mirror of
https://github.com/hrydgard/ppsspp.git
synced 2024-12-13 16:45:46 +00:00
c8e719e165
(Some compilers want it declared inline in the header too)
869 lines
21 KiB
C++
869 lines
21 KiB
C++
// Copyright (c) 2012- PPSSPP Project.
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// This program is free software: you can redistribute it and/or modify
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// it under the terms of the GNU General Public License as published by
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// the Free Software Foundation, version 2.0 or later versions.
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// This program is distributed in the hope that it will be useful,
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// but WITHOUT ANY WARRANTY; without even the implied warranty of
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// MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
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// GNU General Public License 2.0 for more details.
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// A copy of the GPL 2.0 should have been included with the program.
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// If not, see http://www.gnu.org/licenses/
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// Official git repository and contact information can be found at
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// https://github.com/hrydgard/ppsspp and http://www.ppsspp.org/.
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#include "base/basictypes.h"
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#include "Globals.h"
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#include "Core/MemMap.h"
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#include "Core/HLE/sceAtrac.h"
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#include "Core/Config.h"
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#include "Core/Reporting.h"
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#include "SasAudio.h"
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#include <algorithm>
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// #define AUDIO_TO_FILE
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static const u8 f[16][2] = {
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{ 0, 0 },
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{ 60, 0 },
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{ 115, 52 },
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{ 98, 55 },
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{ 122, 60 },
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// TODO: The below values could use more testing, but match initial tests.
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// Not sure if they are used by games, found by tests.
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{ 0, 0 },
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{ 0, 0 },
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{ 52, 0 },
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{ 55, 2 },
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{ 60, 125 },
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{ 0, 0 },
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{ 0, 91 },
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{ 0, 0 },
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{ 2, 216 },
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{ 125, 6 },
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{ 0, 151 },
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};
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void VagDecoder::Start(u32 data, u32 vagSize, bool loopEnabled) {
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loopEnabled_ = loopEnabled;
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loopAtNextBlock_ = false;
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loopStartBlock_ = -1;
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numBlocks_ = vagSize / 16;
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end_ = false;
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data_ = data;
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read_ = data;
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curSample = 28;
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curBlock_ = -1;
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s_1 = 0; // per block?
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s_2 = 0;
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}
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void VagDecoder::DecodeBlock(u8 *&read_pointer) {
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u8 *readp = read_pointer;
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int predict_nr = *readp++;
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int shift_factor = predict_nr & 0xf;
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predict_nr >>= 4;
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int flags = *readp++;
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if (flags == 7) {
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VERBOSE_LOG(SASMIX, "VAG ending block at %d", curBlock_);
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end_ = true;
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return;
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}
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else if (flags == 6) {
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loopStartBlock_ = curBlock_;
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}
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else if (flags == 3) {
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if (loopEnabled_) {
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loopAtNextBlock_ = true;
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}
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}
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// Keep state in locals to avoid bouncing to memory.
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int s1 = s_1;
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int s2 = s_2;
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int coef1 = f[predict_nr][0];
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int coef2 = -f[predict_nr][1];
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for (int i = 0; i < 28; i += 2) {
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u8 d = *readp++;
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int sample1 = (short)((d & 0xf) << 12) >> shift_factor;
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int sample2 = (short)((d & 0xf0) << 8) >> shift_factor;
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s2 = clamp_s16(sample1 + ((s1 * coef1 + s2 * coef2) >> 6));
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s1 = clamp_s16(sample2 + ((s2 * coef1 + s1 * coef2) >> 6));
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samples[i] = s2;
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samples[i + 1] = s1;
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}
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s_1 = s1;
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s_2 = s2;
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curSample = 0;
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curBlock_++;
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if (curBlock_ == numBlocks_) {
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end_ = true;
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}
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read_pointer = readp;
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}
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void VagDecoder::GetSamples(s16 *outSamples, int numSamples) {
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if (end_) {
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memset(outSamples, 0, numSamples * sizeof(s16));
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return;
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}
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u8 *readp = Memory::GetPointer(read_);
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if (!readp) {
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WARN_LOG(SASMIX, "Bad VAG samples address?");
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return;
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}
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u8 *origp = readp;
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for (int i = 0; i < numSamples; i++) {
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if (curSample == 28) {
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if (loopAtNextBlock_) {
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VERBOSE_LOG(SASMIX, "Looping VAG from block %d/%d to %d", curBlock_, numBlocks_, loopStartBlock_);
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// data_ starts at curBlock = -1.
