mirror of
https://github.com/hrydgard/ppsspp.git
synced 2024-11-23 21:39:52 +00:00
6b413a808f
It makes sense to match, and this sounds right in Sol Trigger. Haven't tested in detail yet, but it should prevent cutting off sounds.
314 lines
6.9 KiB
C++
314 lines
6.9 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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// This is not really hardware, it's a software audio mixer running on the Media Engine.
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// From the perspective of a PSP app though, it might as well be.
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#pragma once
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#include "Common/CommonTypes.h"
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#include "Core/HW/BufferQueue.h"
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#include "Core/HW/SasReverb.h"
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class PointerWrap;
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enum {
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PSP_SAS_VOICES_MAX = 32,
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PSP_SAS_PITCH_MIN = 0x0000,
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PSP_SAS_PITCH_BASE = 0x1000,
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PSP_SAS_PITCH_BASE_SHIFT = 12,
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PSP_SAS_PITCH_MAX = 0x4000,
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PSP_SAS_VOL_MAX = 0x1000,
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PSP_SAS_ADSR_CURVE_MODE_LINEAR_INCREASE = 0,
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PSP_SAS_ADSR_CURVE_MODE_LINEAR_DECREASE = 1,
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PSP_SAS_ADSR_CURVE_MODE_LINEAR_BENT = 2,
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PSP_SAS_ADSR_CURVE_MODE_EXPONENT_DECREASE = 3,
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PSP_SAS_ADSR_CURVE_MODE_EXPONENT_INCREASE = 4,
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PSP_SAS_ADSR_CURVE_MODE_DIRECT = 5,
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PSP_SAS_ADSR_ATTACK = 1,
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PSP_SAS_ADSR_DECAY = 2,
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PSP_SAS_ADSR_SUSTAIN = 4,
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PSP_SAS_ADSR_RELEASE = 8,
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PSP_SAS_ENVELOPE_HEIGHT_MAX = 0x40000000,
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PSP_SAS_ENVELOPE_FREQ_MAX = 0x7FFFFFFF,
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PSP_SAS_EFFECT_TYPE_OFF = -1,
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PSP_SAS_EFFECT_TYPE_ROOM = 0,
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PSP_SAS_EFFECT_TYPE_STUDIO_SMALL= 1,
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PSP_SAS_EFFECT_TYPE_STUDIO_MEDIUM = 2,
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PSP_SAS_EFFECT_TYPE_STUDIO_LARGE = 3,
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PSP_SAS_EFFECT_TYPE_HALL = 4,
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PSP_SAS_EFFECT_TYPE_SPACE = 5,
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PSP_SAS_EFFECT_TYPE_ECHO = 6,
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PSP_SAS_EFFECT_TYPE_DELAY = 7,
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PSP_SAS_EFFECT_TYPE_PIPE = 8,
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PSP_SAS_EFFECT_TYPE_MAX = 8,
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PSP_SAS_OUTPUTMODE_MIXED = 0,
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PSP_SAS_OUTPUTMODE_RAW = 1,
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};
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struct WaveformEffect {
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int type;
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int delay;
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int feedback;
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int leftVol;
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int rightVol;
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int isDryOn;
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int isWetOn;
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};
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enum VoiceType {
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VOICETYPE_OFF,
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VOICETYPE_VAG, // default
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VOICETYPE_NOISE,
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VOICETYPE_TRIWAVE, // are these used? there are functions for them (sceSetTriangularWave)
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VOICETYPE_PULSEWAVE,
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VOICETYPE_PCM,
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VOICETYPE_ATRAC3,
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};
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// VAG is a Sony ADPCM audio compression format, which goes all the way back to the PSX.
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// It compresses 28 16-bit samples into a block of 16 bytes.
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class VagDecoder {
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public:
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VagDecoder() : data_(0), read_(0), end_(true) {
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memset(samples, 0, sizeof(samples));
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}
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void Start(u32 dataPtr, u32 vagSize, bool loopEnabled);
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void GetSamples(s16 *outSamples, int numSamples);
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void DecodeBlock(u8 *&readp);
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bool End() const { return end_; }
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void DoState(PointerWrap &p);
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u32 GetReadPtr() const { return read_; }
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private:
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s16 samples[28];
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int curSample;
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u32 data_;
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u32 read_;
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int curBlock_;
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int loopStartBlock_;
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int numBlocks_;
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// rolling state. start at 0, should probably reset to 0 on loops?
