project64/Source/Android/PluginRSP/alist_audio.cpp

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// Project64 - A Nintendo 64 emulator
// http://www.pj64-emu.com/
// Copyright(C) 2001-2021 Project64
// Copyright(C) 2014 Bobby Smiles
// Copyright(C) 2009 Richard Goedeken
// Copyright(C) 2002 Hacktarux
// GNU/GPLv2 licensed: https://gnu.org/licenses/gpl-2.0.html
#include "stdafx.h"
#include <string.h>
#include "alist.h"
#include "common.h"
#include "hle.h"
#include "mem.h"
#include "ucodes.h"
enum { DMEM_BASE = 0x5c0 };
// Helper functions
static uint32_t get_address(CHle * hle, uint32_t so)
{
return alist_get_address(hle, so, hle->alist_audio().segments, N_SEGMENTS);
}
static void set_address(CHle * hle, uint32_t so)
{
alist_set_address(hle, so, hle->alist_audio().segments, N_SEGMENTS);
}
static void clear_segments(CHle * hle)
{
memset(hle->alist_audio().segments, 0, N_SEGMENTS*sizeof(hle->alist_audio().segments[0]));
}
// Audio commands definition
static void SPNOOP(CHle * UNUSED(hle), uint32_t UNUSED(w1), uint32_t UNUSED(w2))
{
}
static void CLEARBUFF(CHle * hle, uint32_t w1, uint32_t w2)
{
uint16_t dmem = (w1 + DMEM_BASE) & 0xFFFF;
uint16_t count = w2;
if (count == 0)
return;
alist_clear(hle, dmem, align(count, 16));
}
static void ENVMIXER(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16) & 0xFF;
uint32_t address = get_address(hle, w2);
alist_envmix_exp(
hle,
flags & A_INIT,
flags & A_AUX,
hle->alist_audio().out, hle->alist_audio().dry_right,
hle->alist_audio().wet_left, hle->alist_audio().wet_right,
hle->alist_audio().in, hle->alist_audio().count,
hle->alist_audio().dry, hle->alist_audio().wet,
hle->alist_audio().vol,
hle->alist_audio().target,
hle->alist_audio().rate,
address);
}
static void ENVMIXER_GE(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16);
uint32_t address = get_address(hle, w2);
alist_envmix_ge(
hle,
flags & A_INIT,
flags & A_AUX,
hle->alist_audio().out, hle->alist_audio().dry_right,
hle->alist_audio().wet_left, hle->alist_audio().wet_right,
hle->alist_audio().in, hle->alist_audio().count,
hle->alist_audio().dry, hle->alist_audio().wet,
hle->alist_audio().vol,
hle->alist_audio().target,
hle->alist_audio().rate,
address);
}
static void RESAMPLE(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16) & 0xFF;
uint16_t pitch = w1 & 0xFFFF;
uint32_t address = get_address(hle, w2);
alist_resample(
hle,
flags & 0x1,
flags & 0x2,
hle->alist_audio().out,
hle->alist_audio().in,
align(hle->alist_audio().count, 16),
pitch << 1,
address);
}
static void SETVOL(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16) & 0xFF;
if (flags & A_AUX)
{
hle->alist_audio().dry = w1 & 0xFFFF;
hle->alist_audio().wet = w2 & 0xFFFF;
}
else
{
unsigned lr = (flags & A_LEFT) ? 0 : 1;
if (flags & A_VOL)
{
hle->alist_audio().vol[lr] = w1 & 0xFFFF;
}
else
{
hle->alist_audio().target[lr] = w1 & 0xFFFF;
hle->alist_audio().rate[lr] = w2;
}
}
}
static void SETLOOP(CHle * hle, uint32_t UNUSED(w1), uint32_t w2)
{
hle->alist_audio().loop = get_address(hle, w2);
}
static void ADPCM(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16) & 0xFF;
uint32_t address = get_address(hle, w2);
alist_adpcm(
hle,
flags & 0x1,
flags & 0x2,
false, // Unsupported in this microcode
hle->alist_audio().out,
hle->alist_audio().in,
align(hle->alist_audio().count, 32),
hle->alist_audio().table,