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read_ = data_ + 16 * loopStartBlock_ + 16;
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readp = Memory::GetPointerUnchecked(read_);
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origp = readp;
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curBlock_ = loopStartBlock_;
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loopAtNextBlock_ = false;
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}
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DecodeBlock(readp);
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if (end_) {
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// Clear the rest of the buffer and return.
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memset(&outSamples[i], 0, (numSamples - i) * sizeof(s16));
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return;
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}
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}
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outSamples[i] = samples[curSample++];
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}
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if (readp > origp) {
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read_ += readp - origp;
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}
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}
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void VagDecoder::DoState(PointerWrap &p) {
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auto s = p.Section("VagDecoder", 1);
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if (!s)
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return;
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p.DoArray(samples, ARRAY_SIZE(samples));
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p.Do(curSample);
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p.Do(data_);
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p.Do(read_);
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p.Do(curBlock_);
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p.Do(loopStartBlock_);
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p.Do(numBlocks_);
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p.Do(s_1);
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p.Do(s_2);
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p.Do(loopEnabled_);
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p.Do(loopAtNextBlock_);
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p.Do(end_);
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}
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int SasAtrac3::setContext(u32 context) {
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contextAddr = context;
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atracID = _AtracGetIDByContext(context);
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if (!sampleQueue)
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sampleQueue = new BufferQueue();
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sampleQueue->clear();
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return 0;
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}
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int SasAtrac3::getNextSamples(s16* outbuf, int wantedSamples) {
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if (atracID < 0)
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return -1;
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u32 finish = 0;
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int wantedbytes = wantedSamples * sizeof(s16);
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while (!finish && sampleQueue->getQueueSize() < wantedbytes) {
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u32 numSamples = 0;
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int remains = 0;
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static s16 buf[0x800];
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_AtracDecodeData(atracID, (u8*)buf, &numSamples, &finish, &remains);
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if (numSamples > 0)
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sampleQueue->push((u8*)buf, numSamples * sizeof(s16));
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else
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finish = 1;
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}
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sampleQueue->pop_front((u8*)outbuf, wantedbytes);
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return finish;
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}
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int SasAtrac3::addStreamData(u8* buf, u32 addbytes) {
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if (atracID > 0) {
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_AtracAddStreamData(atracID, buf, addbytes);
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}
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return 0;
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}
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void SasAtrac3::DoState(PointerWrap &p) {
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auto s = p.Section("SasAtrac3", 1);
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if (!s)
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return;
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p.Do(contextAddr);
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p.Do(atracID);
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if (p.mode == p.MODE_READ && atracID >= 0 && !sampleQueue) {
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sampleQueue = new BufferQueue();
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}
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}
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// http://code.google.com/p/jpcsp/source/browse/trunk/src/jpcsp/HLE/modules150/sceSasCore.java
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static int simpleRate(int n) {
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n &= 0x7F;
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if (n == 0x7F) {
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return 0;
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}
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int rate = ((7 - (n & 0x3)) << 26) >> (n >> 2);
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if (rate == 0) {
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return 1;
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}
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return rate;
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}
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static int exponentRate(int n) {
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n &= 0x7F;
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if (n == 0x7F) {
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return 0;
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}
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int rate = ((7 - (n & 0x3)) << 24) >> (n >> 2);
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if (rate == 0) {
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return 1;
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}
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return rate;
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}
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static int getAttackRate(int bitfield1) {
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return simpleRate(bitfield1 >> 8);
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}
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static int getAttackType(int bitfield1) {
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return (bitfield1 & 0x8000) == 0 ? PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE : PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT;
