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XA code cleanup
Decoding is still the same (byte-exact)
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bnnm
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2c6091fab3
commit
572cd6a621
@ -1,70 +1,97 @@
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#include "coding.h"
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#include "../util.h"
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const int SH = 4;
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const int SHC = 10;
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// todo this is based on Kazzuya's old code; different emus (PCSX, Mame, Mednafen, etc) do
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// XA coefs int math in different ways (see comments below), so may not be 100% accurate.
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// May be implemented like the SNES/SPC700 BRR (per-filter code?).
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double K0[4] = { 0.0, 0.9375, 1.796875, 1.53125};
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double K1[4] = { 0.0, 0.0, -0.8125,-0.859375};
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static int IK0(int fid)
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{ return ((int)((-K0[fid]) * (1 << SHC))); }
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static int IK1(int fid)
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{ return ((int)((-K1[fid]) * (1 << SHC))); }
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static int CLAMP(int value, int Minim, int Maxim)
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{
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if (value < Minim) value = Minim;
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if (value > Maxim) value = Maxim;
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return value;
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}
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/* XA ADPCM gain values */
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static const double K0[4] = { 0.0, 0.9375, 1.796875, 1.53125 };
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static const double K1[4] = { 0.0, 0.0, -0.8125,-0.859375};
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static int IK0(int fid) { return ((int)((-K0[fid]) * (1 << 10))); } /* K0/1 floats to int, K*2^10 = K*1024 */
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static int IK1(int fid) { return ((int)((-K1[fid]) * (1 << 10))); }
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/* Sony XA ADPCM, defined for CD-DA/CD-i in the "Red Book" (private) or "Green Book" (public) specs.
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* The algorithm basically is BRR (Bit Rate Reduction) from the SNES SPC700, while the data layout is new.
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*
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* Decoding is defined in diagrams, roughly as:
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* pcm = clamp( signed_nibble * 2^(12-range) + K0[index]*hist1 + K1[index]*hist2 )
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* - Range (12-range=shift) and filter index are renewed every ~28 samples.
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* - nibble is expanded to a signed 16b sample, reimplemented as:
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* short sample = ((nibble << 12) & 0xf000) >> shift
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* or: int sample = ((nibble << 28) & 0xf0000000) >> (shift + N)
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* - K0/K1 are float coefs are typically redefined with int math in various ways, with non-equivalent rounding:
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* (sample + K0*2^N*hist1 + K1*2^N*hist1 + [(2^N)/2]) / 2^N
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* (sample + K0*2^N*hist1 + K1*2^N*hist1 + [(2^N)/2]) >> N
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* sample + (K0<<N*hist1 + K1<<N*hist1)>>N
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* sample + (K0*2^N*hist1)>>N + (K1*2^N*hist1)>>N
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* etc
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* (rounding differences should be inaudible, so public implementations may be approximations)
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*
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* Various XA descendants (PS-ADPCM, EA-XA, NGC DTK, FADPCM, etc) do filters/rounding slightly
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* differently, using one of the above methods.
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* int coef tables commonly use N = 6 or 8, so K0 0.9375*64 = 60 or 0.9375*256 = 240
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* PS1 XA is apparently upsampled and interpolated to 44100, vgmstream doesn't simulate this.
