff file Expand all files Collapse all files

68
src/audio/SDL_audiocvt.c
View file

@ -45,11 +45,7 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}
@ -64,13 +60,7 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}
@ -87,14 +77,10 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}
@ -102,14 +88,10 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}
@ -127,17 +109,10 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}
@ -145,17 +120,10 @@ SDL_ConvertMono(SDL_AudioCVT * cvt, SDL_AudioFormat 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;
}