2020-05-01 16:04:23 +02:00
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#include "coding.h"
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#include "../util.h"
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/* Decodes Silicon Graphics' N64 VADPCM, big brother of GC ADPCM.
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* Has external coefs like DSP, but allows tables up to 8 groups of 8 coefs (code book) and also
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* up to 8 history samples (order). In practice order must be 2, while files use 2~4 tables, so it
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* ends up being an overcomplex XA. Code respects this quirky configurable hist for doc purposes though.
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*
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* This code is based on N64SoundListTool decoding (by Ice Mario), with bits of the official SDK vadpcm tool
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* decompilation. I can't get proper sound from the later though, so not too sure about accuracy of this
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* implementation. Output sounds correct though.
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*/
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void decode_vadpcm(VGMSTREAMCHANNEL* stream, sample_t* outbuf, int channelspacing, int32_t first_sample, int32_t samples_to_do, int order) {
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uint8_t frame[0x09] = {0};
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off_t frame_offset;
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int frames_in, sample_count = 0;
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size_t bytes_per_frame, samples_per_frame;
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int i, j, k, o;
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int scale, index;
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int codes[16]; /* AKA ix */
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int16_t hist[8] = {0};
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int16_t out[16];
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int16_t* coefs;
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VGM_ASSERT_ONCE(order != 2, "VADPCM: wrong order=%i\n", order);
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if (order != 2) /* only 2 allowed "in the current implementation" */
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order = 2;
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/* up to 8 (hist[0]=oldest) but only uses latest 2 (order=2), so we don't save the whole thing ATM */
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hist[6] = stream->adpcm_history2_16;
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hist[7] = stream->adpcm_history1_16;
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/* external interleave (fixed size), mono */
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bytes_per_frame = 0x09;
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samples_per_frame = (bytes_per_frame - 0x01) * 2; /* always 16 */
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frames_in = first_sample / samples_per_frame;
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first_sample = first_sample % samples_per_frame;
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/* parse frame header */
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frame_offset = stream->offset + bytes_per_frame * frames_in;
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read_streamfile(frame, frame_offset, bytes_per_frame, stream->streamfile); /* ignore EOF errors */
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scale = (frame[0] >> 4) & 0xF;
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index = (frame[0] >> 0) & 0xF;
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scale = 1 << scale;
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VGM_ASSERT_ONCE(index > 8, "DSP: incorrect index at %x\n", (uint32_t)frame_offset);
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if (index > 8) /* assumed */
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index = 8;
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coefs = &stream->vadpcm_coefs[index * (order*8) + 0];
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/* read and pre-scale all nibbles, since groups of 8 are needed */
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for (i = 0, j = 0; i < 16; i += 2, j++) {
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int n0 = (frame[j+1] >> 4) & 0xF;
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int n1 = (frame[j+1] >> 0) & 0xF;
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/* sign extend */
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if (n0 & 8)
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n0 = n0 - 16;
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if (n1 & 8)
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n1 = n1 - 16;
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codes[i+0] = n0 * scale;
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codes[i+1] = n1 * scale;
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}
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/* decode 2 sub-frames of 8 samples (maybe like this since N64 MIPS has registers to spare?) */
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for (j = 0; j < 2; j++) {
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/* SDK dec code and N64ST copy 8 codes to a tmp buf and has some complex logic to move out buf
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* around, but since that's useless N64 asm would be much more optimized... hopefully */
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int* sf_codes = &codes[j*8];
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int16_t* sf_out = &out[j*8];
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/* works with 8 samples at a time, related in twisted ways */
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for( i = 0; i < 8; i++) {
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int sample, delta = 0;
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/* in practice: delta = coefs[0][i] * hist[6] + coefs[1][i] * hist[7],
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* much like XA's coef1*hist1 + coef2*hist2 but with multi coefs */
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for (o = 0; o < order; o++) {
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delta += coefs[o*8 + i] * hist[(8 - order) + o];
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}
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/* adds all previous samples */
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for (k = i-1; k > -1; k--) {
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for (o = 1; o < order; o++) { /* assumed, since only goes coefs[1][k] */
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delta += sf_codes[(i-1) - k] * coefs[(o*8) + k];
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}
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}
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/* scale-filter thing (also seen in DSP) */
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sample = (sf_codes[i] << 11);
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sample = (sample + delta) >> 11;
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if (sample > 32767)
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sample = 32767;
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else if (sample < -32768)
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sample = -32768;
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sf_out[i] = sample;
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}
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/* save subframe hist */
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for (i = 8 - order; i < 8; i++) {
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hist[i] = sf_out[i];
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}
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}
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/* copy samples last, since the whole thing is kinda complex to worry about half copying and stuff */
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for (i = first_sample; i < first_sample + samples_to_do; i++) {
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outbuf[sample_count] = out[i];
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sample_count += channelspacing;
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}
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/* update hist once all frame is actually copied */
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2020-11-01 15:28:04 +01:00
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if (first_sample + sample_count / channelspacing == samples_per_frame) {
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2020-05-01 16:04:23 +02:00
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stream->adpcm_history2_16 = hist[6];
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stream->adpcm_history1_16 = hist[7];
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}
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}
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/*
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int32_t vadpcm_bytes_to_samples(size_t bytes, int channels) {
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if (channels <= 0) return 0;
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return bytes / channels / 0x09 * 16;
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}
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*/
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/* Reads code book, linearly unlike original SDK, that does some strange reordering and pre-scaling
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* to reduce some loops. Format is 8 coefs per 'order' per 'entries' (max 8, but order is always 2). So:
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* - i: table index (selectable filter tables on every decoded frame)
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* - j: order index (coefs for prev N hist samples)
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* - k: coef index (multiplication coefficient for 8 samples in a sub-frame)
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* coefs[i * (order*8) + j * 8 + k * order] = coefs[i][j][k] */
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void vadpcm_read_coefs_be(VGMSTREAM* vgmstream, STREAMFILE* sf, off_t offset, int order, int entries, int ch) {
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int i;
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if (entries > 8)
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entries = 8;
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VGM_ASSERT(order != 2, "VADPCM: wrong order %i found\n", order);
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if (order != 2)
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order = 2;
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/* assumes all channels use same coefs, never seen non-mono files */
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for (i = 0; i < entries * order * 8; i++) {
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vgmstream->ch[ch].vadpcm_coefs[i] = read_s16be(offset + i*2, sf);
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}
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vgmstream->codec_config = order;
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}
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