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[iQue] Import libultra audio library files (#2433)
Note about comments in these files: The libultra audio library was originally available to developers in source code form, unlike most of libultra. These files are based on these source code forms, adapted to meet the style of this project. The comments in the files as of this commit are, apart from the style, comments from the original SGI authors.
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -1 +1,33 @@ | ||
| #include "libaudio.h" | ||
| #include "synthInternals.h" | ||
|
|
||
| Acmd* alAuxBusPull(void* filter, s16* outp, s32 outCount, s32 sampleOffset, Acmd* p) { | ||
| Acmd* ptr = p; | ||
| ALAuxBus* m = (ALAuxBus*)filter; | ||
| ALFilter** sources = m->sources; | ||
| s32 i; | ||
|
|
||
| // clear the output buffers here | ||
| aClearBuffer(ptr++, AL_AUX_L_OUT, outCount << 1); | ||
| aClearBuffer(ptr++, AL_AUX_R_OUT, outCount << 1); | ||
|
|
||
| for (i = 0; i < m->sourceCount; i++) { | ||
| ptr = sources[i]->handler(sources[i], outp, outCount, sampleOffset, ptr); | ||
| } | ||
| return ptr; | ||
| } | ||
|
|
||
| s32 alAuxBusParam(void* filter, s32 paramID, void* param) { | ||
| ALAuxBus* m = (ALAuxBus*)filter; | ||
| ALFilter** sources = m->sources; | ||
|
|
||
| switch (paramID) { | ||
| case AL_FILTER_ADD_SOURCE: | ||
| sources[m->sourceCount++] = (ALFilter*)param; | ||
| break; | ||
|
|
||
| default: | ||
| break; | ||
| } | ||
| return 0; | ||
| } |
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -1 +1,11 @@ | ||
| #include "libaudio.h" | ||
|
|
||
| void alCopy(void* src, void* dest, s32 len) { | ||
| s32 i; | ||
| u8* s = (u8*)src; | ||
| u8* d = (u8*)dest; | ||
|
|
||
| for (i = 0; i < len; i++) { | ||
| *d++ = *s++; | ||
| } | ||
| } |
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| Original file line number | Diff line number | Diff line change |
|---|---|---|
| @@ -1 +1,264 @@ | ||
| #include "libaudio.h" | ||
| #include "synthInternals.h" | ||
|
|
||
| // WARNING: THE FOLLOWING CONSTANT MUST BE KEPT IN SYNC WITH SCALING IN MICROCODE!!! | ||
| #define SCALE 16384 | ||
|
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||
| // the following arrays contain default parameters for a few hopefully useful effects. | ||
| #define ms *(((s32)(44.1f)) & ~7) | ||
|
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| // clang-format off | ||
| static s32 SMALLROOM_PARAMS[2 + 3 * 8] = { | ||
| /* sections */ 3, | ||
| /* length */ 100 ms, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 54 ms, 9830, -9830, 0, 0, 0, 0, | ||
| 19 ms, 38 ms, 3276, -3276, 0x3FFF, 0, 0, 0, | ||
| 0, 60 ms, 5000, 0, 0, 0, 0, 0x5000 | ||
| }; | ||
|
|
||
| static s32 BIGROOM_PARAMS[2 + 4 * 8] = { | ||
| /* sections */ 4, | ||
| /* length */ 100 ms, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 66 ms, 9830, -9830, 0, 0, 0, 0, | ||
| 22 ms, 54 ms, 3276, -3276, 0x3FFF, 0, 0, 0, | ||
| 66 ms, 91 ms, 3276, -3276, 0x3FFF, 0, 0, 0, | ||
| 0, 94 ms, 8000, 0, 0, 0, 0, 0x5000, | ||
| }; | ||
|
|
||
| static s32 ECHO_PARAMS[2 + 1 * 8] = { | ||
| /* sections */ 1, | ||
| /* length */ 200 ms, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 179 ms, 12000, 0, 0x7FFF, 0, 0, 0, | ||
| }; | ||
|
|
||
| static s32 CHORUS_PARAMS[2 + 1 * 8] = { | ||
| /* sections */ 1, | ||
