mirror of https://github.com/ZDoom/raze-gles.git
355 lines
8.1 KiB
C++
355 lines
8.1 KiB
C++
#include <algorithm>
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#include "streamsource.h"
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#include "../../libraries/music_common/fileio.h"
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/**
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* PlayStation XA (ADPCM) source support for MultiVoc
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* Adapted and remixed from superxa2wav
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*
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* taken from EDuke32 and adapted for GZDoom by Christoph Oelckers
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*/
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//#define NO_XA_HEADER
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enum
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{
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kNumOfSamples = 224,
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kNumOfSGs = 18,
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TTYWidth = 80,
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kBufSize = (kNumOfSGs*kNumOfSamples),
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kSamplesMono = (kNumOfSGs*kNumOfSamples),
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kSamplesStereo = (kNumOfSGs*kNumOfSamples/2),
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/* ADPCM */
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XA_DATA_START = (0x44-48)
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};
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inline float constexpr DblToPCMF(double dt) { return float(dt) * (1.f/32768.f); }
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typedef struct {
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MusicIO::FileInterface *reader;
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size_t committed;
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size_t length;
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bool blockIsMono;
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bool blockIs18K;
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bool finished;
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double t1, t2;
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double t1_x, t2_x;
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float block[kBufSize];
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} xa_data;
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typedef int8_t SoundGroup[128];
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typedef struct XASector {
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int8_t sectorFiller[48];
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SoundGroup SoundGroups[18];
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} XASector;
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static double K0[4] = {
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0.0,
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0.9375,
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1.796875,
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1.53125
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};
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static double K1[4] = {
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0.0,
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0.0,
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-0.8125,
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-0.859375
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};
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static int8_t getSoundData(int8_t *buf, int32_t unit, int32_t sample)
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{
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int8_t ret;
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int8_t *p;
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int32_t offset, shift;
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p = buf;
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shift = (unit%2) * 4;
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offset = 16 + (unit / 2) + (sample * 4);
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p += offset;
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ret = (*p >> shift) & 0x0F;
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if (ret > 7) {
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ret -= 16;
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}
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return ret;
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}
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static int8_t getFilter(const int8_t *buf, int32_t unit)
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{
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return (*(buf + 4 + unit) >> 4) & 0x03;
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}
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static int8_t getRange(const int8_t *buf, int32_t unit)
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{
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return *(buf + 4 + unit) & 0x0F;
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}
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static void decodeSoundSectMono(XASector *ssct, xa_data * xad)
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{
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size_t count = 0;
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int8_t snddat, filt, range;
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int32_t unit, sample;
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int32_t sndgrp;
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double tmp2, tmp3, tmp4, tmp5;
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auto &decodeBuf = xad->block;
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for (sndgrp = 0; sndgrp < kNumOfSGs; sndgrp++)
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{
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for (unit = 0; unit < 8; unit++)
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{
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range = getRange(ssct->SoundGroups[sndgrp], unit);
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filt = getFilter(ssct->SoundGroups[sndgrp], unit);
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for (sample = 0; sample < 28; sample++)
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{
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snddat = getSoundData(ssct->SoundGroups[sndgrp], unit, sample);
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tmp2 = (double)(1 << (12 - range));
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tmp3 = (double)snddat * tmp2;
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tmp4 = xad->t1 * K0[filt];
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tmp5 = xad->t2 * K1[filt];
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xad->t2 = xad->t1;
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xad->t1 = tmp3 + tmp4 + tmp5;
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decodeBuf[count++] = DblToPCMF(xad->t1);
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}
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}
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}
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}
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static void decodeSoundSectStereo(XASector *ssct, xa_data * xad)
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{
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size_t count = 0;
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int8_t snddat, filt, range;
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int8_t filt1, range1;
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int32_t unit, sample;
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int32_t sndgrp;
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double tmp2, tmp3, tmp4, tmp5;
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auto &decodeBuf = xad->block;
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for (sndgrp = 0; sndgrp < kNumOfSGs; sndgrp++)
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{
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for (unit = 0; unit < 8; unit+= 2)
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{
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range = getRange(ssct->SoundGroups[sndgrp], unit);
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filt = getFilter(ssct->SoundGroups[sndgrp], unit);
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range1 = getRange(ssct->SoundGroups[sndgrp], unit+1);
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filt1 = getFilter(ssct->SoundGroups[sndgrp], unit+1);
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for (sample = 0; sample < 28; sample++)
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{
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// Channel 1
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snddat = getSoundData(ssct->SoundGroups[sndgrp], unit, sample);
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tmp2 = (double)(1 << (12 - range));
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tmp3 = (double)snddat * tmp2;
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tmp4 = xad->t1 * K0[filt];
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tmp5 = xad->t2 * K1[filt];
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xad->t2 = xad->t1;
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xad->t1 = tmp3 + tmp4 + tmp5;
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decodeBuf[count++] = DblToPCMF(xad->t1);
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// Channel 2
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snddat = getSoundData(ssct->SoundGroups[sndgrp], unit+1, sample);
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tmp2 = (double)(1 << (12 - range1));
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tmp3 = (double)snddat * tmp2;
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tmp4 = xad->t1_x * K0[filt1];
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tmp5 = xad->t2_x * K1[filt1];
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xad->t2_x = xad->t1_x;
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xad->t1_x = tmp3 + tmp4 + tmp5;
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decodeBuf[count++] = DblToPCMF(xad->t1_x);
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}
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}
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}
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}
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//==========================================================================
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//
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// Get one decoded block of data
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//
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//==========================================================================
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static void getNextXABlock(xa_data *xad, bool looping )
