--- uae/src/audio.c 2018/04/24 17:01:43 1.1.1.9 +++ uae/src/audio.c 2018/04/24 17:16:19 1.1.1.16 @@ -1,17 +1,19 @@ /* * UAE - The Un*x Amiga Emulator * - * OS specific functions + * Paula audio emulation * * Copyright 1995, 1996, 1997 Bernd Schmidt * Copyright 1996 Marcus Sundberg * Copyright 1996 Manfred Thole + * Copyright 2006 Toni Wilen + * + * new filter algorithm and anti&sinc interpolators by Antti S. Lankila */ #include "sysconfig.h" #include "sysdeps.h" -#include "config.h" #include "options.h" #include "memory.h" #include "custom.h" @@ -22,57 +24,73 @@ #include "events.h" #include "audio.h" #include "savestate.h" +#include "sinctable.h" +#include "gui.h" + +#define MAX_EV ~0ul + +/* periods less than this value are replaced by this value. */ +#define MIN_ALLOWED_PERIOD 16 +/* reserve ~20 extra slots in sinc queue for cpu volume or some such updates + * even at maximum period. This avoids sinc queue overflow on games like + * battle squadron that write these low period values and do cpu-based + * updates on paula registers, probably volume. */ +#define NUMBER_OF_CPU_UPDATES_ALLOWED 20 + +#define SINC_QUEUE_LENGTH (SINC_QUEUE_MAX_AGE / MIN_ALLOWED_PERIOD + NUMBER_OF_CPU_UPDATES_ALLOWED) + +typedef struct { + int age; + int output; +} sinc_queue_t; + +struct audio_channel_data { + unsigned long adk_mask; + unsigned long evtime; + unsigned long per; + uae_u8 dmaen, intreq2, data_written; + uaecptr lc, pt; + int state, wper; + unsigned int wlen; + int current_sample, last_sample; + int vol; + int *voltbl; + uae_u16 dat, nextdat, len; + int sinc_output_state; + sinc_queue_t sinc_queue[SINC_QUEUE_LENGTH]; + int sinc_queue_length; +}; -struct audio_channel_data audio_channel[6]; +static struct audio_channel_data audio_channel[4]; int sound_available = 0; int sound_table[64][256]; void (*sample_handler) (void); +static void (*sample_prehandler) (unsigned long best_evtime); -/* Zero if we want to produce Paula output, nonzero if we want to produce - AHI output. */ -static int ahi_enabled; - -/* Bit 0 is set if the right channel raised an interrupt, bit 1 is set if the - left channel raised an interrupt. */ -static int ahi_interrupt_state; - -unsigned long sample_evtime, scaled_sample_evtime; -int scaled_sample_evtime_ok; - +static unsigned long scaled_sample_evtime; static unsigned long last_cycles, next_sample_evtime; -void init_sound_table16 (void) -{ - int i,j; +unsigned int obtainedfreq; - for (i = 0; i < 256; i++) - for (j = 0; j < 64; j++) - sound_table[j][i] = j * (uae_s8)i * (currprefs.stereo ? 2 : 1); -} - -void init_sound_table8 (void) +void init_sound_table16 (void) { int i,j; for (i = 0; i < 256; i++) for (j = 0; j < 64; j++) - sound_table[j][i] = (j * (uae_s8)i * (currprefs.stereo ? 2 : 1)) / 256; + sound_table[j][i] = j * (uae_s8)i * (currprefs.sound_stereo ? 2 : 1); } -#define MULTIPLICATION_PROFITABLE - #ifdef MULTIPLICATION_PROFITABLE typedef uae_s8 sample8_t; #define DO_CHANNEL_1(v, c) do { (v) *= audio_channel[c].vol; } while (0) -#define SBASEVAL8(logn) ((logn) == 1 ? SOUND8_BASE_VAL << 7 : SOUND8_BASE_VAL << 8) #define SBASEVAL16(logn) ((logn) == 1 ? SOUND16_BASE_VAL >> 1 : SOUND16_BASE_VAL) #define FINISH_DATA(data,b,logn) do { if (14 - (b) + (logn) > 0) (data) >>= 14 - (b) + (logn); else (data) <<= (b) - 14 - (logn); } while (0); #else typedef uae_u8 sample8_t; #define DO_CHANNEL_1(v, c) do { (v) = audio_channel[c].voltbl[(v)]; } while (0) -#define SBASEVAL8(logn) SOUND8_BASE_VAL #define SBASEVAL16(logn) SOUND16_BASE_VAL -#define FINISH_DATA(b,logn) +#define FINISH_DATA(data,b,logn) #endif /* Always put the right word before the left word. */ @@ -114,177 +132,95 @@ STATIC_INLINE void put_sound_word_left ( #define DO_CHANNEL(v, c) do { (v) &= audio_channel[c].adk_mask; data += v; } while (0); -void sample16_handler (void) +static void