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}
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static int getDecayRate(int bitfield1) {
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int n = (bitfield1 >> 4) & 0x000F;
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if (n == 0)
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return 0x7FFFFFFF;
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return 0x80000000 >> n;
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}
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static int getSustainType(int bitfield2) {
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return (bitfield2 >> 14) & 3;
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}
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static int getSustainRate(int bitfield2) {
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if (getSustainType(bitfield2) == PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE) {
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return exponentRate(bitfield2 >> 6);
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} else {
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return simpleRate(bitfield2 >> 6);
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}
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}
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static int getReleaseType(int bitfield2) {
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return (bitfield2 & 0x0020) == 0 ? PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE : PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE;
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}
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static int getReleaseRate(int bitfield2) {
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int n = bitfield2 & 0x001F;
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if (n == 31) {
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return 0;
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}
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if (getReleaseType(bitfield2) == PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE) {
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if (n == 30) {
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return 0x40000000;
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} else if (n == 29) {
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return 1;
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}
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return 0x10000000 >> n;
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}
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if (n == 0)
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return 0x7FFFFFFF;
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return 0x80000000 >> n;
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}
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static int getSustainLevel(int bitfield1) {
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return ((bitfield1 & 0x000F) + 1) << 26;
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}
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void ADSREnvelope::SetSimpleEnvelope(u32 ADSREnv1, u32 ADSREnv2) {
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attackRate = getAttackRate(ADSREnv1);
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attackType = getAttackType(ADSREnv1);
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decayRate = getDecayRate(ADSREnv1);
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decayType = PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE;
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sustainRate = getSustainRate(ADSREnv2);
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sustainType = getSustainType(ADSREnv2);
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releaseRate = getReleaseRate(ADSREnv2);
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releaseType = getReleaseType(ADSREnv2);
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sustainLevel = getSustainLevel(ADSREnv1);
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if (attackRate < 0 || decayRate < 0 || sustainRate < 0 || releaseRate < 0) {
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ERROR_LOG_REPORT(SCESAS, "Simple ADSR resulted in invalid rates: %04x, %04x", ADSREnv1, ADSREnv2);
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}
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}
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SasInstance::SasInstance()
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: maxVoices(PSP_SAS_VOICES_MAX),
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sampleRate(44100),
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outputMode(PSP_SAS_OUTPUTMODE_MIXED),
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mixBuffer(0),
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sendBuffer(0),
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resampleBuffer(0),
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grainSize(0) {
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#ifdef AUDIO_TO_FILE
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audioDump = fopen("D:\\audio.raw", "wb");
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#endif
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memset(&waveformEffect, 0, sizeof(waveformEffect));
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waveformEffect.type = PSP_SAS_EFFECT_TYPE_OFF;
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waveformEffect.isDryOn = 1;
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}
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SasInstance::~SasInstance() {
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ClearGrainSize();
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}
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void SasInstance::ClearGrainSize() {
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if (mixBuffer)
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delete [] mixBuffer;
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if (sendBuffer)
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delete [] sendBuffer;
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if (resampleBuffer)
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delete [] resampleBuffer;
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mixBuffer = NULL;
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sendBuffer = NULL;
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resampleBuffer = NULL;
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}
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void SasInstance::SetGrainSize(int newGrainSize) {
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grainSize = newGrainSize;
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// If you change the sizes here, don't forget DoState().
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if (mixBuffer)
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delete [] mixBuffer;
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if (sendBuffer)
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delete [] sendBuffer;
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mixBuffer = new s32[grainSize * 2];
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sendBuffer = new s32[grainSize * 2];
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memset(mixBuffer, 0, sizeof(int) * grainSize * 2);
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memset(sendBuffer, 0, sizeof(int) * grainSize * 2);
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if (resampleBuffer)
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delete [] resampleBuffer;
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// 2 samples padding at the start, that's where we copy the two last samples from the channel
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// so that we can do bicubic resampling if necessary. Plus 1 for smoothness hackery.
