mirror of
https://github.com/libretro/ppsspp.git
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822 lines
22 KiB
C++
822 lines
22 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 "../MemMap.h"
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#include "Core/HLE/sceAtrac.h"
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#include "Core/Config.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 s8 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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// Padding to prevent overflow.
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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{ 0, 0 },
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};
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void VagDecoder::Start(u32 data, int vagSize, bool loopEnabled) {
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loopEnabled_ = loopEnabled;
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loopAtNextBlock_ = false;
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loopStartBlock_ = 0;
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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 *&readp) {
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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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for (int i = 0; i < 28; i += 2) {
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int d = *readp++;
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int s = (short)((d & 0xf) << 12);
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DecodeSample(i, s >> shift_factor, predict_nr);
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s = (short)((d & 0xf0) << 8);
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DecodeSample(i + 1, s >> shift_factor, predict_nr);
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}
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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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}
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inline void VagDecoder::DecodeSample(int i, int sample, int predict_nr) {
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samples[i] = (int) (sample + ((s_1 * f[predict_nr][0] + s_2 * f[predict_nr][1]) >> 6));
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s_2 = s_1;
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s_1 = samples[i];
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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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{
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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::GetPointer(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] = end_ ? 0 : 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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{
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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 Atrac3plus_Decoder::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 Atrac3plus_Decoder::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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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 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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return 0x80000000 >> ((bitfield1 >> 4) & 0x000F);
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}
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static int getSustainRate(int bitfield2) {
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return simpleRate(bitfield2 >> 6);
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}
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static int getSustainType(int bitfield2) {
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switch (bitfield2 >> 13) {
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case 0: return PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE;
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case 2: return PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE;
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case 4: return PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT;
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case 6: return PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE;
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}
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ERROR_LOG(SASMIX,"sasSetSimpleADSR,ERROR_SAS_INVALID_ADSR_CURVE_MODE");
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return 0;
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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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return (0x40000000 >> (n + 2));
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}
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return (0x40000000 >> 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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}
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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(0),
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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 = -1;
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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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static inline s16 clamp_s16(int i) {
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if (i > 32767)
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return 32767;
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if (i < -32768)
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return -32768;
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return i;
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}
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void SasInstance::Mix(u32 outAddr, u32 inAddr, int leftVol, int rightVol) {
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int voicesPlayingCount = 0;
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for (int v = 0; v < PSP_SAS_VOICES_MAX; v++) {
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SasVoice &voice = voices[v];
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if (!voice.playing || voice.paused)
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continue;
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voicesPlayingCount++;
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// TODO: Special case no-resample case for speed
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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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case VOICETYPE_PCM:
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if (voice.type == VOICETYPE_PCM && !voice.pcmAddr)
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break;
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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 = (voice.sampleFrac + grainSize * voice.pitch) / PSP_SAS_PITCH_BASE;
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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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// Read N samples into the resample buffer. Could do either PCM or VAG here.
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switch (voice.type) {
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case VOICETYPE_VAG:
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{
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voice.vag.GetSamples(resampleBuffer + 2, numSamples);
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if (voice.vag.End()) {
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// NOTICE_LOG(SAS, "Hit end of VAG audio");
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voice.playing = false;
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voice.on = false; // ??
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}
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}
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break;
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case VOICETYPE_PCM:
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{
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u32 size = std::min(voice.pcmSize * 2 - voice.pcmIndex, (int)(numSamples * sizeof(s16)));
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memset(resampleBuffer + 2, 0, numSamples * sizeof(s16));
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if (!voice.on) {
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voice.pcmIndex = 0;
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break;
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}
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Memory::Memcpy(resampleBuffer + 2, voice.pcmAddr + voice.pcmIndex, size);
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voice.pcmIndex += size;
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if (voice.pcmIndex >= voice.pcmSize * 2) {
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voice.pcmIndex = 0;
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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 = voice.atrac3.getNextSamples(resampleBuffer + 2, numSamples);
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if (ret) {
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// Hit atrac3 voice end
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voice.playing = false;
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voice.on = false; // ??
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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(resampleBuffer + 2, 0, numSamples * sizeof(s16));
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}
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break;
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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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u32 sampleFrac = voice.sampleFrac;
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const int MAX_CONFIG_VOLUME = 20;
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int volumeShift = (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
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// Ideally, the shift right should be there too but for now I'm concerned about
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// not overflowing.