sinc_prehandler (unsigned long best_evtime) { - if (ahi_enabled) { - uae_u16 word = audio_channel[4].current_sample + audio_channel[5].current_sample; - word >>= 1; - word += SOUND16_BASE_VAL; - PUT_SOUND_WORD (word); - } else { - 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 += data1; - data0 += data2; - data0 += data3; - { - uae_u32 data = SBASEVAL16(2) + data0; - FINISH_DATA (data, 16, 2); - PUT_SOUND_WORD (data); + int i, j, output; + struct audio_channel_data *acd; + + for (i = 0; i < 4; i++) { + acd = &audio_channel[i]; + output = (acd->current_sample * acd->vol) & acd->adk_mask; + + /* age the sinc queue and truncate it when necessary */ + for (j = 0; j < acd->sinc_queue_length; j += 1) { + acd->sinc_queue[j].age += best_evtime; + if (acd->sinc_queue[j].age >= SINC_QUEUE_MAX_AGE) { + acd->sinc_queue_length = j; + break; + } + } + /* if output state changes, record the state change and also + * write data into sinc queue for mixing in the BLEP */ + if (acd->sinc_output_state != output) { + if (acd->sinc_queue_length > SINC_QUEUE_LENGTH - 1) { + write_log ("warning: sinc queue truncated. Last age: %d.\n", + acd->sinc_queue[SINC_QUEUE_LENGTH-1].age); + acd->sinc_queue_length = SINC_QUEUE_LENGTH - 1; + } + /* make room for new and add the new value */ + memmove (&acd->sinc_queue[1], &acd->sinc_queue[0], + sizeof(acd->sinc_queue[0]) * acd->sinc_queue_length); + acd->sinc_queue_length += 1; + acd->sinc_queue[0].age = best_evtime; + acd->sinc_queue[0].output = output - acd->sinc_output_state; + acd->sinc_output_state = output; } } - check_sound_buffers (); } -void sample16i_rh_handler (void) -{ - if (ahi_enabled) { - uae_u16 word = audio_channel[4].current_sample + audio_channel[5].current_sample; - word >>= 1; - word += SOUND16_BASE_VAL; - PUT_SOUND_WORD (word); - } else { - 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; - data0 += (data1 * (256 - ratio) + data1p * ratio) >> 8; - delta = audio_channel[2].per; - ratio = ((audio_channel[2].evtime % delta) << 8) / delta; - data0 += (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; - { - uae_u32 data = SBASEVAL16(2) + data0; - FINISH_DATA (data, 16, 2); - PUT_SOUND_WORD (data); - } +/* this interpolator performs BLEP mixing (bleps are shaped like integrated sinc + * functions) with a type of BLEP that matches the filtering configuration. */ +STATIC_INLINE void samplexx_sinc_handler (int *datasp) +{ + int i, n; + int const *winsinc; + +#if 1 + /* Amiga 500 filter model is default for now. Put n=2 for A1200. */ + n = 0; + if (gui_ledstate & 1) /* power led */ + n += 1; +#else + if (sound_use_filter_sinc) { + n = (sound_use_filter_sinc == FILTER_MODEL_A500) ? 0 : 2; + if (led_filter_on) + n += 1; + } else { + n = 4; + } +#endif + winsinc = winsinc_integral[n]; - check_sound_buffers (); + for (i = 0; i < 4; i += 1) { + int j, v; + struct audio_channel_data *acd = &audio_channel[i]; + /* The sum rings with harmonic components up to infinity... */ + int sum = acd->sinc_output_state << 17; + /* ...but we cancel them through mixing in BLEPs instead */ + for (j = 0; j < acd->sinc_queue_length; j += 1) + sum -= winsinc[acd->sinc_queue[j].age] * acd->sinc_queue[j].output; + v = sum >> 17; + if (v > 32767) + v = 32767; + else if (v < -32768) + v = -32768; + datasp[i] = v; } } -void sample16i_crux_handler (void) -{ - if (ahi_enabled) { - uae_u16 word = audio_channel[4].current_sample + audio_channel[5].current_sample; - word >>= 1; - word += SOUND16_BASE_VAL; - PUT_SOUND_WORD (word); - } else { - 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 += data1; - { - uae_u32 data = SBASEVAL16(2) + data0; - FINISH_DATA (data, 16, 2); - PUT_SOUND_WORD (data); - } - } +static void sample16i_sinc_handler (void) +{ + int datas[4], data1; + + samplexx_sinc_handler (datas); + data1 = datas[0] + datas[3] + datas[1] + datas[2]; + FINISH_DATA (data1, 16, 2); + PUT_SOUND_WORD (data1); check_sound_buffers (); } -void sample8_handler (void) +void sample16_handler (void) { uae_u32 data0 = audio_channel[0].current_sample; uae_u32 data1 = audio_channel[1].current_sample; @@ -302,327 +238,406 @@ void sample8_handler (void) data0 += data2; data0 += data3; { - uae_u32 data = SBASEVAL8(2) + data0; - FINISH_DATA (data, 8, 2); - PUT_SOUND_BYTE (data); + uae_u32 data = SBASEVAL16(2) + data0; + FINISH_DATA (data, 16, 2); + PUT_SOUND_WORD (data); } - check_sound_buffers (); } -#ifdef HAVE_STEREO_SUPPORT -void sample16s_handler (void) +static void sample16i_rh_handler (void) { - if (ahi_enabled) { - PUT_SOUND_WORD_RIGHT (audio_channel[4].dat); - PUT_SOUND_WORD_LEFT (audio_channel[5].dat); - } else { - 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 = SBASEVAL16(1) + data0; - FINISH_DATA (data, 16, 1); - put_sound_word_right (data); - } + unsigned long delta, ratio; - data1 += data2; - { - uae_u32 data = SBASEVAL16(1) + data1; - FINISH_DATA (data, 16, 1); - put_sound_word_left (data); - } + 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; + data0 += (data1 * (256 - ratio) + data1p * ratio) >> 8; + delta = audio_channel[2].per; + ratio = ((audio_channel[2].evtime % delta) << 8) / delta; + data0 += (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; + + { + uae_u32 data = SBASEVAL16(2) + data0; + FINISH_DATA (data, 16, 2); + PUT_SOUND_WORD (data); } check_sound_buffers (); } -void sample16si_crux_handler (void) +static void sample16i_crux_handler (void) { - if (ahi_enabled) { - PUT_SOUND_WORD_RIGHT (audio_channel[4].dat); - PUT_SOUND_WORD_LEFT (audio_channel[5].dat); - } else { - 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); - 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; + 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; - { - uae_u32 data = SBASEVAL16 (1) + data0; - FINISH_DATA (data, 16, 1); - put_sound_word_right (data); - } + 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 += data1; + { + uae_u32 data = SBASEVAL16(2) + data0; + FINISH_DATA (data, 16, 2); + PUT_SOUND_WORD (data); + } + check_sound_buffers (); +} - { - uae_u32 data = SBASEVAL16 (1) + data1; - FINISH_DATA (data, 16, 1); - put_sound_word_left (data); - } - } +#ifdef HAVE_STEREO_SUPPORT +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]; + FINISH_DATA (data1, 16, 1); + put_sound_word_left (data1); + FINISH_DATA (data2, 16, 1); + put_sound_word_right (data2); check_sound_buffers (); } -void sample16si_rh_handler (void) +void sample16s_handler (void) { - if (ahi_enabled) { - PUT_SOUND_WORD_RIGHT (audio_channel[4].dat); - PUT_SOUND_WORD_LEFT (audio_channel[5].dat); - } else { - 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; + DO_CHANNEL_1 (data0, 0); + DO_CHANNEL_1 (data1, 1); + DO_CHANNEL_1 (data2, 2); + DO_CHANNEL_1 (data3, 3); - 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; - { - uae_u32 data = SBASEVAL16 (1) + data0; - FINISH_DATA (data, 16, 1); - put_sound_word_right (data); - } + 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 = SBASEVAL16(1) + data0; + FINISH_DATA (data, 16, 1); + put_sound_word_right (data); + } + + data1 += data2; + { + uae_u32 data = SBASEVAL16(1) + data1; + FINISH_DATA (data, 16, 1); + put_sound_word_left (data); + } - { - uae_u32 data = SBASEVAL16 (1) + data1; - FINISH_DATA (data, 16, 1); - put_sound_word_left (data); - } - } check_sound_buffers (); } -void sample8s_handler (void) +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; - data0 += data3; { - uae_u32 data = SBASEVAL8(1) + data0; - FINISH_DATA (data, 8, 1); - PUT_SOUND_BYTE_RIGHT (data); + 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; { - uae_u32 data = SBASEVAL8(1) + data1; - FINISH_DATA (data, 8, 1); - PUT_SOUND_BYTE_LEFT (data); + uae_u32 data = SBASEVAL16 (1) + data0; + FINISH_DATA (data, 16, 1); + put_sound_word_right (data); } + { + uae_u32 data = SBASEVAL16 (1) + data1; + FINISH_DATA (data, 16, 1); + put_sound_word_left (data); + } check_sound_buffers (); } -#else -void sample8s_handler (void) + +static void sample16si_rh_handler (void) { - sample8_handler(); + 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; + { + uae_u32 data = SBASEVAL16 (1) + data0; + FINISH_DATA (data, 16, 1); + put_sound_word_right (data); + } + + { + uae_u32 data = SBASEVAL16 (1) + data1; + FINISH_DATA (data, 16, 1); + put_sound_word_left (data); + } + check_sound_buffers (); } + +#else void sample16s_handler (void) { - sample16_handler(); + sample16_handler (); } -void sample16si_crux_handler (void) +static void sample16si_crux_handler (void) { - sample16i_crux_handler(); + sample16i_crux_handler (); } -void sample16si_rh_handler (void) +static void sample16si_rh_handler (void) { - sample16i_rh_handler(); + sample16i_rh_handler (); } #endif -static uae_u8 int2ulaw (int ch) +void switch_audio_interpol (void) { - int mask; +#if defined HAVE_8BIT_AUDIO_SUPPORT || defined HAVE_ULAW_AUDIO_SUPPORT + if (currprefs.sound_bits == 8) + /* only supported for 16-bit audio */ + return; +#endif - if (ch < 0) { - ch = -ch; - mask = 0x7f; - } - else { - mask = 0xff; - } - - if (ch < 32) { - ch = 0xF0 | ( 15 - (ch/2) ); - } else if (ch < 96) { - ch = 0xE0 | ( 15 - (ch-32)/4 ); - } else if (ch < 224) { - ch = 0xD0 | ( 15 - (ch-96)/8 ); - } else if (ch < 480) { - ch = 0xC0 | ( 15 - (ch-224)/16 ); - } else if (ch < 992 ) { - ch = 0xB0 | ( 15 - (ch-480)/32 ); - } else if (ch < 2016) { - ch = 0xA0 | ( 15 - (ch-992)/64 ); - } else if (ch < 4064) { - ch = 0x90 | ( 15 - (ch-2016)/128 ); - } else if (ch < 8160) { - ch = 0x80 | ( 15 - (ch-4064)/256 ); + if (currprefs.sound_interpol == 0) { + changed_prefs.sound_interpol = 1; + write_log ("Interpol on: rh\n"); + } else if (currprefs.sound_interpol == 1) { + changed_prefs.sound_interpol = 2; + write_log ("Interpol on: crux\n"); + } else if (currprefs.sound_interpol == 2) { + changed_prefs.sound_interpol = 3; + write_log ("Interpol on: sinc\n"); } else { - ch = 0x80; + changed_prefs.sound_interpol = 0; + write_log ("Interpol off\n"); } - return (uae_u8)(mask & ch); + return; } - -void sample_ulaw_handler (void) -{ - int nr; - uae_u32 data = 0; - - for (nr = 0; nr < 4; nr++) { - if (!(adkcon & (0x11 << nr))) { - uae_u32 d = audio_channel[nr].current_sample; - DO_CHANNEL_1 (d, nr); - data += d; - } - } - PUT_SOUND_BYTE (int2ulaw (data)); - check_sound_buffers (); -} - + void schedule_audio (void) { - unsigned long best = ~0ul; + unsigned long best = MAX_EV; int i; eventtab[ev_audio].active = 0; eventtab[ev_audio].oldcycles = get_cycles (); - for (i = 0; i < 6; i++) { + for (i = 0; i < 4; i++) { struct audio_channel_data *cdp = audio_channel + i; - if (cdp->state != 0) { + if (cdp->evtime != MAX_EV) { if (best > cdp->evtime) { best = cdp->evtime; eventtab[ev_audio].active = 1; } - } + } } eventtab[ev_audio].evtime = get_cycles () + best; } -static void audio_handler (int nr) +/* + * TODO: This function has been moved here from the audio back-end layer + * since it was common to all. + * Needs further cleaning up and a better name - or replacing entirely. + */ +void update_sound (unsigned int freq) +{ + if (obtainedfreq) { + if (0 /*is_vsync ()*/) { + if (currprefs.ntscmode) + scaled_sample_evtime = (unsigned long)(MAXHPOS_NTSC * MAXVPOS_NTSC * freq * CYCLE_UNIT + obtainedfreq - 1) / obtainedfreq; + else + scaled_sample_evtime = (unsigned long)(MAXHPOS_PAL * MAXVPOS_PAL * freq * CYCLE_UNIT + obtainedfreq - 1) / obtainedfreq; + } else { + scaled_sample_evtime = (unsigned long)(312.0 * 50 * CYCLE_UNIT / (obtainedfreq / 227.0)); + } + } +} + +static void audio_handler (unsigned int nr) { struct