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resampleBuffer = new s16[grainSize * 4 + 3];
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}
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void SasVoice::ReadSamples(s16 *output, int numSamples) {
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// Read N samples into the resample buffer. Could do either PCM or VAG here.
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switch (type) {
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case VOICETYPE_VAG:
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vag.GetSamples(output, numSamples);
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break;
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case VOICETYPE_PCM:
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{
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int needed = numSamples;
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s16 *out = output;
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while (needed > 0) {
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u32 size = std::min(pcmSize - pcmIndex, needed);
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if (!on) {
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pcmIndex = 0;
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break;
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}
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Memory::Memcpy(out, pcmAddr + pcmIndex * sizeof(s16), size * sizeof(s16));
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pcmIndex += size;
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needed -= size;
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out += size;
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if (pcmIndex >= pcmSize) {
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if (!loop) {
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// All out, quit. We'll end in HaveSamplesEnded().
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break;
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}
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pcmIndex = pcmLoopPos;
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}
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}
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if (needed > 0) {
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memset(out, 0, needed * sizeof(s16));
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}
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}
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break;
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case VOICETYPE_ATRAC3:
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{
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int ret = atrac3.getNextSamples(output, numSamples);
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if (ret) {
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// Hit atrac3 voice end
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playing = false;
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on = false; // ??
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envelope.End();
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}
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}
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break;
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default:
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{
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memset(output, 0, numSamples * sizeof(s16));
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}
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break;
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}
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}
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bool SasVoice::HaveSamplesEnded() {
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switch (type) {
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case VOICETYPE_VAG:
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return vag.End();
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case VOICETYPE_PCM:
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return pcmIndex >= pcmSize;
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case VOICETYPE_ATRAC3:
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// TODO: Is it here, or before the samples are processed?
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return false;
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default:
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return false;
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}
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}
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void SasInstance::MixVoice(SasVoice &voice) {
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switch (voice.type) {
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case VOICETYPE_VAG:
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if (voice.type == VOICETYPE_VAG && !voice.vagAddr)
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break;
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// else fallthrough! Don't change the check above.
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case VOICETYPE_PCM:
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if (voice.type == VOICETYPE_PCM && !voice.pcmAddr)
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break;
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// else fallthrough! Don't change the check above.
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default:
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// Load resample history (so we can use a wide filter)
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resampleBuffer[0] = voice.resampleHist[0];
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resampleBuffer[1] = voice.resampleHist[1];
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// Figure out number of samples to read.
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// Actually this is not entirely correct - we need to get one extra sample, and store it
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// for the next time around. A little complicated...
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// But for now, see Smoothness HACKERY below :P
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u32 numSamples = ((u32)voice.sampleFrac + (u32)grainSize * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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if ((int)numSamples > grainSize * 4) {
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ERROR_LOG(SASMIX, "numSamples too large, clamping: %i vs %i", numSamples, grainSize * 4);
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numSamples = grainSize * 4;
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}
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// This feels a bit hacky. The first 32 samples after a keyon are 0s.
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const bool ignorePitch = voice.type == VOICETYPE_PCM && voice.pitch > PSP_SAS_PITCH_BASE;
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if (voice.envelope.NeedsKeyOn()) {
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int delay = ignorePitch ? 32 : (32 * (u32)voice.pitch) >> PSP_SAS_PITCH_BASE_SHIFT;
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// VAG seems to have an extra sample delay (not shared by PCM.)
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if (voice.type == VOICETYPE_VAG)
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++delay;
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voice.ReadSamples(resampleBuffer + 2 + delay, numSamples - delay);
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} else {
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voice.ReadSamples(resampleBuffer + 2, numSamples);
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}
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// Smoothness HACKERY
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resampleBuffer[2 + numSamples] = resampleBuffer[2 + numSamples - 1];
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// Save resample history
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voice.resampleHist[0] = resampleBuffer[2 + numSamples - 2];
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voice.resampleHist[1] = resampleBuffer[2 + numSamples - 1];
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// Resample to the correct pitch, writing exactly "grainSize" samples.