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int s_1;
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int s_2;
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bool loopEnabled_;
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bool loopAtNextBlock_;
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bool end_;
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};
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class SasAtrac3 {
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public:
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SasAtrac3() : contextAddr_(0), atracID_(-1), sampleQueue_(0), end_(false) {}
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~SasAtrac3() { if (sampleQueue_) delete sampleQueue_; }
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int setContext(u32 context);
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void getNextSamples(s16 *outbuf, int wantedSamples);
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int addStreamData(u32 bufPtr, u32 addbytes);
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void DoState(PointerWrap &p);
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bool End() const {
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return end_;
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}
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private:
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u32 contextAddr_;
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int atracID_;
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BufferQueue *sampleQueue_;
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bool end_;
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};
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class ADSREnvelope {
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public:
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ADSREnvelope();
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void SetSimpleEnvelope(u32 ADSREnv1, u32 ADSREnv2);
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void WalkCurve(int type, int rate);
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void KeyOn();
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void KeyOff();
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void End();
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inline void Step();
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int GetHeight() const {
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return height_ > (s64)PSP_SAS_ENVELOPE_HEIGHT_MAX ? PSP_SAS_ENVELOPE_HEIGHT_MAX : height_;
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}
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bool NeedsKeyOn() const {
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return state_ == STATE_KEYON;
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}
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bool HasEnded() const {
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return state_ == STATE_OFF;
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}
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int attackRate;
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int decayRate;
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int sustainRate;
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int releaseRate;
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int attackType;
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int decayType;
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int sustainType;
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int sustainLevel;
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int releaseType;
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void DoState(PointerWrap &p);
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private:
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// Actual PSP values.
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enum ADSRState {
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// Okay, this one isn't a real value but it might be.
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STATE_KEYON_STEP = -42,
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STATE_KEYON = -2,
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STATE_OFF = -1,
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STATE_ATTACK = 0,
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STATE_DECAY = 1,
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STATE_SUSTAIN = 2,
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STATE_RELEASE = 3,
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};
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void SetState(ADSRState state);
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ADSRState state_;
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s64 height_; // s64 to avoid having to care about overflow when calculating. TODO: this should be fine as s32
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};
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// A SAS voice.
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// TODO: Look into pre-decoding the VAG samples on SetVoice instead of decoding them on the fly.
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// It's not very likely that games encode VAG dynamically.
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struct SasVoice {
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SasVoice()
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: playing(false),
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paused(false),
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on(false),
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type(VOICETYPE_OFF),
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vagAddr(0),
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vagSize(0),
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pcmAddr(0),
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pcmSize(0),
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pcmIndex(0),
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pcmLoopPos(0),
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sampleRate(44100),
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sampleFrac(0),
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pitch(PSP_SAS_PITCH_BASE),
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loop(false),
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noiseFreq(0),
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volumeLeft(PSP_SAS_VOL_MAX),
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volumeRight(PSP_SAS_VOL_MAX),
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effectLeft(PSP_SAS_VOL_MAX),
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effectRight(PSP_SAS_VOL_MAX) {
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memset(resampleHist, 0, sizeof(resampleHist));
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}
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void Reset();
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void KeyOn();
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void KeyOff();
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void ChangedParams(bool changedVag);
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void DoState(PointerWrap &p);
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void ReadSamples(s16 *output, int numSamples);
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bool HaveSamplesEnded() const;
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bool playing;
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bool paused; // a voice can be playing AND paused. In that case, it won't play.
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bool on; // key-on, key-off.
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VoiceType type;
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u32 vagAddr;
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u32 vagSize;
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u32 pcmAddr;
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int pcmSize;
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int pcmIndex;
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int pcmLoopPos;
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int sampleRate;
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int sampleFrac;
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int pitch;
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bool loop;
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int noiseFreq;
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int volumeLeft;
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int volumeRight;
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// volume to "Send" (audio-lingo) to the effects processing engine, like reverb
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int effectLeft;
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int effectRight;
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s16 resampleHist[2];
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ADSREnvelope envelope;
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VagDecoder vag;
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SasAtrac3 atrac3;
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};
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class SasInstance {
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public:
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SasInstance();
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~SasInstance();
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void ClearGrainSize();
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void SetGrainSize(int newGrainSize);
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int GetGrainSize() const { return grainSize; }
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int maxVoices;
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int sampleRate;
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int outputMode;
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int *mixBuffer;
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int *sendBuffer;
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s16 *sendBufferDownsampled;
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s16 *sendBufferProcessed;
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s16 *resampleBuffer;
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FILE *audioDump;
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void Mix(u32 outAddr, u32 inAddr = 0, int leftVol = 0, int rightVol = 0);
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void MixVoice(SasVoice &voice);
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// Applies reverb to send buffer, according to waveformEffect.
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void ApplyWaveformEffect();
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void SetWaveformEffectType(int type);
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void WriteMixedOutput(s16 *outp, const s16 *inp, int leftVol, int rightVol);
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void GetDebugText(char *text, size_t bufsize);
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void DoState(PointerWrap &p);
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SasVoice voices[PSP_SAS_VOICES_MAX];
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WaveformEffect waveformEffect;
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private:
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SasReverb reverb_;
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int grainSize;
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};
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