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mixBuffer[i * 2] += (sample * voice.volumeLeft ) >> volumeShift; // Max = 16 and Min = 12(default)
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mixBuffer[i * 2 + 1] += (sample * voice.volumeRight) >> volumeShift; // Max = 16 and Min = 12(default)
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sendBuffer[i * 2] += sample * voice.volumeLeftSend >> 12;
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sendBuffer[i * 2 + 1] += sample * voice.volumeRightSend >> 12;
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voice.envelope.Step();
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}
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voice.sampleFrac = sampleFrac;
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// Let's hope grainSize is a power of 2.
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//voice.sampleFrac &= grainSize * PSP_SAS_PITCH_BASE - 1;
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voice.sampleFrac -= numSamples * PSP_SAS_PITCH_BASE;
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if (voice.envelope.HasEnded())
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{
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// NOTICE_LOG(SAS, "Hit end of envelope");
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voice.playing = false;
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}
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}
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}
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//if (voicesPlayingCount)
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// NOTICE_LOG(SAS, "Sas mixed %i voices", voicesPlayingCount);
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// Okay, apply effects processing to the Send buffer alone here.
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// Reverb, echo, what have you.
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// TODO
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// Alright, all voices mixed. Let's convert and clip, and at the same time, wipe mixBuffer for next time. Could also dither.
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s16 *outp = (s16 *)Memory::GetPointer(outAddr);
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const s16 *inp = inAddr ? (s16*)Memory::GetPointer(inAddr) : 0;
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if (outputMode == 0) {
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if (inp) {
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for (int i = 0; i < grainSize * 2; i += 2) {
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int sampleL = mixBuffer[i] + sendBuffer[i] + ((*inp++) * leftVol >> 12);
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int sampleR = mixBuffer[i + 1] + sendBuffer[i + 1] + ((*inp++) * rightVol >> 12);
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*outp++ = clamp_s16(sampleL);
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*outp++ = clamp_s16(sampleR);
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}
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} else {
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for (int i = 0; i < grainSize * 2; i += 2) {
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*outp++ = clamp_s16(mixBuffer[i] + sendBuffer[i]);
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*outp++ = clamp_s16(mixBuffer[i + 1] + sendBuffer[i + 1]);
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}
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}
|
|
} else {
|
|
for (int i = 0; i < grainSize * 2; i += 2) {
|
|
int sampleL = mixBuffer[i] + sendBuffer[i];
|
|
if (inp)
|
|
sampleL += (*inp++) * leftVol >> 12;
|
|
*outp++ = clamp_s16(sampleL);
|
|
}
|
|
}
|
|
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::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;
|
|
}
|
|
|
|
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);
|
|
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);
|
|
p.Do(sampleRate);
|
|
|
|
p.Do(sampleFrac);
|
|
p.Do(pitch);
|
|
p.Do(loop);
|
|
|
|
p.Do(noiseFreq);
|
|
|
|
p.Do(volumeLeft);
|
|
p.Do(volumeRight);
|
|
p.Do(volumeLeftSend);
|
|
p.Do(volumeRightSend);
|
|
p.Do(effectLeft);
|
|
p.Do(effectRight);
|
|
p.DoArray(resampleHist, ARRAY_SIZE(resampleHist));
|
|
|
|
envelope.DoState(p);
|
|
vag.DoState(p);
|
|
atrac3.DoState(p);
|
|
}
|
|
|
|
// This is horribly stolen from JPCSP.
|
|
// Need to find a real solution.