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// This is a HORRIBLE resampler by the way.
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// TODO: Special case no-resample case (and 2x and 0.5x) for speed, it's not uncommon
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u32 sampleFrac = voice.sampleFrac;
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// We need to shift by 12 anyway, so combine that with the volume shift.
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int volumeShift = (12 + MAX_CONFIG_VOLUME - g_Config.iSFXVolume);
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if (volumeShift < 0) volumeShift = 0;
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for (int i = 0; i < grainSize; i++) {
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// For now: nearest neighbour, not even using the resample history at all.
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int sample = resampleBuffer[sampleFrac / PSP_SAS_PITCH_BASE + 2];
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sampleFrac += voice.pitch;
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// The maximum envelope height (PSP_SAS_ENVELOPE_HEIGHT_MAX) is (1 << 30) - 1.
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// Reduce it to 14 bits, by shifting off 15. Round up by adding (1 << 14) first.
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int envelopeValue = voice.envelope.GetHeight();
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envelopeValue = (envelopeValue + (1 << 14)) >> 15;
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// We just scale by the envelope before we scale by volumes.
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// Again, we round up by adding (1 << 14) first (*after* multiplying.)
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sample = ((sample * envelopeValue) + (1 << 14)) >> 15;
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|
|
// We mix into this 32-bit temp buffer and clip in a second loop
|
|
// Ideally, the shift right should be there too but for now I'm concerned about
|
|
// not overflowing.
|
|
mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> volumeShift; // Max = 16 and Min = 12(default)
|
|
mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> volumeShift; // Max = 16 and Min = 12(default)
|
|
sendBuffer[i * 2] += sample * voice.effectLeft >> volumeShift;
|
|
sendBuffer[i * 2 + 1] += sample * voice.effectRight >> volumeShift;
|
|
voice.envelope.Step();
|
|
}
|
|
|
|
voice.sampleFrac = sampleFrac;
|
|
// Let's hope grainSize is a power of 2.
|
|
//voice.sampleFrac &= grainSize * PSP_SAS_PITCH_BASE - 1;
|
|
voice.sampleFrac -= numSamples * PSP_SAS_PITCH_BASE;
|
|
|
|
if (voice.HaveSamplesEnded())
|
|
voice.envelope.End();
|
|
if (voice.envelope.HasEnded())
|
|
{
|
|
// NOTICE_LOG(SCESAS, "Hit end of envelope");
|
|
voice.playing = false;
|
|
voice.on = false;
|
|
}
|
|
}
|
|
}
|
|
|
|
void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
|
|
int voicesPlayingCount = 0;
|
|
|
|
for (int v = 0; v < PSP_SAS_VOICES_MAX; v++) {
|
|
SasVoice &voice = voices[v];
|
|
if (!voice.playing || voice.paused)
|
|
continue;
|
|
voicesPlayingCount++;
|
|
MixVoice(voice);
|
|
}
|
|
|
|
// Okay, apply effects processing to the Send buffer.
|
|
// TODO: Is this only done in PSP_SAS_OUTPUTMODE_MIXED?
|
|
//if (waveformEffect.type != PSP_SAS_EFFECT_TYPE_OFF)
|
|
// ApplyReverb();
|
|
|
|
// Then mix the send buffer in with the rest.
|
|
|
|
// Alright, all voices mixed. Let's convert and clip, and at the same time, wipe mixBuffer for next time. Could also dither.
|
|
s16 *outp = (s16 *)Memory::GetPointer(outAddr);
|
|
const s16 *inp = inAddr ? (s16*)Memory::GetPointer(inAddr) : 0;
|
|
if (outputMode == PSP_SAS_OUTPUTMODE_MIXED) {
|
|
// TODO: Mix send when it has proper values, probably based on dry/wet?
