static int filter(int input, struct filter_state *fs)
{
float tmp, normal_output, led_output;
switch (sound_use_filter) {
case FILTER_MODEL_A500:
fs->rc1 = a500e_filter1_a0 * input + (1 - a500e_filter1_a0) * fs->rc1 + DENORMAL_OFFSET;
fs->rc2 = a500e_filter2_a0 * fs->rc1 + (1-a500e_filter2_a0) * fs->rc2;
normal_output = fs->rc2;
fs->rc3 = filter_a0 * normal_output + (1 - filter_a0) * fs->rc3;
fs->rc4 = filter_a0 * fs->rc3 + (1 - filter_a0) * fs->rc4;
fs->rc5 = filter_a0 * fs->rc4 + (1 - filter_a0) * fs->rc5;
led_output = fs->rc5;
break;
case FILTER_MODEL_A1200:
normal_output = input;
fs->rc2 = filter_a0 * normal_output + (1 - filter_a0) * fs->rc2 + DENORMAL_OFFSET;
fs->rc3 = filter_a0 * fs->rc2 + (1 - filter_a0) * fs->rc3;
fs->rc4 = filter_a0 * fs->rc3 + (1 - filter_a0) * fs->rc4;
led_output = fs->rc4;
break;
default:
fprintf(stderr, "Unknown filter mode\n");
exit(-1);
}
return clamp_sample(gui_ledstate ? led_output : normal_output);
}
void Resampler::clamp(Sample* Pdst, int n)
{
while (n > 0)
{
*Pdst = clamp_sample(*Pdst);
++Pdst;
n--;
}
}
/* this interpolator performs BLEP mixing (bleps are shaped like integrated sinc
* functions) with a type of BLEP that matches the filtering configuration. */
static void sample16si_sinc_handler (void)
{
int i, n;
int const *winsinc;
int datas[4];
if (sound_use_filter) {
n = (sound_use_filter == FILTER_MODEL_A500) ? 0 : 2;
if (gui_ledstate)
n += 1;
} else {
n = 4;
}
winsinc = winsinc_integral[n];
for (i = 0; i < 4; i += 1) {
int j;
struct audio_channel_data *acd = &audio_channel[i];
/* The sum rings with harmonic components up to infinity... */
int sum = acd->output_state << 17;
/* ...but we cancel them through mixing in BLEPs instead */
int offsetpos = acd->sinc_queue_head & (SINC_QUEUE_LENGTH - 1);
for (j = 0; j < SINC_QUEUE_LENGTH; j += 1) {
int age = acd->sinc_queue_time - acd->sinc_queue[offsetpos].time;
if (age >= SINC_QUEUE_MAX_AGE)
break;
sum -= winsinc[age] * acd->sinc_queue[offsetpos].output;
offsetpos = (offsetpos + 1) & (SINC_QUEUE_LENGTH - 1);
}
datas[i] = sum >> 16;
}
*(sndbufpt++) = clamp_sample(datas[0] + datas[3]);
*(sndbufpt++) = clamp_sample(datas[1] + datas[2]);
check_sound_buffers();
}
static inline void process_nibble(unsigned nibble, int* idelta,
int* sample1, int* sample2,
const ADPCMCOEFSET* coeff)
{
int sample;
int snibble;
/* nibble is in fact a signed 4 bit integer => propagate sign if needed */
snibble = (nibble & 0x08) ? (nibble - 16) : nibble;
sample = ((*sample1 * coeff->iCoef1) + (*sample2 * coeff->iCoef2)) / 256 +
snibble * *idelta;
clamp_sample(&sample);
*sample2 = *sample1;
*sample1 = sample;
*idelta = ((MS_Delta[nibble] * *idelta) / 256);
if (*idelta < 16) *idelta = 16;
}
