mirror of https://github.com/PCSX2/pcsx2.git
324 lines
8.3 KiB
C++
324 lines
8.3 KiB
C++
/* ZeroSPU2
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* Copyright (C) 2006-2010 zerofrog
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*
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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; either version 2 of the License, or
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* (at your option) any later version.
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*
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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 for more details.
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*
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* You should have received a copy of the GNU General Public License
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* along with this program; if not, write to the Free Software
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* Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA
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*/
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#include "zerospu2.h"
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#include "zeroworker.h"
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#include "SoundTouch/SoundTouch.h"
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#include "SoundTouch/WavFile.h"
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s32 g_logsound = 0;
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WavOutFile* g_pWavRecord=NULL; // used for recording
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const s32 f[5][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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s32 predict_nr, shift_factor, flags;
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s32 s_1, s_2;
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static __forceinline s32 SetPacket(s32 val)
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{
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s32 ret;
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if (val & 0x8000) val |= 0xffff0000;
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ret = (val >> shift_factor);
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ret += ((s_1 * f[predict_nr][0]) >> 6) + ((s_2 * f[predict_nr][1]) >> 6);
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s_2 = s_1;
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s_1 = ret;
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return ret;
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}
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void SPU2Loop(VOICE_PROCESSED* pChannel, u32 ch)
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{
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u8* start;
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u32 nSample;
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for (u32 ns = 0; ns < NSSIZE; ns++)
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{
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// fmod freq channel
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if (pChannel->bFMod == 1 && iFMod[ns]) pChannel->FModChangeFrequency(ns);
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while(pChannel->spos >= 0x10000 )
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{
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if (pChannel->iSBPos == 28) // 28 reached?
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{
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start=pChannel->pCurr; // set up the current pos
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// special "stop" sign
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if (start == (u8*)-1) //!pChannel->bOn
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{
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pChannel->bOn = false; // -> turn everything off
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pChannel->ADSRX.lVolume = 0;
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pChannel->ADSRX.EnvelopeVol = 0;
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return; // -> and done for this channel
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}
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predict_nr = (s32)start[0];
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shift_factor = predict_nr&0xf;
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predict_nr >>= 4;
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flags=(s32)start[1];
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start += 2;
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pChannel->iSBPos = 0;
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// decode the 16byte packet
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s_1 = pChannel->s_1;
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s_2 = pChannel->s_2;
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for (nSample=0; nSample<28; ++start)
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{
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s32 d = (s32)*start;
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s32 s;
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s = ((d & 0xf)<<12);
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pChannel->SB[nSample++] = SetPacket(s);
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s = ((d & 0xf0) << 8);
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pChannel->SB[nSample++] = SetPacket(s);
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}
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// irq occurs no matter what core accesses the address
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for (s32 core = 0; core < 2; ++core)
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{
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if (spu2attr(core).irq) // some callback and irq active?
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{
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// if irq address reached or irq on looping addr, when stop/loop flag is set
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u8* pirq = (u8*)pSpuIrq[core];
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if ((pirq > (start - 16) && pirq <= start) ||
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((flags & 1) && (pirq > (pChannel->pLoop - 16) && pirq <= pChannel->pLoop)))
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{
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IRQINFO |= 4 << core;
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SPU2_LOG("SPU2Worker:interrupt\n");
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irqCallbackSPU2();
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}
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}
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}
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// flag handler
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if ((flags & 4) && (!pChannel->bIgnoreLoop))
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pChannel->pLoop = start - 16; // loop address
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if (flags & 1) // 1: stop/loop
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{
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// We play this block out first...
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dwEndChannel2[ch / 24] |= (1 << (ch % 24));
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if (flags != 3 || pChannel->pLoop == NULL)
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{ // and checking if pLoop is set avoids crashes, yeah
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start = (u8*)-1;
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pChannel->bStop = true;
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pChannel->bIgnoreLoop = false;
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}
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else
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{
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start = pChannel->pLoop;
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}
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}
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pChannel->pCurr = start; // store values for next cycle
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pChannel->s_1 = s_1;
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pChannel->s_2 = s_2;
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}
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pChannel->StoreInterpolationVal(pChannel->SB[pChannel->iSBPos++]); // get sample data
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pChannel->spos -= 0x10000;
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}
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s32 sval = (MixADSR(pChannel) * pChannel->iGetVal()) / 1023; // mix adsr with noise or sample val.
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if (pChannel->bFMod == 2) // fmod freq channel
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{
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// -> store 1T sample data, use that to do fmod on next channel
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if (!pChannel->bNoise) iFMod[ns] = sval;
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}
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else
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{
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if (pChannel->bVolumeL)
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s_buffers[ns][0] += (sval * pChannel->leftvol) >> 14;
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if (pChannel->bVolumeR)
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s_buffers[ns][1] += (sval * pChannel->rightvol) >> 14;
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}
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pChannel->spos += pChannel->sinc;
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}
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}
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// simulate SPU2 for 1ms
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void SPU2Worker()
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{
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// assume s_buffers are zeroed out
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if (dwNewChannel2[0] || dwNewChannel2[1]) s_pAudioBuffers[s_nCurBuffer].newchannels++;
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VOICE_PROCESSED* pChannel = voices;
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for (u32 ch=0; ch < SPU_NUMBER_VOICES; ch++, pChannel++) // loop em all... we will collect 1 ms of sound of each playing channel
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{
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if (pChannel->bNew)
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{
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pChannel->StartSound(); // start new sound
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dwEndChannel2[ch / 24] &= ~(1 << (ch % 24)); // clear end channel bit
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dwNewChannel2[ch / 24] &= ~(1 << (ch % 24)); // clear channel bit
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}
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if (!pChannel->bOn) continue;
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if (pChannel->iActFreq != pChannel->iUsedFreq) // new psx frequency?
