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optimized channel volume/pan handling

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svn-id: r10028
This commit is contained in:
Max Horn 2003-09-05 23:27:11 +00:00
parent 457f2fc211
commit eac128f011
3 changed files with 38 additions and 40 deletions
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25
sound/mixer.cpp
View File

@ -68,8 +68,6 @@ public:
_volume = volume;
}
virtual void setChannelPan(const int8 pan) {
if (pan != 0)
printf("Pan set to %d\n", pan);
_pan = pan;
}
virtual int getVolume() const {
@ -503,7 +501,21 @@ void Channel::mix(int16 *data, uint len) {
destroy();
} else {
assert(_converter);
_converter->flow(*_input, data, len, getVolume(), _volume, _pan);
// The pan value ranges from -127 to +127. That's 255 different values.
// From the channel pan/volume and the global volume, we compute the
// effective volume for the left and right channel.
// Note the slightly odd divisor: the 255 reflects the fact that
// the maximal value for _volume is 255, while the 254 is there
// because the maximal left/right pan value is 2*127 = 254.
// The value getVolume() returns is in the range 0 - 256.
// Hence, the vol_l/vol_r values will be in that range, too
int vol = getVolume() * _volume;
st_volume_t vol_l = (127 - _pan) * vol / (255 * 254);
st_volume_t vol_r = (127 + _pan) * vol / (255 * 254);
_converter->flow(*_input, data, len, vol_l, vol_r);
}
}
@ -575,11 +587,10 @@ void ChannelStream::mix(int16 *data, uint len) {
if (_finished) {
destroy();
}
return;
} else {
// Invoke the parent implementation.
Channel::mix(data, len);
}
assert(_converter);
_converter->flow(*_input, data, len, getVolume(), _volume, _pan);
}
#ifdef USE_MAD

51
sound/rate.cpp
View File

@ -80,7 +80,7 @@ protected:
public:
LinearRateConverter(st_rate_t inrate, st_rate_t outrate);
int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan);
int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r);
int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) {
return (ST_SUCCESS);
}
@ -124,10 +124,10 @@ LinearRateConverter<stereo, reverseStereo>::LinearRateConverter(st_rate_t inrate
* Return number of samples processed.
*/
template<bool stereo, bool reverseStereo>
int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan)
int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r)
{
st_sample_t *ostart, *oend;
st_sample_t out[2], tmpOut;
st_sample_t out[2];
const int numChannels = stereo ? 2 : 1;
int i;
@ -159,30 +159,19 @@ int LinearRateConverter<stereo, reverseStereo>::flow(AudioInputStream &input, st
while (ipos > opos) {
// interpolate
tmpOut = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
out[0] = out[1] = (st_sample_t)((tmpOut * vol) >> 8);
out[0] = out[1] = (st_sample_t)(ilast[0] + (((icur[0] - ilast[0]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
if (stereo) {
// interpolate
tmpOut = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
// adjust volume
out[reverseStereo ? 0 : 1] = (st_sample_t)((tmpOut * vol) >> 8);
out[reverseStereo ? 0 : 1] = (st_sample_t)(ilast[1] + (((icur[1] - ilast[1]) * opos_frac + (1UL << (FRAC_BITS-1))) >> FRAC_BITS));
}
byte pan_l = abs(pan - 128);
byte pan_r = abs(pan + 128);
out[0] = (st_sample_t)((out[0] * vol_p) >> 8);
out[1] = (st_sample_t)((out[1] * vol_p) >> 8);
out[0] = (st_sample_t)((out[0] * pan_l) >> 8);
out[1] = (st_sample_t)((out[1] * pan_r) >> 8);
// output left channel
clampedAdd(*obuf++, out[0]);
clampedAdd(*obuf++, (out[0] * (int)vol_l) >> 8);
// output right channel
clampedAdd(*obuf++, out[1]);
clampedAdd(*obuf++, (out[1] * (int)vol_r) >> 8);
// Increment output position
unsigned long tmp = opos_frac + opos_inc_frac;
opos += opos_inc + (tmp >> FRAC_BITS);
@ -208,22 +197,20 @@ the_end:
template<bool stereo, bool reverseStereo>
class CopyRateConverter : public RateConverter {
public:
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan) {
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) {
int16 tmp[2];
st_size_t len = osamp;
assert(input.isStereo() == stereo);
while (!input.eos() && len--) {
tmp[0] = tmp[1] = (input.read() * vol) >> 8;
tmp[0] = tmp[1] = input.read();
if (stereo)
tmp[reverseStereo ? 0 : 1] = (input.read() * vol) >> 8;
byte pan_l = abs(pan - 128);
byte pan_r = abs(pan + 128);
tmp[0] = ((tmp[0] * vol_p) >> 8);
tmp[1] = ((tmp[1] * vol_p) >> 8);
tmp[0] = ((tmp[0] * pan_l) >> 8);
tmp[1] = ((tmp[1] * pan_r) >> 8);
clampedAdd(*obuf++, tmp[0]);
clampedAdd(*obuf++, tmp[1]);
tmp[reverseStereo ? 0 : 1] = input.read();
// output left channel
clampedAdd(*obuf++, (tmp[0] * (int)vol_l) >> 8);
// output right channel
clampedAdd(*obuf++, (tmp[1] * (int)vol_r) >> 8);
}
return (ST_SUCCESS);
}

2
sound/rate.h
View File

@ -62,7 +62,7 @@ class RateConverter {
public:
RateConverter() {}
virtual ~RateConverter() {}
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol, byte vol_p, int8 pan) = 0;
virtual int flow(AudioInputStream &input, st_sample_t *obuf, st_size_t osamp, st_volume_t vol_l, st_volume_t vol_r) = 0;
virtual int drain(st_sample_t *obuf, st_size_t osamp, st_volume_t vol) = 0;
};