void direct_stereo_sample8_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;
DO_CHANNEL_1 (data0, 0);
DO_CHANNEL_1 (data1, 1);
DO_CHANNEL_1 (data2, 2);
DO_CHANNEL_1 (data3, 3);
data0 &= audio_channel[0].adk_mask;
data1 &= audio_channel[1].adk_mask;
data2 &= audio_channel[2].adk_mask;
data3 &= audio_channel[3].adk_mask;
data0 += data3;
{
uae_u32 data = SBASEVAL8(1) + data0;
FINISH_DATA (8, 1);
_farnspokeb(direct_sndbufpt++, data);
}
data1 += data2;
{
uae_u32 data = SBASEVAL8(1) + data1;
FINISH_DATA (8, 1);
_farnspokeb(direct_sndbufpt++, data);
}
direct_check_sound_buffers();
eventtab[ev_sample].evtime = cycles + sample_evtime;
eventtab[ev_sample].oldcycles = cycles;
}
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 ();
}
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 ();
}