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pChannel->VoiceChangeFrequency();
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// loop until 1 ms of data is reached
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SPU2Loop(pChannel, ch);
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}
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// mix all channels
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MixChannels(0);
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MixChannels(1);
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if ( g_bPlaySound )
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{
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assert( s_pCurOutput != NULL);
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for (u32 ns = 0; ns < NSSIZE; ns++)
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{
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// clamp and write
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clampandwrite16(s_pCurOutput[0],s_buffers[ns][0]);
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clampandwrite16(s_pCurOutput[1],s_buffers[ns][1]);
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s_pCurOutput += 2;
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s_buffers[ns][0] = 0;
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s_buffers[ns][1] = 0;
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}
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// check if end reached
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if ((uptr)s_pCurOutput - (uptr)s_pAudioBuffers[s_nCurBuffer].pbuf >= 4 * NS_TOTAL_SIZE)
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{
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if ( conf.options & OPTION_RECORDING )
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{
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static s32 lastrectime = 0;
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if (timeGetTime() - lastrectime > 5000)
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{
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WARN_LOG("ZeroSPU2: recording\n");
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lastrectime = timeGetTime();
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}
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LogRawSound(s_pAudioBuffers[s_nCurBuffer].pbuf, 4, s_pAudioBuffers[s_nCurBuffer].pbuf+2, 4, NS_TOTAL_SIZE);
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}
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if ( s_nQueuedBuffers >= ArraySize(s_pAudioBuffers)-1 )
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{
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//ZeroSPU2: dropping packets! game too fast
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s_nDropPacket += NSFRAMES;
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s_GlobalTimeStamp = GetMicroTime();
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}
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else {
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// submit to final mixer
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#ifdef ZEROSPU2_DEVBUILD
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if ( g_logsound )
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LogRawSound(s_pAudioBuffers[s_nCurBuffer].pbuf, 4, s_pAudioBuffers[s_nCurBuffer].pbuf + 2, 4, NS_TOTAL_SIZE);
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#endif
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if ( g_startcount == 0xffffffff )
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{
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g_startcount = timeGetTime();
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g_packetcount = 0;
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}
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if ( conf.options & OPTION_TIMESTRETCH )
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{
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u32 newtotal = s_nTotalDuration - s_nDurations[s_nCurDuration];
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u64 newtime = GetMicroTime();
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u32 duration;
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if (s_GlobalTimeStamp == 0) s_GlobalTimeStamp = newtime - NSFRAMES * 1000;
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duration = (u32)(newtime-s_GlobalTimeStamp);
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s_nDurations[s_nCurDuration] = duration;
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s_nTotalDuration = newtotal + duration;
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s_nCurDuration = (s_nCurDuration+1)%ArraySize(s_nDurations);
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s_GlobalTimeStamp = newtime;
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s_pAudioBuffers[s_nCurBuffer].timestamp = timeGetTime();
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s_pAudioBuffers[s_nCurBuffer].avgtime = s_nTotalDuration/ArraySize(s_nDurations);
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}
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s_pAudioBuffers[s_nCurBuffer].len = 4 * NS_TOTAL_SIZE;
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InterlockedExchangeAdd((long*)&s_nQueuedBuffers, 1);
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s_nCurBuffer = (s_nCurBuffer+1)%ArraySize(s_pAudioBuffers);
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s_pAudioBuffers[s_nCurBuffer].newchannels = 0; // reset
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}
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// restart
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s_pCurOutput = (s16*)s_pAudioBuffers[s_nCurBuffer].pbuf;
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}
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}
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}
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// size is in bytes
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void LogPacketSound(void* packet, s32 memsize)
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{
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u16 buf[28];
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u8* pstart = (u8*)packet;
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s_1 = s_2 = 0;
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for (s32 i = 0; i < memsize; i += 16)
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{
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predict_nr = (s32)pstart[0];
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shift_factor = predict_nr&0xf;
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predict_nr >>= 4;
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pstart += 2;
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for (s32 nSample = 0;nSample < 28; ++pstart)
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{
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s32 d = (s32)*pstart;
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s32 temp;
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temp = ((d & 0xf) << 12);
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buf[nSample++] = SetPacket(temp);
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temp = ((d & 0xf0) << 8);
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buf[nSample++] = SetPacket(temp);
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}
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LogRawSound(buf, 2, buf, 2, 28);
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}
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}
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void LogRawSound(void* pleft, s32 leftstride, void* pright, s32 rightstride, s32 numsamples)
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{
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if (g_pWavRecord == NULL )
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g_pWavRecord = new WavOutFile(RECORD_FILENAME, SAMPLE_RATE, 16, 2);
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u8* left = (u8*)pleft;
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u8* right = (u8*)pright;
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static vector<s16> tempbuf;
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tempbuf.resize(2 * numsamples);
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for (s32 i = 0; i < numsamples; ++i)
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{
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tempbuf[2*i+0] = *(s16*)left;
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tempbuf[2*i+1] = *(s16*)right;
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left += leftstride;
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right += rightstride;
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}
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g_pWavRecord->write(&tempbuf[0], numsamples*2);
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}
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