hle->alist_audio().loop,
address);
}
static void LOADBUFF(CHle * hle, uint32_t UNUSED(w1), uint32_t w2)
{
uint32_t address = get_address(hle, w2);
if (hle->alist_audio().count == 0)
{
return;
}
alist_load(hle, hle->alist_audio().in, address, hle->alist_audio().count);
}
static void SAVEBUFF(CHle * hle, uint32_t UNUSED(w1), uint32_t w2)
{
uint32_t address = get_address(hle, w2);
if (hle->alist_audio().count == 0)
{
return;
}
alist_save(hle, hle->alist_audio().out, address, hle->alist_audio().count);
}
static void SETBUFF(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16) & 0xFF;
if (flags & A_AUX)
{
hle->alist_audio().dry_right = (w1 + DMEM_BASE) & 0xFFFF;
hle->alist_audio().wet_left = (w2 >> 16) + DMEM_BASE;
hle->alist_audio().wet_right = (w2 + DMEM_BASE) & 0xFFFF;
}
else
{
hle->alist_audio().in = (w1 + DMEM_BASE) & 0xFFFF;
hle->alist_audio().out = ((w2 >> 16) + DMEM_BASE) & 0xFFFF;
hle->alist_audio().count = w2 & 0xFFFF;
}
}
static void DMEMMOVE(CHle * hle, uint32_t w1, uint32_t w2)
{
uint16_t dmemi = (w1 + DMEM_BASE) & 0xFFFF;
uint16_t dmemo = (w2 >> 16) + DMEM_BASE;
uint16_t count = (w2)& 0xFFFF;
if (count == 0)
return;
alist_move(hle, dmemo, dmemi, align(count, 16));
}
static void LOADADPCM(CHle * hle, uint32_t w1, uint32_t w2)
{
uint16_t count = (w1 & 0xFFFF);
uint32_t address = get_address(hle, w2);
dram_load_u16(hle, (uint16_t*)hle->alist_audio().table, address, align(count, 8) >> 1);
}
static void INTERLEAVE(CHle * hle, uint32_t UNUSED(w1), uint32_t w2)
{
uint16_t left = (w2 >> 16) + DMEM_BASE;
uint16_t right = (w2 + DMEM_BASE) & 0xFFFF;
if (hle->alist_audio().count == 0)
return;
alist_interleave(hle, hle->alist_audio().out, left, right, align(hle->alist_audio().count, 16));
}
static void MIXER(CHle * hle, uint32_t w1, uint32_t w2)
{
int16_t gain = (w1)& 0xFFFF;
uint16_t dmemi = ((w2 >> 16) + DMEM_BASE) & 0xFFFF;
uint16_t dmemo = (w2 + DMEM_BASE) & 0xFFFF;
if (hle->alist_audio().count == 0)
return;
alist_mix(hle, dmemo, dmemi, align(hle->alist_audio().count, 32), gain);
}
static void SEGMENT(CHle * hle, uint32_t UNUSED(w1), uint32_t w2)
{
set_address(hle, w2);
}
static void POLEF(CHle * hle, uint32_t w1, uint32_t w2)
{
uint8_t flags = (w1 >> 16);
uint16_t gain = w1;
uint32_t address = get_address(hle, w2);
if (hle->alist_audio().count == 0)
return;
alist_polef(
hle,
flags & A_INIT,
hle->alist_audio().out,
hle->alist_audio().in,
align(hle->alist_audio().count, 16),
gain,
hle->alist_audio().table,
address);
}
// Global functions
void alist_process_audio(CHle * hle)
{
static const acmd_callback_t ABI[0x10] = {
SPNOOP, ADPCM, CLEARBUFF, ENVMIXER,
LOADBUFF, RESAMPLE, SAVEBUFF, SEGMENT,
SETBUFF, SETVOL, DMEMMOVE, LOADADPCM,
MIXER, INTERLEAVE, POLEF, SETLOOP
};
clear_segments(hle);
alist_process(hle, ABI, 0x10);
}
void alist_process_audio_ge(CHle * hle)
{
static const acmd_callback_t ABI[0x10] =
{
SPNOOP, ADPCM, CLEARBUFF, ENVMIXER_GE,
LOADBUFF, RESAMPLE, SAVEBUFF, SEGMENT,
SETBUFF, SETVOL, DMEMMOVE, LOADADPCM,
MIXER, INTERLEAVE, POLEF, SETLOOP
};
clear_segments(hle);
alist_process(hle, ABI, 0x10);
}
void alist_process_audio_bc(CHle * hle)
{
static const acmd_callback_t ABI[0x10] =
{
SPNOOP, ADPCM, CLEARBUFF, ENVMIXER_GE,
LOADBUFF, RESAMPLE, SAVEBUFF, SEGMENT,
SETBUFF, SETVOL, DMEMMOVE, LOADADPCM,
MIXER, INTERLEAVE, POLEF, SETLOOP
};
clear_segments(hle);
alist_process(hle, ABI, 0x10);
}