zmusic/source/streamsources/music_xa.cpp
Christoph Oelckers a4eae42ec5 - initial commit
2020-01-02 17:52:30 +01:00

354 lines
8.1 KiB
C++

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