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90%, core1/done/audio/n_reverb.c done

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Banjo Kazooie
2022-10-10 19:06:51 -05:00
parent acb28bee21
commit d163e882e9
8 changed files with 385 additions and 27 deletions
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2
README.md
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@@ -1,4 +1,4 @@
# banjo (89.9076%)
# banjo (90.0795%)
<img src="./progress/progress_total.svg">

2
include/file_and_line.h
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@@ -1,6 +1,6 @@
#ifndef _FILE_AND_LINE_H_
#define _FILE_AND_LINE_H_
#include <assert.h>
extern void func_8033F000(const char *, const char *, int);

6
progress/progress_core1.svg
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</mask>
<g mask="url(#anybadge_1)">
<path fill="#555" d="M0 0h44v20H0z"/>
<path fill="#6ae000" d="M44 0h67v20H44z"/>
<path fill="#65e000" d="M44 0h67v20H44z"/>
<path fill="url(#b)" d="M0 0h111v20H0z"/>
</g>
<g fill="#fff" text-anchor="middle" font-family="DejaVu Sans,Verdana,Geneva,sans-serif" font-size="11">
@@ -17,7 +17,7 @@
<text x="22.0" y="14">core1</text>
</g>
<g fill="#fff" text-anchor="middle" font-family="DejaVu Sans,Verdana,Geneva,sans-serif" font-size="11">
<text x="78.5" y="15" fill="#010101" fill-opacity=".3">76.3559%</text>
<text x="77.5" y="14">76.3559%</text>
<text x="78.5" y="15" fill="#010101" fill-opacity=".3">77.5401%</text>
<text x="77.5" y="14">77.5401%</text>
</g>
</svg>
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progress/progress_total.svg
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</mask>
<g mask="url(#anybadge_1)">
<path fill="#555" d="M0 0h150v20H0z"/>
<path fill="#2de000" d="M150 0h67v20H150z"/>
<path fill="#2ce000" d="M150 0h67v20H150z"/>
<path fill="url(#b)" d="M0 0h217v20H0z"/>
</g>
<g fill="#fff" text-anchor="middle" font-family="DejaVu Sans,Verdana,Geneva,sans-serif" font-size="11">
@@ -17,7 +17,7 @@
<text x="75.0" y="14">Banjo-Kazooie (us.v10)</text>
</g>
<g fill="#fff" text-anchor="middle" font-family="DejaVu Sans,Verdana,Geneva,sans-serif" font-size="11">
<text x="184.5" y="15" fill="#010101" fill-opacity=".3">89.9076%</text>
<text x="183.5" y="14">89.9076%</text>
<text x="184.5" y="15" fill="#010101" fill-opacity=".3">90.0795%</text>
<text x="183.5" y="14">90.0795%</text>
</g>
</svg>
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375
src/core1/done/audio/n_reverb.c Normal file
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#include <ultra64.h>
#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;
}

1
src/core1/done/audio/n_seqplayer.c
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@@ -865,7 +865,6 @@ ALSound *__n_lookupSoundQuick(N_ALSeqPlayer *seqp, u8 key, u8 vel, u8 chan)
s16 __n_vsVol(N_ALVoiceState *vs, N_ALSeqPlayer *seqp)
{
u32 t1, t2;
ALChanState *temp_a2;
t1 = (vs->tremelo*vs->velocity*vs->envGain*seqp->chanState[vs->channel].unkA) >> 13;
t2 = (vs->sound->sampleVolume*seqp->vol*seqp->chanState[vs->channel].vol) >> 14;

16
src/core1/n_reverb.c
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@@ -1,16 +0,0 @@
#include <ultra64.h>
#include "functions.h"
#include "variables.h"
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025F6D0.s")
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025F784.s")
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025F8B0.s")
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025F9F4.s")
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025FC10.s")
#pragma GLOBAL_ASM("asm/nonmatchings/core1/n_reverb/func_8025FE6C.s")

4
subyaml/core1.us.v10.yaml
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@@ -94,7 +94,7 @@ segments:
- [0x21AF0, c, done/audio/code_21AF0] #DONE
- [0x21B50, c, done/audio/code_21B50] #DONE
- [0x21C50, c, done/audio/n_synallocfx] #DONE
- [0x21CB0, c, n_reverb]
- [0x21CB0, c, done/audio/n_reverb]
- [0x22740, c, done/audio/n_auxbus] #DONE
- [0x227F0, c, done/audio/n_drvrNew] #DONE
- [0x22D50, c, done/audio/n_save] #DONE
@@ -285,7 +285,7 @@ segments:
- [0x3A950, .rodata, code_1D5D0]
- [0x3A9B0, .rodata, done/audio/n_synthesizer]
- [0x3A9C0, bin, data_3A9C0] # .rodata, done/audio/n_csplayer]
- [0x3AD30, bin, data_3AD30]
- [0x3AD30, .rodata, done/audio/n_reverb]
- [0x3AD60, .rodata, done/audio/n_drvrNew]
- [0x3AD80, bin, data_3AD80]
- [0x3ADD0, .rodata, done/audio/n_seqplayer]