|
|
if (inp) {
|
|
for (int i = 0; i < grainSize * 2; i += 2) {
|
|
int sampleL = mixBuffer[i + 0] + ((*inp++) * leftVol >> 12);
|
|
int sampleR = mixBuffer[i + 1] + ((*inp++) * rightVol >> 12);
|
|
*outp++ = clamp_s16(sampleL);
|
|
*outp++ = clamp_s16(sampleR);
|
|
}
|
|
} else {
|
|
for (int i = 0; i < grainSize * 2; i += 2) {
|
|
*outp++ = clamp_s16(mixBuffer[i + 0]);
|
|
*outp++ = clamp_s16(mixBuffer[i + 1]);
|
|
}
|
|
}
|
|
} else {
|
|
s16 *outpL = outp + grainSize * 0;
|
|
s16 *outpR = outp + grainSize * 1;
|
|
s16 *outpSendL = outp + grainSize * 2;
|
|
s16 *outpSendR = outp + grainSize * 3;
|
|
WARN_LOG_REPORT_ONCE(sasraw, SCESAS, "sceSasCore: raw outputMode");
|
|
for (int i = 0; i < grainSize * 2; i += 2) {
|
|
*outpL++ = clamp_s16(mixBuffer[i + 0]);
|
|
*outpR++ = clamp_s16(mixBuffer[i + 1]);
|
|
*outpSendL++ = clamp_s16(sendBuffer[i + 0]);
|
|
*outpSendR++ = clamp_s16(sendBuffer[i + 1]);
|
|
}
|
|
}
|
|
memset(mixBuffer, 0, grainSize * sizeof(int) * 2);
|
|
memset(sendBuffer, 0, grainSize * sizeof(int) * 2);
|
|
|
|
#ifdef AUDIO_TO_FILE
|
|
fwrite(Memory::GetPointer(outAddr), 1, grainSize * 2 * 2, audioDump);
|
|
#endif
|
|
}
|
|
|
|
void SasInstance::ApplyReverb() {
|
|
// for (int i = 0; i < grainSize * 2; i += 2) {
|
|
// modify sendBuffer
|
|
// }
|
|
}
|
|
|
|
void SasInstance::DoState(PointerWrap &p) {
|
|
auto s = p.Section("SasInstance", 1);
|
|
if (!s)
|
|
return;
|
|
|
|
p.Do(grainSize);
|
|
if (p.mode == p.MODE_READ) {
|
|
if (grainSize > 0) {
|
|
SetGrainSize(grainSize);
|
|
} else {
|
|
ClearGrainSize();
|
|
}
|
|
}
|
|
|
|
p.Do(maxVoices);
|
|
p.Do(sampleRate);
|
|
p.Do(outputMode);
|
|
|
|
// SetGrainSize() / ClearGrainSize() should've made our buffers match.
