#include #include "n_libaudio.h" #include "n_synth.h" #define RANGE 2.0 extern ALGlobals *alGlobals; #ifdef AUD_PROFILE extern u32 cnt_index, reverb_num, reverb_cnt, reverb_max, reverb_min, lastCnt[]; extern u32 load_num, load_cnt, load_max, load_min, save_num, save_cnt, save_max, save_min; #endif /* * macros */ #define SWAP(in, out) \ { \ s16 t = out; \ out = in; \ in = t; \ } #define N_REVERB_OUTCOUNT 0xb8 Acmd *_n_loadOutputBuffer(ALFx *r, ALDelay *d, s32 buff, Acmd *p); //_loadOutputBuffer Acmd *_n_loadBuffer(ALFx *r, s16 *curr_ptr, s32 buff, s32 count, Acmd *p); //_loadBuffer Acmd *__n_saveBuffer(ALFx *r, s16 *curr_ptr, s32 buff, Acmd *p); //__saveBuffer Acmd *__n_filterBuffer(ALLowPass *lp, s32 buff, Acmd *p); //__filterBuffer f32 _doModFunc(ALDelay *d, s32 count); // static s32 L_INC[] = { L0_INC, L1_INC, L2_INC }; /*********************************************************************** * Reverb filter public interfaces ***********************************************************************/ Acmd *n_alFxPull(void) { ALFx *r = (ALFx *)n_syn->auxBus->fx; ALFilter *source = r->filter.source; s16 i, buff1, buff2, input, output; s16 *in_ptr, *out_ptr, gain, *prev_out_ptr = 0; ALDelay *d, *pd; Acmd *ptr; #ifdef AUD_PROFILE lastCnt[++cnt_index] = osGetCount(); #endif // assert(source); ptr = n_alAuxBusPull(); /* * pull channels going into this effect first */ // ptr = (*source->handler)(source, outp, outCount, sampleOffset, p); input = N_AL_AUX_L_OUT; output = N_AL_AUX_R_OUT; buff1 = N_AL_TEMP_0; buff2 = N_AL_TEMP_1; // aSetBuffer(ptr++, 0, 0, 0, outCount<<1); /* set the buffer size */ aMix(ptr++, 0, 0xda83, N_AL_AUX_L_OUT, input); /* .707L = L - .293L */ aMix(ptr++, 0, 0x5a82, N_AL_AUX_R_OUT, input); /* mix the AuxL and AuxR into the AuxL */ /* and write the mixed value to the delay line at r->input */ ptr = __n_saveBuffer(r, r->input, input, ptr); aClearBuffer(ptr++, output, N_REVERB_OUTCOUNT << 1); /* clear the AL_AUX_R_OUT */ for (i = 0; i < r->section_count; i++) { d = &r->delay[i]; /* get the ALDelay structure */ in_ptr = &r->input[-d->input]; out_ptr = &r->input[-d->output]; if (in_ptr == prev_out_ptr) { SWAP(buff1, buff2); } else { /* load data at in_ptr into buff1 */ ptr = _n_loadBuffer(r, in_ptr, buff1, N_REVERB_OUTCOUNT, ptr); } ptr = _n_loadOutputBuffer(r, d, buff2, ptr); if (d->ffcoef) { aMix(ptr++, 0, (u16)d->ffcoef, buff1, buff2); if (!d->rs && !d->lp) { ptr = __n_saveBuffer(r, out_ptr, buff2, ptr); } } if (d->fbcoef) { aMix(ptr++, 0, (u16)d->fbcoef, buff2, buff1); ptr = __n_saveBuffer(r, in_ptr, buff1, ptr); } if (d->lp) ptr = __n_filterBuffer(d->lp, buff2, ptr); if (!d->rs) ptr = __n_saveBuffer(r, out_ptr, buff2, ptr); if (d->gain) aMix(ptr++, 0, (u16)d->gain, buff2, output); prev_out_ptr = &r->input[d->output]; } /* * bump the master delay line input pointer * modulo the length */ r->input += N_REVERB_OUTCOUNT; if (r->input > &r->base[r->length]) r->input -= r->length; /* * output already in AL_AUX_R_OUT * just copy to AL_AUX_L_OUT */ aDMEMMove(ptr++, output, N_AL_AUX_L_OUT, N_REVERB_OUTCOUNT<<1); #ifdef AUD_PROFILE PROFILE_AUD(reverb_num, reverb_cnt, reverb_max, reverb_min); #endif return ptr; } /* * This routine gets called by alSynSetFXParam. No checking takes place to * verify the validity of the paramID or the param value. input and output * values must be 8 byte aligned, so round down any param passed. */ s32 n_alFxParamHdl(void *filter, s32 paramID, void *param) { ALFx *f = (ALFx *) filter; s32 p = (paramID - 2) % 8; s32 s = (paramID - 2) / 8; s32 val = *(s32*)param; #define INPUT_PARAM 0 #define OUTPUT_PARAM 1 #define FBCOEF_PARAM 2 #define FFCOEF_PARAM 3 #define GAIN_PARAM 4 #define CHORUSRATE_PARAM 5 #define CHORUSDEPTH_PARAM 6 #define LPFILT_PARAM 7 switch(p) { case INPUT_PARAM: f->delay[s].input = (u32)val & 0xFFFFFFF8; break; case OUTPUT_PARAM: f->delay[s].output = (u32)val & 0xFFFFFFF8; break; case FFCOEF_PARAM: f->delay[s].ffcoef = (s16)val; break; case FBCOEF_PARAM: f->delay[s].fbcoef = (s16)val; break; case GAIN_PARAM: f->delay[s].gain = (s16)val; break; case CHORUSRATE_PARAM: /* f->delay[s].rsinc = ((f32)val)/0xffffff; */ f->delay[s].rsinc = ((((f32)val)/1000) * RANGE)/n_syn->outputRate; break; /* * the following constant is derived from: * * ratio = 2^(cents/1200) * * and therefore for hundredths of a cent * x * ln(ratio) = --------------- * (120,000)/ln(2) * where * 120,000/ln(2) = 173123.40... */ #define CONVERT 173123.404906676 #define LENGTH (f->delay[s].output - f->delay[s].input) case CHORUSDEPTH_PARAM: /*f->delay[s].rsgain = (((f32)val) / CONVERT) * LENGTH; */ f->delay[s].rsgain = (((f32)val) / CONVERT) * LENGTH; break; case LPFILT_PARAM: if(f->delay[s].lp) { f->delay[s].lp->fc = (s16)val; _init_lpfilter(f->delay[s].lp); } break; } return 0; } Acmd *_n_loadOutputBuffer(ALFx *r, ALDelay *d, s32 buff, Acmd *p) { Acmd *ptr = p; s32 ratio, count, rbuff = N_AL_TEMP_2; s16 *out_ptr; f32 fincount, fratio, delta; s32 ramalign = 0, length; s32 inCount = 0xb8; s8 tmp; static f32 val=0.0, lastval=-10.0; static f32 blob=0; /* * The following section implements the chorus resampling. Modulate where you pull * the samples from, since you need varying amounts of samples. */ if (d->rs) { length = d->output - d->input; delta = _doModFunc(d, inCount); /* get the number of samples to modulate by */ /* * find ratio of delta to delay length and quantize * to same resolution as resampler */ delta /= length; /* convert delta from number of samples to a pitch ratio */ delta = (s32)(delta * UNITY_PITCH); /* quantize to value microcode will use */ delta = delta / UNITY_PITCH; fratio = 1.0 - delta; /* pitch ratio needs to be centered around 1, not zero */ /* d->rs->delta is the difference between the fractional and integer value * of the samples needed. fratio * incount + rs->delta gives the number of samples * needed for this frame. */ fincount = d->rs->delta + (fratio * (f32)inCount); count = (s32) fincount; /* quantize to s32 */ d->rs->delta = fincount - (f32)count; /* calculate the round off and store */ /* * d->rsdelta is amount the out_ptr has deviated from its starting position. * You calc the out_ptr by taking d->output - d->rsdelta, and then using the * negative of that as an index into the delay buffer. loadBuffer that uses this * value then bumps it up if it is below the delay buffer. */ out_ptr = &r->input[-(d->output - d->rsdelta)]; ramalign = (((s32)out_ptr & 0x7) >> 1); /* calculate the number of samples needed to align the buffer*/ #ifdef _DEBUG #if 0 if(length > 0) { if (length - d->rsdelta > (s32)r->length) { __osError(ERR_ALMODDELAYOVERFLOW, 1, length - d->rsdelta - r->length); } } else if(length < 0) { if ((-length) - d->rsdelta > (s32)r->length) { __osError(ERR_ALMODDELAYOVERFLOW, 1, (-length) - d->rsdelta - r->length); } } #endif #endif /* load the rbuff with samples, note that there will be ramalign worth of * samples at the begining which you don't care about. */ ptr = _n_loadBuffer(r, out_ptr - ramalign, rbuff, count + ramalign, ptr); /* convert fratio to 16 bit fraction for microcode use */ ratio = (s32)(fratio * UNITY_PITCH); /* set the buffers, and do the resample */ // aSetBuffer(ptr++, 0, rbuff + (ramalign<<1), buff, inCount<<1); tmp = (buff>>8); n_aResample(ptr++, osVirtualToPhysical(d->rs->state), d->rs->first, ratio, rbuff + (ramalign<<1), tmp); d->rs->first = 0; /* turn off first time flag */ d->rsdelta += count - inCount; /* add the number of samples to d->rsdelta */ } else { out_ptr = &r->input[-d->output]; ptr = _n_loadBuffer(r, out_ptr, buff, inCount, ptr); } return ptr; } /* * This routine is for loading data from the delay line buff. If the * address of curr_ptr < r->base, it will force it to be within r->base * space, If the load goes past the end of r->base it will wrap around. * Cause count bytes of data at curr_ptr (within the delay line) to be * loaded into buff. (Buff is a dmem buffer) */ Acmd *_n_loadBuffer(ALFx *r, s16 *curr_ptr, s32 buff, s32 count, Acmd *p) { Acmd *ptr = p; s32 after_end, before_end; s16 *updated_ptr, *delay_end; #ifdef AUD_PROFILE lastCnt[++cnt_index] = osGetCount(); #endif delay_end = &r->base[r->length]; #ifdef _DEBUG if(curr_ptr > delay_end) __osError(ERR_ALMODDELAYOVERFLOW, 1, delay_end - curr_ptr); #endif if (curr_ptr < r->base) curr_ptr += r->length; updated_ptr = curr_ptr + count; if (updated_ptr > delay_end) { after_end = updated_ptr - delay_end; before_end = delay_end - curr_ptr; n_aLoadBuffer(ptr++, before_end << 1, buff, osVirtualToPhysical(curr_ptr)); n_aLoadBuffer(ptr++, after_end<<1, buff + (before_end<<1), osVirtualToPhysical(r->base)); } else { n_aLoadBuffer(ptr++, count<<1, buff, osVirtualToPhysical(curr_ptr)); } #ifdef AUD_PROFILE PROFILE_AUD(load_num, load_cnt, load_max, load_min); #endif return ptr; } /* * This routine is for writing data to the delay line buff. If the * address of curr_ptr < r->base, it will force it to be within r->base * space. If the write goes past the end of r->base, it will wrap around * Cause count bytes of data at buff to be written to delay line, curr_ptr. */ Acmd *__n_saveBuffer(ALFx *r, s16 *curr_ptr, s32 buff, Acmd *p) { Acmd *ptr = p; s32 after_end, before_end; s16 *updated_ptr, *delay_end; #ifdef AUD_PROFILE lastCnt[++cnt_index] = osGetCount(); #endif delay_end = &r->base[r->length]; if (curr_ptr < r->base) /* probably just security */ curr_ptr += r->length; /* shouldn't occur */ updated_ptr = curr_ptr + 0xb8; if (updated_ptr > delay_end) { /* if the data wraps past end of r->base */ after_end = updated_ptr - delay_end; before_end = delay_end - curr_ptr; n_aSaveBuffer(ptr++, before_end<<1, buff, osVirtualToPhysical(curr_ptr)); n_aSaveBuffer(ptr++, after_end<<1, buff+(before_end<<1), osVirtualToPhysical(r->base)); } else { n_aSaveBuffer(ptr++, 0xb8 << 1, buff, osVirtualToPhysical(curr_ptr)); } #ifdef AUD_PROFILE PROFILE_AUD(save_num, save_cnt, save_max, save_min); #endif return ptr; } Acmd *__n_filterBuffer(ALLowPass *lp, s32 buff, Acmd *p) { Acmd* ptr = p; s8 t8; n_aLoadADPCM(ptr++, 32, osVirtualToPhysical(lp->fcvec.fccoef)); t8 = buff>>8; n_aPoleFilter(ptr++, lp->first, lp->fgain, t8, osVirtualToPhysical(lp->fstate)); lp->first = 0; return ptr; }