|
|
static const short expCurve[] = {
|
|
0x0000, 0x0380, 0x06E4, 0x0A2D, 0x0D5B, 0x1072, 0x136F, 0x1653,
|
|
0x1921, 0x1BD9, 0x1E7B, 0x2106, 0x237F, 0x25E4, 0x2835, 0x2A73,
|
|
0x2CA0, 0x2EBB, 0x30C6, 0x32C0, 0x34AB, 0x3686, 0x3852, 0x3A10,
|
|
0x3BC0, 0x3D63, 0x3EF7, 0x4081, 0x41FC, 0x436E, 0x44D3, 0x462B,
|
|
0x477B, 0x48BF, 0x49FA, 0x4B2B, 0x4C51, 0x4D70, 0x4E84, 0x4F90,
|
|
0x5095, 0x5191, 0x5284, 0x5370, 0x5455, 0x5534, 0x5609, 0x56D9,
|
|
0x57A3, 0x5867, 0x5924, 0x59DB, 0x5A8C, 0x5B39, 0x5BE0, 0x5C81,
|
|
0x5D1C, 0x5DB5, 0x5E48, 0x5ED5, 0x5F60, 0x5FE5, 0x6066, 0x60E2,
|
|
0x615D, 0x61D2, 0x6244, 0x62B2, 0x631D, 0x6384, 0x63E8, 0x644A,
|
|
0x64A8, 0x6503, 0x655B, 0x65B1, 0x6605, 0x6653, 0x66A2, 0x66ED,
|
|
0x6737, 0x677D, 0x67C1, 0x6804, 0x6844, 0x6882, 0x68BF, 0x68F9,
|
|
0x6932, 0x6969, 0x699D, 0x69D2, 0x6A03, 0x6A34, 0x6A63, 0x6A8F,
|
|
0x6ABC, 0x6AE6, 0x6B0E, 0x6B37, 0x6B5D, 0x6B84, 0x6BA7, 0x6BCB,
|
|
0x6BED, 0x6C0E, 0x6C2D, 0x6C4D, 0x6C6B, 0x6C88, 0x6CA4, 0x6CBF,
|
|
0x6CD9, 0x6CF3, 0x6D0C, 0x6D24, 0x6D3B, 0x6D52, 0x6D68, 0x6D7D,
|
|
0x6D91, 0x6DA6, 0x6DB9, 0x6DCA, 0x6DDE, 0x6DEF, 0x6DFF, 0x6E10,
|
|
0x6E20, 0x6E30, 0x6E3E, 0x6E4C, 0x6E5A, 0x6E68, 0x6E76, 0x6E82,
|
|
0x6E8E, 0x6E9B, 0x6EA5, 0x6EB1, 0x6EBC, 0x6EC6, 0x6ED1, 0x6EDB,
|
|
0x6EE4, 0x6EED, 0x6EF6, 0x6EFE, 0x6F07, 0x6F10, 0x6F17, 0x6F20,
|
|
0x6F27, 0x6F2E, 0x6F35, 0x6F3C, 0x6F43, 0x6F48, 0x6F4F, 0x6F54,
|
|
0x6F5B, 0x6F60, 0x6F66, 0x6F6B, 0x6F70, 0x6F74, 0x6F79, 0x6F7E,
|
|
0x6F82, 0x6F87, 0x6F8A, 0x6F90, 0x6F93, 0x6F97, 0x6F9A, 0x6F9E,
|
|
0x6FA1, 0x6FA5, 0x6FA8, 0x6FAC, 0x6FAD, 0x6FB1, 0x6FB4, 0x6FB6,
|
|
0x6FBA, 0x6FBB, 0x6FBF, 0x6FC1, 0x6FC4, 0x6FC6, 0x6FC8, 0x6FC9,
|
|
0x6FCD, 0x6FCF, 0x6FD0, 0x6FD2, 0x6FD4, 0x6FD6, 0x6FD7, 0x6FD9,
|
|
0x6FDB, 0x6FDD, 0x6FDE, 0x6FDE, 0x6FE0, 0x6FE2, 0x6FE4, 0x6FE5,
|
|
0x6FE5, 0x6FE7, 0x6FE9, 0x6FE9, 0x6FEB, 0x6FEC, 0x6FEC, 0x6FEE,
|
|
0x6FEE, 0x6FF0, 0x6FF0, 0x6FF2, 0x6FF2, 0x6FF3, 0x6FF3, 0x6FF5,
|
|
0x6FF5, 0x6FF7, 0x6FF7, 0x6FF7, 0x6FF9, 0x6FF9, 0x6FF9, 0x6FFA,
|
|
0x6FFA, 0x6FFA, 0x6FFC, 0x6FFC, 0x6FFC, 0x6FFE, 0x6FFE, 0x6FFE,
|
|
0x7000
|
|
};
|
|
|
|
static int durationFromRate(int rate)
|
|
{
|
|
if (rate == 0) {
|
|
return PSP_SAS_ENVELOPE_FREQ_MAX;
|
|
} else {
|
|
// From experimental tests on a PSP:
|
|
// rate=0x7FFFFFFF => duration=0x10
|
|
// rate=0x3FFFFFFF => duration=0x22
|
|
// rate=0x1FFFFFFF => duration=0x44
|
|
// rate=0x0FFFFFFF => duration=0x81
|
|
// rate=0x07FFFFFF => duration=0xF1
|
|
// rate=0x03FFFFFF => duration=0x1B9
|
|
//
|
|
// The correct curve model is still unknown.
|
|
// We use the following approximation:
|
|
// duration = 0x7FFFFFFF / rate * 0x10
|
|
return PSP_SAS_ENVELOPE_FREQ_MAX / rate * 0x10;
|
|
}
|
|
}
|
|
|
|
const short expCurveReference = 0x7000;
|
|
|
|
// This needs a rewrite / rethink. Doing all this per sample is insane.