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*
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* Info (Green Book): https://www.lscdweb.com/data/downloadables/2/8/cdi_may94_r2.pdf
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*/
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void decode_xa(VGMSTREAM * vgmstream, sample * outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int channel) {
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static int head_table[8] = {0,2,8,10};
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VGMSTREAMCHANNEL * stream = &(vgmstream->ch[channel]);
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int predict_nr, shift_factor, sample;
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int32_t hist1=stream->adpcm_history1_32;
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int32_t hist2=stream->adpcm_history2_32;
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int HeadTable[8]={0,2,8,10};
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off_t sp_offset;
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int i;
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int frames_in, sample_count = 0;
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int32_t coef1, coef2, coef_index, shift_factor;
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int32_t hist1 = stream->adpcm_history1_32;
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int32_t hist2 = stream->adpcm_history2_32;
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short scale;
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int i;
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int32_t sample_count;
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/* external interleave (fixed size), mono/stereo */
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frames_in = first_sample / (28*2 / channelspacing);
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first_sample = first_sample % 28;
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int framesin = first_sample / (56 / channelspacing);
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/* hack for mono/stereo handling */
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vgmstream->xa_get_high_nibble = !vgmstream->xa_get_high_nibble;
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if (first_sample && channelspacing==1)
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vgmstream->xa_get_high_nibble = !vgmstream->xa_get_high_nibble;
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/* parse current sound unit (subframe) sound parameters */
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sp_offset = stream->offset+head_table[frames_in]+vgmstream->xa_get_high_nibble;
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coef_index = (read_8bit(sp_offset,stream->streamfile) >> 4) & 0xf;
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shift_factor = (read_8bit(sp_offset,stream->streamfile) ) & 0xf;
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VGM_ASSERT(coef_index > 4 || shift_factor > 12, "XA: incorrect coefs/shift at %lx\n", sp_offset);
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if (coef_index > 4)
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coef_index = 0; /* only 4 filters are used, rest is apparently 0 */
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if (shift_factor > 12)
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shift_factor = 9; /* supposedly, from Nocash PSX docs */
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coef1 = IK0(coef_index);
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coef2 = IK1(coef_index);
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first_sample = first_sample % 28;
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vgmstream->xa_get_high_nibble=!vgmstream->xa_get_high_nibble;
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/* decode nibbles */
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for (i = first_sample; i < first_sample + samples_to_do; i++) {
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int32_t new_sample;
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uint8_t nibbles = (uint8_t)read_8bit(stream->offset+0x10+frames_in+(i*0x04),stream->streamfile);
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if((first_sample) && (channelspacing==1))
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vgmstream->xa_get_high_nibble=!vgmstream->xa_get_high_nibble;
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new_sample = vgmstream->xa_get_high_nibble ?
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(nibbles >> 4) & 0x0f :
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(nibbles ) & 0x0f;
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new_sample = (int16_t)((new_sample << 12) & 0xf000) >> shift_factor; /* 16b sign extend + scale */
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new_sample = new_sample << 4;
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new_sample = new_sample - ((coef1*hist1 + coef2*hist2) >> 10);
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predict_nr = read_8bit(stream->offset+HeadTable[framesin]+vgmstream->xa_get_high_nibble,stream->streamfile) >> 4;
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shift_factor = read_8bit(stream->offset+HeadTable[framesin]+vgmstream->xa_get_high_nibble,stream->streamfile) & 0xf;
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hist2 = hist1;
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hist1 = new_sample; /* must go before clamp, somehow */
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new_sample = new_sample >> 4;
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new_sample = clamp16(new_sample);
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for (i=first_sample,sample_count=0; i<first_sample+samples_to_do; i++,sample_count+=channelspacing) {
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short sample_byte = (short)read_8bit(stream->offset+16+framesin+(i*4),stream->streamfile);
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outbuf[sample_count] = new_sample;
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sample_count += channelspacing;
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}
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scale = ((vgmstream->xa_get_high_nibble ?
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sample_byte >> 4 :
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sample_byte & 0x0f)<<12);
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sample = (short)(scale & 0xf000) >> shift_factor;
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sample <<= SH;
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sample -= (IK0(predict_nr) * hist1 + (IK1(predict_nr) * hist2)) >> SHC;
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hist2=hist1;
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hist1=sample;
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sample = CLAMP(sample, -32768 << SH, 32767 << SH);
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outbuf[sample_count] = (short)(sample >> SH);
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}
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stream->adpcm_history1_32=hist1;
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stream->adpcm_history2_32=hist2;
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stream->adpcm_history1_32 = hist1;
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stream->adpcm_history2_32 = hist2;
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}
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size_t xa_bytes_to_samples(size_t bytes, int channels, int is_blocked) {
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