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libSDL
Commit d0a80f6b authored by Sam Lantinga's avatar Sam Lantinga
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Fixed bug #292 

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
parent a42c7f14
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Showing with 18 additions and 50 deletions
src/audio/SDL_audiocvt.c
View file @ d0a80f6b
......@@ -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;
}
......
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