static inline void sample_backend(int left, int right)
{
#if AUDIO_DEBUG
int nr;
for (nr = 0; nr < 4; nr++) {
struct audio_channel_data *cdp = audio_channel + nr;
if (cdp->state != 0 && cdp->datpt != 0 && (dmacon & (1 << nr)) && cdp->datpt >= cdp->datptend) {
fprintf(stderr, "Audio output overrun on channel %d: %.8x/%.8x\n", nr, cdp->datpt, cdp->datptend);
}
}
#endif
/* samples are in range -16384 (-128*64*2) and 16256 (127*64*2) */
left <<= 16 - 14 - 1;
right <<= 16 - 14 - 1;
/* [-32768, 32512] */
if (sound_use_filter) {
left = filter(left, &sound_filter_state[0]);
right = filter(right, &sound_filter_state[1]);
}
*(sndbufpt++) = left;
*(sndbufpt++) = right;
check_sound_buffers();
}
/* This interpolator examines sample points when Paula switches the output
* voltage and computes the average of Paula's output */
static void sample16i_anti_handler (void)
{
int datas[4], data1;
samplexx_anti_handler (datas);
data1 = datas[0] + datas[3] + datas[1] + datas[2];
put_sound_word_mono_func (data1);
check_sound_buffers ();
}
static void sample16si_sinc_handler (void)
{
int datas[4], data1, data2;
samplexx_sinc_handler (datas);
data1 = datas[0] + datas[3];
data2 = datas[1] + datas[2];
data1 = FINISH_DATA (data1);
data2 = FINISH_DATA (data2);
put_sound_word_stereo_func(data1, data2);
check_sound_buffers ();
}
void sample16s_handler (void)
{
uae_u32 data_l = audio_channel[0].adk_mask ? audio_channel[0].current_sample * audio_channel[0].vol : 0;
uae_u32 data_r = audio_channel[1].adk_mask ? audio_channel[1].current_sample * audio_channel[1].vol : 0;
if(audio_channel[2].adk_mask)
data_r += audio_channel[2].current_sample * audio_channel[2].vol;
if(audio_channel[3].adk_mask)
data_l += audio_channel[3].current_sample * audio_channel[3].vol;
data_l = FINISH_DATA(data_l);
data_r = FINISH_DATA(data_r);
put_sound_word_stereo_func(data_l, data_r);
check_sound_buffers();
}
static void sample16si_rh_handler (void)
{
unsigned long delta, ratio;
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 data3p = audio_channel[3].last_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
/* linear interpolation and summing up... */
delta = audio_channel[0].per;
ratio = ((audio_channel[0].evtime % delta) << 8) / delta;
data0 = (data0 * (256 - ratio) + data0p * ratio) >> 8;
delta = audio_channel[1].per;
ratio = ((audio_channel[1].evtime % delta) << 8) / delta;
data1 = (data1 * (256 - ratio) + data1p * ratio) >> 8;
delta = audio_channel[2].per;
ratio = ((audio_channel[2].evtime % delta) << 8) / delta;
data1 += (data2 * (256 - ratio) + data2p * ratio) >> 8;
delta = audio_channel[3].per;
ratio = ((audio_channel[3].evtime % delta) << 8) / delta;
data0 += (data3 * (256 - ratio) + data3p * ratio) >> 8;
data0 = FINISH_DATA (data0);
data1 = FINISH_DATA (data1);
put_sound_word_stereo_func(data0, data1);
check_sound_buffers ();
}
void sample16_handler (void)
{
uae_u32 data;
if(audio_channel[0].adk_mask)
data = audio_channel[0].current_sample * audio_channel[0].vol;
else
data = 0;
if(audio_channel[1].adk_mask)
data += audio_channel[1].current_sample * audio_channel[1].vol;
if(audio_channel[2].adk_mask)
data += audio_channel[2].current_sample * audio_channel[2].vol;
if(audio_channel[3].adk_mask)
data += audio_channel[3].current_sample * audio_channel[3].vol;
put_sound_word_mono_func (data);
check_sound_buffers ();
}
/* 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 void sample16si_crux_handler (void)
{
uae_u32 data0 = audio_channel[0].current_sample;
uae_u32 data1 = audio_channel[1].current_sample;
uae_u32 data2 = audio_channel[2].current_sample;
uae_u32 data3 = audio_channel[3].current_sample;
uae_u32 data0p = audio_channel[0].last_sample;
uae_u32 data1p = audio_channel[1].last_sample;
uae_u32 data2p = audio_channel[2].last_sample;
uae_u32 data3p = audio_channel[3].last_sample;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
DO_CHANNEL_1 (data0p, 0);
DO_CHANNEL_1 (data1p, 1);
DO_CHANNEL_1 (data2p, 2);
DO_CHANNEL_1 (data3p, 3);
data0 &= audio_channel[0].adk_mask;
data0p &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data1p &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data2p &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data3p &= audio_channel[3].adk_mask;
{
struct audio_channel_data *cdp;
unsigned long ratio, ratio1;
#define INTERVAL (scaled_sample_evtime * 3)
cdp = audio_channel + 0;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data0 = (data0 * ratio + data0p * (4096 - ratio)) >> 12;
cdp = audio_channel + 1;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data1 = (data1 * ratio + data1p * (4096 - ratio)) >> 12;
cdp = audio_channel + 2;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data2 = (data2 * ratio + data2p * (4096 - ratio)) >> 12;
cdp = audio_channel + 3;
ratio1 = cdp->per - cdp->evtime;
ratio = (ratio1 << 12) / INTERVAL;
if (cdp->evtime < scaled_sample_evtime || ratio1 >= INTERVAL)
ratio = 4096;
data3 = (data3 * ratio + data3p * (4096 - ratio)) >> 12;
}
data1 += data2;
data0 += data3;
data0 = FINISH_DATA (data0);
data1 = FINISH_DATA (data1);
put_sound_word_stereo_func(data0, data1);
check_sound_buffers ();
}