| /* length */ 20 ms, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 5 ms, 0x4000, 0, 0x7FFF, 7600, 700, 0, | ||
| }; | ||
|
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||
| static s32 FLANGE_PARAMS[2 + 1 * 8] = { | ||
| /* sections */ 1, | ||
| /* length */ 20 ms, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 5 ms, 0, 0x5FFF, 0x7FFF, 380, 500, 0, | ||
| }; | ||
|
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||
| static s32 NULL_PARAMS[2 + 1 * 8] = { | ||
| /* sections */ 0, | ||
| /* length */ 0, | ||
| /* input output fbcoef ffcoef gain rate depth coef */ | ||
| 0, 0, 0, 0, 0, 0, 0, 0, | ||
| }; | ||
| // clang-format on | ||
|
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||
| void _init_lpfilter(ALLowPass* lp) { | ||
| s32 i; | ||
| s32 temp; | ||
| s16 fc; | ||
| f64 ffc; | ||
| f64 fcoef; | ||
|
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| temp = lp->fc * SCALE; | ||
| fc = temp >> 15; | ||
| lp->fgain = SCALE - fc; | ||
|
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| lp->first = true; | ||
| for (i = 0; i < 8; i++) { | ||
| lp->fcvec.fccoef[i] = 0; | ||
| } | ||
|
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| lp->fcvec.fccoef[i++] = fc; | ||
| fcoef = ffc = (f64)fc / SCALE; | ||
|
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| for (; i < 16; i++) { | ||
| fcoef *= ffc; | ||
| lp->fcvec.fccoef[i] = (s16)(s32)(fcoef * SCALE); | ||
| } | ||
| } | ||
|
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||
| void alFxNew(ALFx* r, ALSynConfig* c, ALHeap* hp) { | ||
| u16 i; | ||
| u16 j; | ||
| u16 k; | ||
| s32* param = NULL; | ||
| ALFilter* f = &r->filter; | ||
| ALDelay* d; | ||
|
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| alFilterNew(f, NULL, alFxParam, AL_FX); | ||
| f->handler = alFxPull; | ||
| r->paramHdl = (ALSetFXParam)alFxParamHdl; | ||
|
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||
| switch (c->fxType) { | ||
| case AL_FX_SMALLROOM: | ||
| param = SMALLROOM_PARAMS; | ||
| break; | ||
|
|
||
| case AL_FX_BIGROOM: | ||
| param = BIGROOM_PARAMS; | ||
| break; | ||
|
|
||
| case AL_FX_ECHO: | ||
| param = ECHO_PARAMS; | ||
| break; | ||
|
|
||
| case AL_FX_CHORUS: | ||
| param = CHORUS_PARAMS; | ||
| break; | ||
|
|
||
| case AL_FX_FLANGE: | ||
| param = FLANGE_PARAMS; | ||
| break; | ||
|
|
||
| case AL_FX_CUSTOM: | ||
| param = c->params; | ||
| break; | ||
|
|
||
| default: | ||
| param = NULL_PARAMS; | ||
| break; | ||
| } | ||
|
|
||
| j = 0; | ||
|
|
||
| r->section_count = param[j++]; | ||
| r->length = param[j++]; | ||
|
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||
| r->delay = alHeapAlloc(hp, r->section_count, sizeof(ALDelay)); | ||
| r->base = alHeapAlloc(hp, r->length, sizeof(s16)); | ||
| r->input = r->base; | ||
|
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||
| for (k = 0; k < r->length; k++) { | ||
| r->base[k] = 0; | ||
| } | ||
|
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||
| for (i = 0; i < r->section_count; i++) { | ||
| d = &r->delay[i]; | ||
| d->input = param[j++]; | ||
| d->output = param[j++]; | ||
| d->fbcoef = param[j++]; | ||
| d->ffcoef = param[j++]; | ||
| d->gain = param[j++]; | ||
|
|
||
| if (param[j] != 0) { | ||
| #define RANGE 2.0 | ||
| d->rsinc = ((((f32)param[j++]) / 1000.0f) * RANGE) / c->outputRate; | ||
|
|
||
| // 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 | ||