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{
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XASector ssct;
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int coding;
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const int SUBMODE_REAL_TIME_SECTOR = (1 << 6);
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const int SUBMODE_FORM = (1 << 5);
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const int SUBMODE_AUDIO_DATA = (1 << 2);
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do
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{
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size_t bytes = xad->length - xad->reader->tell();
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if (sizeof(XASector) < bytes)
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bytes = sizeof(XASector);
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xad->reader->read(&ssct, (int)bytes);
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}
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while (ssct.sectorFiller[46] != (SUBMODE_REAL_TIME_SECTOR | SUBMODE_FORM | SUBMODE_AUDIO_DATA));
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coding = ssct.sectorFiller[47];
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xad->committed = 0;
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xad->blockIsMono = (coding & 3) == 0;
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xad->blockIs18K = (((coding >> 2) & 3) == 1);
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if (!xad->blockIsMono)
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{
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decodeSoundSectStereo(&ssct, xad);
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}
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else
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{
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decodeSoundSectMono(&ssct, xad);
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}
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if (xad->length == xad->reader->tell())
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{
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if (looping)
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{
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xad->reader->seek(XA_DATA_START, SEEK_SET);
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xad->t1 = xad->t2 = xad->t1_x = xad->t2_x = 0;
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}
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else
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xad->finished = true;
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}
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xad->finished = false;
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}
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//==========================================================================
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//
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// XASong
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//
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//==========================================================================
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class XASong : public StreamSource
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{
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public:
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XASong(MusicIO::FileInterface *readr);
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SoundStreamInfo GetFormat() override;
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bool Start() override;
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bool GetData(void *buffer, size_t len) override;
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protected:
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xa_data xad;
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};
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//==========================================================================
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//
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// XASong - Constructor
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//
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//==========================================================================
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XASong::XASong(MusicIO::FileInterface * reader)
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{
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reader->seek(0, SEEK_END);
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xad.length = reader->tell();
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reader->seek(XA_DATA_START, SEEK_SET);
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xad.reader = reader;
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xad.t1 = xad.t2 = xad.t1_x = xad.t2_x = 0;
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getNextXABlock(&xad, false);
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}
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SoundStreamInfo XASong::GetFormat()
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{
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auto SampleRate = xad.blockIs18K? 18900 : 37800;
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return { 64*1024, SampleRate, 2};
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}
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//==========================================================================
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//
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// XASong :: Play
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//
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//==========================================================================
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bool XASong::Start()
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{
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if (xad.finished && m_Looping)
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{
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xad.reader->seek(XA_DATA_START, SEEK_SET);
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xad.t1 = xad.t2 = xad.t1_x = xad.t2_x = 0;
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xad.finished = false;
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}
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return true;
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}
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//==========================================================================
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//
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// XASong :: Read
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//
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//==========================================================================
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bool XASong::GetData(void *vbuff, size_t len)
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{
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float *dest = (float*)vbuff;
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while (len > 0)
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{
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auto ptr = xad.committed;
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auto block = xad.block + ptr;
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if (ptr < kBufSize)
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{
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// commit the data
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if (xad.blockIsMono)
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{
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size_t numsamples = len / 8;
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size_t availdata = kBufSize - ptr;
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for(size_t tocopy = std::min(numsamples, availdata); tocopy > 0; tocopy--)
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{
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float f = *block++;
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*dest++ = f;
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*dest++ = f;
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len -= 8;
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ptr++;
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}
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xad.committed = ptr;
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}
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else
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{
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size_t availdata = (kBufSize - ptr) * 4;
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size_t tocopy = std::min(availdata, len);
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memcpy(dest, block, tocopy);
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dest += tocopy / 4;
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len -= tocopy;
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xad.committed += tocopy / 4;
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}
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}
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if (xad.finished)
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{
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memset(dest, 0, len);
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return true;
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}
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if (len > 0)
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{
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// we ran out of data and need more
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getNextXABlock(&xad, m_Looping);
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// repeat until done.
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}
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else break;
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}
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return !xad.finished;
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}
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//==========================================================================
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//
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// XA_OpenSong
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//
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//==========================================================================
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StreamSource *XA_OpenSong(MusicIO::FileInterface *reader)
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{
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return new XASong(reader);
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}
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