#include "coding.h" #include "../util.h" /* Decodes Silicon Graphics' N64 VADPCM, big brother of GC ADPCM. * Has external coefs like DSP, but allows tables up to 8 groups of 8 coefs (code book) and also * up to 8 history samples (order). In practice order must be 2, while files use 2~4 tables, so it * ends up being an overcomplex XA. Code respects this quirky configurable hist for doc purposes though. * * This code is based on N64SoundListTool decoding (by Ice Mario), with bits of the official SDK vadpcm tool * decompilation. I can't get proper sound from the later though, so not too sure about accuracy of this * implementation. Output sounds correct though. */ void decode_vadpcm(VGMSTREAMCHANNEL* stream, sample_t* outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int order) { uint8_t frame[0x09] = {0}; off_t frame_offset; int frames_in, sample_count = 0; size_t bytes_per_frame, samples_per_frame; int i, j, k, o; int scale, index; int codes[16]; /* AKA ix */ int16_t hist[8] = {0}; int16_t out[16]; int16_t* coefs; VGM_ASSERT_ONCE(order != 2, "VADPCM: wrong order=%i\n", order); if (order != 2) /* only 2 allowed "in the current implementation" */ order = 2; /* up to 8 (hist[0]=oldest) but only uses latest 2 (order=2), so we don't save the whole thing ATM */ hist[6] = stream->adpcm_history2_16; hist[7] = stream->adpcm_history1_16; /* external interleave (fixed size), mono */ bytes_per_frame = 0x09; samples_per_frame = (bytes_per_frame - 0x01) * 2; /* always 16 */ frames_in = first_sample / samples_per_frame; first_sample = first_sample % samples_per_frame; /* parse frame header */ frame_offset = stream->offset + bytes_per_frame * frames_in; read_streamfile(frame, frame_offset, bytes_per_frame, stream->streamfile); /* ignore EOF errors */ scale = (frame[0] >> 4) & 0xF; index = (frame[0] >> 0) & 0xF; scale = 1 << scale; VGM_ASSERT_ONCE(index > 8, "DSP: incorrect index at %x\n", (uint32_t)frame_offset); if (index > 8) /* assumed */ index = 8; coefs = &stream->vadpcm_coefs[index * (order*8) + 0]; /* read and pre-scale all nibbles, since groups of 8 are needed */ for (i = 0, j = 0; i < 16; i += 2, j++) { int n0 = (frame[j+1] >> 4) & 0xF; int n1 = (frame[j+1] >> 0) & 0xF; /* sign extend */ if (n0 & 8) n0 = n0 - 16; if (n1 & 8) n1 = n1 - 16; codes[i+0] = n0 * scale; codes[i+1] = n1 * scale; } /* decode 2 sub-frames of 8 samples (maybe like this since N64 MIPS has registers to spare?) */ for (j = 0; j < 2; j++) { /* SDK dec code and N64ST copy 8 codes to a tmp buf and has some complex logic to move out buf * around, but since that's useless N64 asm would be much more optimized... hopefully */ int* sf_codes = &codes[j*8]; int16_t* sf_out = &out[j*8]; /* works with 8 samples at a time, related in twisted ways */ for( i = 0; i < 8; i++) { int sample, delta = 0; /* in practice: delta = coefs[0][i] * hist[6] + coefs[1][i] * hist[7], * much like XA's coef1*hist1 + coef2*hist2 but with multi coefs */ for (o = 0; o < order; o++) { delta += coefs[o*8 + i] * hist[(8 - order) + o]; } /* adds all previous samples */ for (k = i-1; k > -1; k--) { for (o = 1; o < order; o++) { /* assumed, since only goes coefs[1][k] */ delta += sf_codes[(i-1) - k] * coefs[(o*8) + k]; } } /* scale-filter thing (also seen in DSP) */ sample = (sf_codes[i] << 11); sample = (sample + delta) >> 11; if (sample > 32767) sample = 32767; else if (sample < -32768) sample = -32768; sf_out[i] = sample; } /* save subframe hist */ for (i = 8 - order; i < 8; i++) { hist[i] = sf_out[i]; } } /* copy samples last, since the whole thing is kinda complex to worry about half copying and stuff */ for (i = first_sample; i < first_sample + samples_to_do; i++) { outbuf[sample_count] = out[i]; sample_count += channelspacing; } /* update hist once all frame is actually copied */ if (first_sample + sample_count / channelspacing == samples_per_frame) { stream->adpcm_history2_16 = hist[6]; stream->adpcm_history1_16 = hist[7]; } } /* int32_t vadpcm_bytes_to_samples(size_t bytes, int channels) { if (channels <= 0) return 0; return bytes / channels / 0x09 * 16; } */ /* Reads code book, linearly unlike original SDK, that does some strange reordering and pre-scaling * to reduce some loops. Format is 8 coefs per 'order' per 'entries' (max 8, but order is always 2). So: * - i: table index (selectable filter tables on every decoded frame) * - j: order index (coefs for prev N hist samples) * - k: coef index (multiplication coefficient for 8 samples in a sub-frame) * coefs[i * (order*8) + j * 8 + k * order] = coefs[i][j][k] */ void vadpcm_read_coefs_be(VGMSTREAM* vgmstream, STREAMFILE* sf, off_t offset, int order, int entries, int ch) { int i; if (entries > 8) entries = 8; VGM_ASSERT(order != 2, "VADPCM: wrong order %i found\n", order); if (order != 2) order = 2; /* assumes all channels use same coefs, never seen non-mono files */ for (i = 0; i < entries * order * 8; i++) { vgmstream->ch[ch].vadpcm_coefs[i] = read_s16be(offset + i*2, sf); } vgmstream->codec_config = order; }