|
|
if (mixBuffer != NULL && grainSize > 0) {
|
|
p.DoArray(mixBuffer, grainSize * 2);
|
|
}
|
|
if (sendBuffer != NULL && grainSize > 0) {
|
|
p.DoArray(sendBuffer, grainSize * 2);
|
|
}
|
|
if (resampleBuffer != NULL && grainSize > 0) {
|
|
p.DoArray(resampleBuffer, grainSize * 4 + 3);
|
|
}
|
|
|
|
int n = PSP_SAS_VOICES_MAX;
|
|
p.Do(n);
|
|
if (n != PSP_SAS_VOICES_MAX)
|
|
{
|
|
ERROR_LOG(HLE, "Savestate failure: wrong number of SAS voices");
|
|
return;
|
|
}
|
|
p.DoArray(voices, ARRAY_SIZE(voices));
|
|
p.Do(waveformEffect);
|
|
}
|
|
|
|
void SasVoice::Reset() {
|
|
resampleHist[0] = 0;
|
|
resampleHist[1] = 0;
|
|
}
|
|
|
|
void SasVoice::KeyOn() {
|
|
envelope.KeyOn();
|
|
switch (type) {
|
|
case VOICETYPE_VAG:
|
|
if (Memory::IsValidAddress(vagAddr)) {
|
|
vag.Start(vagAddr, vagSize, loop);
|
|
} else {
|
|
ERROR_LOG(SASMIX, "Invalid VAG address %08x", vagAddr);
|
|
return;
|
|
}
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
playing = true;
|
|
on = true;
|
|
paused = false;
|
|
sampleFrac = 0;
|
|
}
|
|
|
|
void SasVoice::KeyOff() {
|
|
on = false;
|
|
envelope.KeyOff();
|
|
}
|
|
|
|
void SasVoice::ChangedParams(bool changedVag) {
|
|
if (!playing && on) {
|
|
playing = true;
|
|
if (changedVag)
|
|
vag.Start(vagAddr, vagSize, loop);
|
|
}
|
|
// TODO: restart VAG somehow
|
|
}
|
|
|
|
void SasVoice::DoState(PointerWrap &p)
|
|
{
|
|
auto s = p.Section("SasVoice", 1, 3);
|
|
if (!s)
|
|
return;
|
|
|
|
p.Do(playing);
|
|
p.Do(paused);
|
|
p.Do(on);
|
|
|
|
p.Do(type);
|
|
|
|
p.Do(vagAddr);
|
|
p.Do(vagSize);
|
|
p.Do(pcmAddr);
|
|
p.Do(pcmSize);
|
|
p.Do(pcmIndex);
|
|
if (s >= 2) {
|
|
p.Do(pcmLoopPos);
|
|
} else {
|
|
pcmLoopPos = 0;
|
|
}
|
|
p.Do(sampleRate);
|
|
|
|
p.Do(sampleFrac);
|
|
p.Do(pitch);
|
|
p.Do(loop);
|
|
if (s < 2 && type == VOICETYPE_PCM) {
|
|
// We set loop incorrectly before, and always looped.
|
|
// Let's keep always looping, since it's usually right.
|
|
loop = true;
|
|
}
|
|
|
|
p.Do(noiseFreq);
|
|
|
|
p.Do(volumeLeft);
|
|
p.Do(volumeRight);
|
|
if (s < 3) {
|
|
// There were extra variables here that were for the same purpose.
|
|
p.Do(effectLeft);
|
|
p.Do(effectRight);
|
|
}
|
|
p.Do(effectLeft);
|
|
p.Do(effectRight);
|
|
p.DoArray(resampleHist, ARRAY_SIZE(resampleHist));
|
|
|
|
envelope.DoState(p);
|
|
vag.DoState(p);
|
|
atrac3.DoState(p);
|
|
}
|
|
|
|
ADSREnvelope::ADSREnvelope()
|
|
: attackRate(0),
|
|
decayRate(0),
|
|
sustainRate(0),
|
|
releaseRate(0),
|
|
attackType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE),
|
|
decayType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
|
|
sustainType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
|
|
sustainLevel(0),
|
|
releaseType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
|
|
state_(STATE_OFF),
|
|
height_(0) {
|
|
}
|
|
|
|
void ADSREnvelope::WalkCurve(int type, int rate) {
|
|
s64 expDelta;
|
|
switch (type) {
|
|
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE:
|
|
height_ += rate;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE:
|
|
height_ -= rate;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT:
|
|
if (height_ <= (s64)PSP_SAS_ENVELOPE_HEIGHT_MAX * 3 / 4) {
|
|
height_ += rate;
|
|
} else {
|
|
height_ += rate / 4;
|
|
}
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE:
|
|
expDelta = height_ - PSP_SAS_ENVELOPE_HEIGHT_MAX;
|
|
// Flipping the sign so that we can shift in the top bits.
|
|
expDelta += (-expDelta * rate) >> 32;
|
|
height_ = expDelta + PSP_SAS_ENVELOPE_HEIGHT_MAX - (rate + 3UL) / 4UL;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_EXPONENT_INCREASE:
|
|
expDelta = height_ - PSP_SAS_ENVELOPE_HEIGHT_MAX;
|
|
// Flipping the sign so that we can shift in the top bits.