|
|
static int getExpCurveAt(int index, int duration) {
|
|
const short curveLength = sizeof(expCurve) / sizeof(short);
|
|
|
|
if (duration == 0) {
|
|
// Avoid division by zero, and thus undefined behaviour in conversion to int.
|
|
return 0;
|
|
}
|
|
|
|
float curveIndex = (index * curveLength) / (float) duration;
|
|
int curveIndex1 = (int) curveIndex;
|
|
int curveIndex2 = curveIndex1 + 1;
|
|
float curveIndexFraction = curveIndex - curveIndex1;
|
|
|
|
if (curveIndex1 < 0) {
|
|
return expCurve[0];
|
|
} else if (curveIndex2 >= curveLength || curveIndex2 < 0) {
|
|
return expCurve[curveLength - 1];
|
|
}
|
|
|
|
float sample = expCurve[curveIndex1] * (1.f - curveIndexFraction) + expCurve[curveIndex2] * curveIndexFraction;
|
|
return (short)(sample);
|
|
}
|
|
|
|
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_INCREASE),
|
|
sustainLevel(0x100),
|
|
releaseType(PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE),
|
|
state_(STATE_OFF),
|
|
steps_(0),
|
|
height_(0) {
|
|
}
|
|
|
|
void ADSREnvelope::WalkCurve(int rate, int type) {
|
|
short expFactor;
|
|
int duration;
|
|
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:
|
|
// NOTICE_LOG(SAS, "UNIMPL EXP DECR");
|
|
duration = durationFromRate(rate);
|
|
expFactor = getExpCurveAt(steps_, duration);
|
|
height_ = (s64)expFactor * PSP_SAS_ENVELOPE_HEIGHT_MAX / expCurveReference;
|
|
height_ = PSP_SAS_ENVELOPE_HEIGHT_MAX - height_;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_EXPONENT_INCREASE:
|
|
duration = durationFromRate(rate);
|
|
expFactor = getExpCurveAt(steps_, duration);
|
|
height_ = (s64)expFactor * PSP_SAS_ENVELOPE_HEIGHT_MAX / expCurveReference;
|
|
break;
|
|
|
|
case PSP_SAS_ADSR_CURVE_MODE_DIRECT:
|
|
height_ = rate; // Simple :)
|
|
break;
|
|
}
|
|
}
|
|
|
|
void ADSREnvelope::SetState(ADSRState state) {
|
|
steps_ = 0;
|
|
state_ = state;
|
|
}
|
|
|
|
void ADSREnvelope::Step() {
|
|
switch (state_) {
|
|
case STATE_ATTACK:
|
|
WalkCurve(attackRate, attackType);
|
|
if (height_ > PSP_SAS_ENVELOPE_HEIGHT_MAX || height_ < 0)
|
|
SetState(STATE_DECAY);
|
|
break;
|
|
case STATE_DECAY:
|
|
WalkCurve(decayRate, decayType);
|
|
if (height_ > PSP_SAS_ENVELOPE_HEIGHT_MAX || height_ < sustainLevel)
|
|
SetState(STATE_SUSTAIN);
|
|
break;
|
|
case STATE_SUSTAIN:
|
|
WalkCurve(sustainRate, sustainType);
|
|
if (height_ <= 0) {
|
|
height_ = 0;
|
|
SetState(STATE_RELEASE);
|
|
}
|
|
break;
|
|
case STATE_RELEASE:
|
|
WalkCurve(releaseRate, releaseType);
|
|
if (height_ <= 0) {
|
|
height_ = 0;
|
|
SetState(STATE_OFF);
|
|
}
|
|
break;
|
|
case STATE_OFF:
|
|
// Do nothing
|
|
break;
|
|
}
|
|
steps_++;
|
|
}
|
|
|
|
void ADSREnvelope::KeyOn() {
|
|
SetState(STATE_ATTACK);
|
|
height_ = 0;
|
|
}
|
|
|
|
void ADSREnvelope::KeyOff() {
|
|
SetState(STATE_RELEASE);
|
|
height_ = sustainLevel;
|
|
}
|
|
|
|
void ADSREnvelope::DoState(PointerWrap &p) {
|
|
auto s = p.Section("ADSREnvelope", 1);
|
|
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);
|
|
p.Do(state_);
|
|
p.Do(steps_);
|
|
p.Do(height_);
|
|
}
|