| d->rsgain = (((f32)param[j++]) / CONVERT) * (d->output - d->input); | ||
| d->rsval = 1.0f; | ||
| d->rsdelta = 0.0f; | ||
| d->rs = alHeapAlloc(hp, 1, sizeof(ALResampler)); | ||
| d->rs->state = alHeapAlloc(hp, 1, sizeof(RESAMPLE_STATE)); | ||
| d->rs->delta = 0.0f; | ||
| d->rs->first = true; | ||
| } else { | ||
| d->rs = NULL; | ||
| j++; | ||
| j++; | ||
| } | ||
|
|
||
| if (param[j] != 0) { | ||
| d->lp = alHeapAlloc(hp, 1, sizeof(ALLowPass)); | ||
| d->lp->fstate = alHeapAlloc(hp, 1, sizeof(POLEF_STATE)); | ||
| d->lp->fc = param[j++]; | ||
| _init_lpfilter(d->lp); | ||
| } else { | ||
| d->lp = NULL; | ||
| j++; | ||
| } | ||
| } | ||
| } | ||
|
|
||
| void alEnvmixerNew(ALEnvMixer* e, ALHeap* hp) { | ||
| alFilterNew(&e->filter, alEnvmixerPull, alEnvmixerParam, AL_ENVMIX); | ||
| e->state = alHeapAlloc(hp, 1, sizeof(ENVMIX_STATE)); | ||
| e->first = true; | ||
| e->motion = AL_STOPPED; | ||
| e->volume = 1; | ||
| e->ltgt = 1; | ||
| e->rtgt = 1; | ||
| e->cvolL = 1; | ||
| e->cvolR = 1; | ||
| e->dryamt = 0; | ||
| e->wetamt = 0; | ||
| e->lratm = 1; | ||
| e->lratl = 0; | ||
| e->lratm = 1; | ||
| e->lratl = 0; | ||
| e->delta = 0; | ||
| e->segEnd = 0; | ||
| e->pan = 0; | ||
| e->ctrlList = NULL; | ||
| e->ctrlTail = NULL; | ||
| e->sources = NULL; | ||
| } | ||
|
|
||
| void alLoadNew(ALLoadFilter* f, ALDMANew dmaNew, ALHeap* hp) { | ||
| s32 i; | ||
|
|
||
| // init filter superclass | ||
| alFilterNew(&f->filter, alAdpcmPull, alLoadParam, AL_ADPCM); | ||
|
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||
| f->state = alHeapAlloc(hp, 1, sizeof(ADPCM_STATE)); | ||
| f->lstate = alHeapAlloc(hp, 1, sizeof(ADPCM_STATE)); | ||
|
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| f->dma = dmaNew(&f->dmaState); | ||
|
|
||
| // init the adpcm state | ||
| f->lastsam = 0; | ||
| f->first = true; | ||
| f->memin = 0; | ||
| } | ||
|
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||
| void alResampleNew(ALResampler* r, ALHeap* hp) { | ||
| alFilterNew(&r->filter, alResamplePull, alResampleParam, AL_RESAMPLE); | ||
|
|
||
| // Init resampler state | ||
| r->state = alHeapAlloc(hp, 1, sizeof(RESAMPLE_STATE)); | ||
| r->delta = 0.0f; | ||
| r->first = true; | ||
| r->motion = AL_STOPPED; | ||
| r->ratio = 1.0f; | ||
| r->upitch = 0; | ||
| r->ctrlList = NULL; | ||
| r->ctrlTail = NULL; | ||
| // state in the ucode is initialized by the A_INIT flag | ||
| } | ||
|
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||
| void alAuxBusNew(ALAuxBus* m, void* sources, s32 maxSources) { | ||
| alFilterNew(&m->filter, alAuxBusPull, alAuxBusParam, AL_AUXBUS); | ||
| m->sourceCount = 0; | ||
| m->maxSources = maxSources; | ||
| m->sources = (ALFilter**)sources; | ||
| } | ||
|
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| void alMainBusNew(ALMainBus* m, void* sources, s32 maxSources) { | ||
| alFilterNew(&m->filter, alMainBusPull, alMainBusParam, AL_MAINBUS); | ||
| m->sourceCount = 0; | ||
| m->maxSources = maxSources; | ||
| m->sources = (ALFilter**)sources; | ||
| } | ||
|
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| void alSaveNew(ALSave* f) { | ||
| // init filter superclass | ||
| alFilterNew(&f->filter, alSavePull, alSaveParam, AL_SAVE); | ||
|
|
||
| // init the save state, which is a virtual dram address | ||
| f->dramout = 0; | ||
| f->first = true; | ||
| } |
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