audio_channel_data *cdp = audio_channel + nr; + cdp->evtime = MAX_EV; switch (cdp->state) { case 0: - fprintf(stderr, "Bug in sound code\n"); + write_log ("Bug in sound code\n"); break; case 1: @@ -633,7 +648,7 @@ static void audio_handler (int nr) INTREQ(0x8000 | (0x80 << nr)); if (cdp->wlen != 1) cdp->wlen = (cdp->wlen - 1) & 0xFFFF; - cdp->nextdat = chipmem_bank.wget(cdp->pt); + cdp->nextdat = chipmem_wget (cdp->pt); cdp->pt += 2; break; @@ -698,6 +713,7 @@ static void audio_handler (int nr) if ((INTREQR() & (0x80 << nr)) && !cdp->dmaen) { cdp->state = 0; + cdp->evtime = MAX_EV; cdp->last_sample = 0; cdp->current_sample = 0; break; @@ -737,122 +753,66 @@ static void audio_handler (int nr) } } -static void ahi_handler (int nr) +static void audio_channel_enable_dma (struct audio_channel_data *cdp) { - struct audio_channel_data *cdp = audio_channel + nr; - - switch (cdp->state) { - case 0: - fprintf(stderr, "Bug in sound code\n"); - break; - - case 1: - /* We come here at the first hsync after DMA was turned on. */ - cdp->evtime = maxhpos * CYCLE_UNIT; - - cdp->state = 5; - INTREQ (0xA000); - if (cdp->wlen != 1) - cdp->wlen = (cdp->wlen - 1) & 0xFFFF; - cdp->nextdat = chipmem_bank.wget (cdp->pt); - - cdp->pt += 2; - break; - - case 5: - /* We come here at the second hsync after DMA was turned on. */ - if (currprefs.produce_sound == 0) - cdp->per = PERIOD_MAX; - - cdp->evtime = cdp->per; - cdp->dat = cdp->nextdat; - - cdp->state = 2; + if (cdp->evtime == MAX_EV) { + cdp->state = 1; + cdp->pt = cdp->lc; + cdp->wper = cdp->per; + cdp->wlen = cdp->len; cdp->data_written = 2; - break; - - case 2: - if (currprefs.produce_sound == 0) - cdp->per = PERIOD_MAX; - - cdp->evtime = cdp->per; - - if (cdp->intreq2 && cdp->dmaen) - INTREQ(0xA000); - cdp->intreq2 = 0; - - cdp->dat = cdp->nextdat; - - if (cdp->dmaen) - cdp->data_written = 2; - break; - - default: - cdp->state = 0; - break; + cdp->evtime = eventtab[ev_hsync].evtime - get_cycles (); } } -void aud0_handler (void) -{ - audio_handler (0); -} -void aud1_handler (void) -{ - audio_handler (1); -} -void aud2_handler (void) -{ - audio_handler (2); -} -void aud3_handler (void) +static void audio_channel_disable_dma (struct audio_channel_data *cdp) { - audio_handler (3); + if (cdp->state == 1 || cdp->state == 5) { + cdp->state = 0; + cdp->evtime = MAX_EV; + cdp->last_sample = 0; + cdp->current_sample = 0; + } } void audio_reset (void) { int i; + struct audio_channel_data *cdp; + if (savestate_state != STATE_RESTORE) { - memset (audio_channel, 0, 4 * sizeof *audio_channel); - audio_channel[0].per = PERIOD_MAX; - audio_channel[1].per = PERIOD_MAX; - audio_channel[2].per = PERIOD_MAX; - audio_channel[3].per = PERIOD_MAX; - audio_channel[0].voltbl = sound_table[0]; - audio_channel[1].voltbl = sound_table[0]; - audio_channel[2].voltbl = sound_table[0]; - audio_channel[3].voltbl = sound_table[0]; + for (i = 0; i < 4; i++) { + cdp = &audio_channel[i]; + memset (cdp, 0, sizeof *audio_channel); + cdp->per = PERIOD_MAX; + cdp->vol = 0; + cdp->evtime = MAX_EV; + } } else for (i = 0; i < 4; i++) audio_channel[i].dmaen = (dmacon & 0x200) && (dmacon & (1 << i)); - - memset (audio_channel + 4, 0, 2 * sizeof *audio_channel); - audio_channel[4].per = PERIOD_MAX; - audio_channel[5].per = PERIOD_MAX; - #ifndef MULTIPLICATION_PROFITABLE - for (i = 0; i < 6; i++) - audio_channel[nr].voltbl = sound_table[audio_channel[nr].vol]; + for (i = 0; i < 4; i++) + audio_channel[i].voltbl = sound_table[audio_channel[i].vol]; #endif - last_cycles = 0; - ahi_enabled = 0; - ahi_interrupt_state = 0; + last_cycles = get_cycles (); next_sample_evtime = scaled_sample_evtime; schedule_audio (); + events_schedule (); } STATIC_INLINE int sound_prefs_changed (void) { return (changed_prefs.produce_sound != currprefs.produce_sound - || changed_prefs.stereo != currprefs.stereo + || changed_prefs.sound_stereo != currprefs.sound_stereo || changed_prefs.mixed_stereo != currprefs.mixed_stereo || changed_prefs.sound_maxbsiz != currprefs.sound_maxbsiz || changed_prefs.sound_freq != currprefs.sound_freq - || changed_prefs.sound_bits != currprefs.sound_bits); + || changed_prefs.sound_bits != currprefs.sound_bits + || changed_prefs.sound_interpol != currprefs.sound_interpol); } void check_prefs_changed_audio (void) @@ -861,10 +821,11 @@ void check_prefs_changed_audio (void) close_sound (); currprefs.produce_sound = changed_prefs.produce_sound; - currprefs.stereo = changed_prefs.stereo; + currprefs.sound_stereo = changed_prefs.sound_stereo; currprefs.mixed_stereo = changed_prefs.mixed_stereo; currprefs.sound_bits = changed_prefs.sound_bits; currprefs.sound_freq = changed_prefs.sound_freq; + currprefs.sound_interpol = changed_prefs.sound_interpol; currprefs.sound_maxbsiz = changed_prefs.sound_maxbsiz; if (currprefs.produce_sound >= 2) { if (init_audio ()) { @@ -872,28 +833,36 @@ void check_prefs_changed_audio (void) next_sample_evtime = scaled_sample_evtime; } else if (! sound_available) { - fprintf (stderr, "Sound is not supported.\n"); + write_log ("Sound is not supported.\n"); } else { - fprintf (stderr, "Sorry, can't initialize sound.\n"); + write_log ("Sorry, can't initialize sound.\n"); currprefs.produce_sound = 0; /* So we don't do this every frame */ changed_prefs.produce_sound = 0; } } + compute_vsynctime (); } /* Select the right interpolation method. */ if (sample_handler == sample16_handler || sample_handler == sample16i_crux_handler - || sample_handler == sample16i_rh_handler) + || sample_handler == sample16i_rh_handler + || sample_handler == sample16i_sinc_handler) { sample_handler = (currprefs.sound_interpol == 0 ? sample16_handler : currprefs.sound_interpol == 1 ? sample16i_rh_handler - : sample16i_crux_handler); - else if (sample_handler == sample16s_handler + : currprefs.sound_interpol == 2 ? sample16i_crux_handler + : sample16i_sinc_handler); + } else if (sample_handler == sample16s_handler || sample_handler == sample16si_crux_handler - || sample_handler == sample16si_rh_handler) + || sample_handler == sample16si_rh_handler + || sample_handler == sample16si_sinc_handler) sample_handler = (currprefs.sound_interpol == 0 ? sample16s_handler : currprefs.sound_interpol == 1 ? sample16si_rh_handler - : sample16si_crux_handler); + : currprefs.sound_interpol == 2 ? sample16si_crux_handler + : sample16si_sinc_handler); + sample_prehandler = NULL; + if (sample_handler == sample16si_sinc_handler || sample_handler == sample16i_sinc_handler) + sample_prehandler = sinc_prehandler; if (currprefs.produce_sound == 0) { eventtab[ev_audio].active = 0; events_schedule (); @@ -909,54 +878,70 @@ void update_audio (void) n_cycles = get_cycles () - last_cycles; for (;;) { - int best = -1; unsigned long int best_evtime = n_cycles + 1; - if (audio_channel[0].state != 0 && best_evtime > audio_channel[0].evtime) - best = 0, best_evtime = audio_channel[0].evtime; - if (audio_channel[1].state != 0 && best_evtime > audio_channel[1].evtime) - best = 1, best_evtime = audio_channel[1].evtime; - if (audio_channel[2].state != 0 && best_evtime > audio_channel[2].evtime) - best = 2, best_evtime = audio_channel[2].evtime; - if (audio_channel[3].state != 0 && best_evtime > audio_channel[3].evtime) - best = 3, best_evtime = audio_channel[3].evtime; - if (audio_channel[4].state != 0 && best_evtime > audio_channel[4].evtime) - best = 4, best_evtime = audio_channel[4].evtime; - if (audio_channel[5].state != 0 && best_evtime > audio_channel[5].evtime) - best = 5, best_evtime = audio_channel[5].evtime; + if (audio_channel[0].evtime != MAX_EV && best_evtime > audio_channel[0].evtime) + best_evtime = audio_channel[0].evtime; + if (audio_channel[1].evtime != MAX_EV && best_evtime > audio_channel[1].evtime) + best_evtime = audio_channel[1].evtime; + if (audio_channel[2].evtime != MAX_EV && best_evtime > audio_channel[2].evtime) + best_evtime = audio_channel[2].evtime; + if (audio_channel[3].evtime != MAX_EV && best_evtime > audio_channel[3].evtime) + best_evtime = audio_channel[3].evtime; if (currprefs.produce_sound > 1 && best_evtime > next_sample_evtime) best_evtime = next_sample_evtime; if (best_evtime > n_cycles) break; - next_sample_evtime -= best_evtime; - audio_channel[0].evtime -= best_evtime; - audio_channel[1].evtime -= best_evtime; - audio_channel[2].evtime -= best_evtime; - audio_channel[3].evtime -= best_evtime; - audio_channel[4].evtime -= best_evtime; - audio_channel[5].evtime -= best_evtime; + if (audio_channel[0].evtime != MAX_EV) + audio_channel[0].evtime -= best_evtime; + if (audio_channel[1].evtime != MAX_EV) + audio_channel[1].evtime -= best_evtime; + if (audio_channel[2].evtime != MAX_EV) + audio_channel[2].evtime -= best_evtime; + if (audio_channel[3].evtime != MAX_EV) + audio_channel[3].evtime -= best_evtime; n_cycles -= best_evtime; - if (next_sample_evtime == 0 && currprefs.produce_sound > 1) { - next_sample_evtime = scaled_sample_evtime; - (*sample_handler) (); + if (currprefs.produce_sound > 1) { + next_sample_evtime -= best_evtime; + if (sample_prehandler) + sample_prehandler (best_evtime / CYCLE_UNIT); + if (next_sample_evtime == 0) { + next_sample_evtime = scaled_sample_evtime; + (*sample_handler) (); + } } - if (audio_channel[0].evtime == 0 && audio_channel[0].state != 0) + if (audio_channel[0].evtime == 0) audio_handler (0); - if (audio_channel[1].evtime == 0 && audio_channel[1].state != 0) + if (audio_channel[1].evtime == 0) audio_handler (1); - if (audio_channel[2].evtime == 0 && audio_channel[2].state != 0) + if (audio_channel[2].evtime == 0) audio_handler (2); - if (audio_channel[3].evtime == 0 && audio_channel[3].state != 0) + if (audio_channel[3].evtime == 0) audio_handler (3); - if (audio_channel[4].evtime == 0 && audio_channel[4].state != 0) - ahi_handler (4); - if (audio_channel[5].evtime == 0 && audio_channel[5].state != 0) - ahi_handler (5); } last_cycles = get_cycles () - n_cycles; } +void update_audio_dmacon (void) +{ + unsigned int i; + update_audio (); + + for (i = 0; i < 4; i++) { + struct audio_channel_data *cdp = audio_channel + i; + int chan_ena = (dmacon & 0x200) && (dmacon & (1<dmaen == chan_ena) + continue; + cdp->dmaen = chan_ena; + if (cdp->dmaen) + audio_channel_enable_dma (cdp); + else + audio_channel_disable_dma (cdp); + } + schedule_audio (); +} + void audio_evhandler (void) { if (currprefs.produce_sound == 0) @@ -966,7 +951,33 @@ void audio_evhandler (void) schedule_audio (); } -void AUDxDAT (int nr, uae_u16 v) +void audio_hsync (int dmaaction) +{ + int nr; + + update_audio (); + + /* Sound data is fetched at the beginning of each line */ + for (nr = 0; nr < 4; nr++) { + struct audio_channel_data *cdp = audio_channel + nr; + + if (cdp->data_written == 2) { + cdp->data_written = 0; + cdp->nextdat = chipmem_wget (cdp->pt); + cdp->pt += 2; + if (cdp->state == 2 || cdp->state == 3) { + if (cdp->wlen == 1) { + cdp->pt = cdp->lc; + cdp->wlen = cdp->len; + cdp->intreq2 = 1; + } else + cdp->wlen = (cdp->wlen - 1) & 0xFFFF; + } + } + } +} + +void AUDxDAT (unsigned int nr, uae_u16 v) { struct audio_channel_data *cdp = audio_channel + nr; @@ -986,21 +997,21 @@ void AUDxDAT (int nr, uae_u16 v) } } -void AUDxLCH (int nr, uae_u16 v) +void AUDxLCH (unsigned int nr, uae_u16 v) { update_audio (); audio_channel[nr].lc = (audio_channel[nr].lc & 0xffff) | ((uae_u32)v << 16); } -void AUDxLCL (int nr, uae_u16 v) +void AUDxLCL (unsigned int nr, uae_u16 v) { update_audio (); audio_channel[nr].lc = (audio_channel[nr].lc & ~0xffff) | (v & 0xFFFE); } -void AUDxPER (int nr, uae_u16 v) +void