|
|
expDelta += (-expDelta * rate) >> 32;
|
|
height_ = expDelta + 0x4000 + PSP_SAS_ENVELOPE_HEIGHT_MAX;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_DIRECT:
|
|
height_ = rate; // Simple :)
|
|
break;
|
|
}
|
|
}
|
|
|
|
void ADSREnvelope::SetState(ADSRState state) {
|
|
if (height_ > PSP_SAS_ENVELOPE_HEIGHT_MAX) {
|
|
height_ = PSP_SAS_ENVELOPE_HEIGHT_MAX;
|
|
}
|
|
// TODO: Also check for height_ < 0 and set to 0?
|
|
state_ = state;
|
|
}
|
|
|
|
inline void ADSREnvelope::Step() {
|
|
switch (state_) {
|
|
case STATE_ATTACK:
|
|
WalkCurve(attackType, attackRate);
|
|
if (height_ >= PSP_SAS_ENVELOPE_HEIGHT_MAX || height_ < 0)
|
|
SetState(STATE_DECAY);
|
|
break;
|
|
case STATE_DECAY:
|
|
WalkCurve(decayType, decayRate);
|
|
if (height_ < sustainLevel)
|
|
SetState(STATE_SUSTAIN);
|
|
break;
|
|
case STATE_SUSTAIN:
|
|
WalkCurve(sustainType, sustainRate);
|
|
if (height_ <= 0) {
|
|
height_ = 0;
|
|
SetState(STATE_RELEASE);
|
|
}
|
|
break;
|
|
case STATE_RELEASE:
|
|
WalkCurve(releaseType, releaseRate);
|
|
if (height_ <= 0) {
|
|
height_ = 0;
|
|
SetState(STATE_OFF);
|
|
}
|
|
break;
|
|
case STATE_OFF:
|
|
// Do nothing
|
|
break;
|
|
|
|
case STATE_KEYON:
|
|
height_ = 0;
|
|
SetState(STATE_KEYON_STEP);
|
|
break;
|
|
case STATE_KEYON_STEP:
|
|
// This entire state is pretty much a hack to reproduce PSP behavior.
|
|
// The STATE_KEYON state is a real state, but not sure how it switches.
|
|
// It takes 32 steps at 0 for keyon to "kick in", 31 should shift to 0 anyway.
|
|
height_++;
|
|
if (height_ >= 31) {
|
|
height_ = 0;
|
|
SetState(STATE_ATTACK);
|
|
}
|
|
break;
|
|
}
|
|
}
|
|
|
|
void ADSREnvelope::KeyOn() {
|
|
SetState(STATE_KEYON);
|
|
}
|
|
|
|
void ADSREnvelope::KeyOff() {
|
|
SetState(STATE_RELEASE);
|
|
}
|
|
|
|
void ADSREnvelope::End() {
|
|
SetState(STATE_OFF);
|
|
height_ = 0;
|
|
}
|
|
|
|
void ADSREnvelope::DoState(PointerWrap &p) {
|
|
auto s = p.Section("ADSREnvelope", 1, 2);
|
|
if (!s) {
|
|
return;
|
|
}
|
|
|
|
p.Do(attackRate);
|
|
p.Do(decayRate);
|
|
p.Do(sustainRate);
|
|
p.Do(releaseRate);
|
|
p.Do(attackType);
|
|
p.Do(decayType);
|
|
p.Do(sustainType);
|
|
p.Do(sustainLevel);
|
|
p.Do(releaseType);
|
|
if (s < 2) {
|
|
p.Do(state_);
|
|
if (state_ == 4) {
|
|
state_ = STATE_OFF;
|
|
}
|
|
int stepsLegacy;
|
|
p.Do(stepsLegacy);
|
|
} else {
|
|
p.Do(state_);
|
|
}
|
|
p.Do(height_);
|
|
}
|