AUDxPER (unsigned int nr, uae_u16 v) { unsigned long per = v * CYCLE_UNIT; update_audio (); @@ -1010,27 +1021,32 @@ void AUDxPER (int nr, uae_u16 v) if (per < maxhpos * CYCLE_UNIT / 2 && currprefs.produce_sound < 3) per = maxhpos * CYCLE_UNIT / 2; + /* the sinc code registers paula output state changes, but has a finite + * buffer in which to do so. Hence, we forbid very low values; this should + * only limit the accurate rendering of supersonic sounds, which are + * filtered away on the sinc output path anyway. */ + if (currprefs.produce_sound == 3 && sample_handler == sample16si_sinc_handler && per < MIN_ALLOWED_PERIOD * CYCLE_UNIT) + per = MIN_ALLOWED_PERIOD * CYCLE_UNIT; - if (audio_channel[nr].per == PERIOD_MAX - && per != PERIOD_MAX) - { + if (audio_channel[nr].per == PERIOD_MAX && per != PERIOD_MAX + && audio_channel[nr].evtime != MAX_EV) { audio_channel[nr].evtime = CYCLE_UNIT; if (currprefs.produce_sound > 0) { schedule_audio (); events_schedule (); } } + audio_channel[nr].per = per; } -void AUDxLEN (int nr, uae_u16 v) +void AUDxLEN (unsigned int nr, uae_u16 v) { update_audio (); - audio_channel[nr].len = v; } -void AUDxVOL (int nr, uae_u16 v) +void AUDxVOL (unsigned int nr, uae_u16 v) { int v2 = v & 64 ? 63 : v & 63; @@ -1042,73 +1058,28 @@ void AUDxVOL (int nr, uae_u16 v) #endif } -int init_audio (void) +void update_adkmasks (void) { - int retval; - /* Some backward compatibility hacks until every port initializes - scaled_sample_evtime... */ - scaled_sample_evtime_ok = 0; - retval = init_sound (); - if (! scaled_sample_evtime_ok) - scaled_sample_evtime = sample_evtime * CYCLE_UNIT; - return retval; -} - -static uae_u32 ahi_entry (void) -{ - /* D0 contains the function code: - 0: get current output sample rate. - 1: get scaled divider for that frequency - 2: enable/disable AHI output (boolean value in D1) - 3: set right AHI channel info (LC in A0, period in D1, length D2, dmaen D3). - 4: set left AHI channel info (LC in A0, period in D1, length D2, dmaen D3). - 5: return interrupt state. */ - switch (m68k_dreg (regs, 0)) { - case 0: - return currprefs.sound_freq; - case 1: - return scaled_sample_evtime; - case 2: - ahi_enabled = m68k_dreg (regs, 1); - return 0; - case 3: - /* Must set a frequency lower than sample rate, i.e. using a higher - divider. */ - if (m68k_dreg (regs, 1) < scaled_sample_evtime) - return 0; - audio_channel[4].lc = m68k_areg (regs, 0); - audio_channel[4].per = m68k_dreg (regs, 1); - audio_channel[4].len = m68k_dreg (regs, 2); - audio_channel[4].dmaen = m68k_dreg (regs, 3); - return 1; - case 4: - if (m68k_dreg (regs, 1) < scaled_sample_evtime) - return 0; - audio_channel[5].lc = m68k_areg (regs, 0); - audio_channel[5].per = m68k_dreg (regs, 1); - audio_channel[5].len = m68k_dreg (regs, 2); - audio_channel[5].dmaen = m68k_dreg (regs, 3); - return 1; - case 5: - return ahi_interrupt_state; - default: - return 0; - } + unsigned long t; + + t = adkcon | (adkcon >> 4); + audio_channel[0].adk_mask = (((t >> 0) & 1) - 1); + audio_channel[1].adk_mask = (((t >> 1) & 1) - 1); + audio_channel[2].adk_mask = (((t >> 2) & 1) - 1); + audio_channel[3].adk_mask = (((t >> 3) & 1) - 1); } -void ahi_install (void) -{ - uaecptr a = here (); - org (0xF0FFB0); - calltrap (deftrap (ahi_entry)); - dw (RTS); - org (a); +int init_audio (void) +{ + int result = init_sound (); + update_sound (vblank_hz); + return result; } /* audio save/restore code FIXME: not working correctly */ /* help needed */ -uae_u8 *restore_audio (uae_u8 *src, int i) +const uae_u8 *restore_audio (int i, const uae_u8 *src) { struct audio_channel_data *acd; uae_u16 p; @@ -1131,8 +1102,7 @@ uae_u8 *restore_audio (uae_u8 *src, int return src; } - -uae_u8 *save_audio (int *len, int i) +uae_u8 *save_audio (int i, int *len) { struct audio_channel_data *acd; uae_u8 *dst = malloc (100);