early-access version 1432
This commit is contained in:
pineappleEA
392
externals/ffmpeg/libavcodec/dstdec.c
vendored
Executable file
392
externals/ffmpeg/libavcodec/dstdec.c
vendored
Executable file
@@ -0,0 +1,392 @@
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/*
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* Direct Stream Transfer (DST) decoder
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* Copyright (c) 2014 Peter Ross <pross@xvid.org>
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*
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* This file is part of FFmpeg.
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*
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* FFmpeg is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Lesser General Public
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* License as published by the Free Software Foundation; either
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* version 2.1 of the License, or (at your option) any later version.
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*
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* FFmpeg is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Lesser General Public License for more details.
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*
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* You should have received a copy of the GNU Lesser General Public
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* License along with FFmpeg; if not, write to the Free Software
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* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
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*/
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/**
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* @file
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* Direct Stream Transfer (DST) decoder
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* ISO/IEC 14496-3 Part 3 Subpart 10: Technical description of lossless coding of oversampled audio
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*/
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#include "libavutil/avassert.h"
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#include "libavutil/intreadwrite.h"
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#include "internal.h"
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#include "get_bits.h"
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#include "avcodec.h"
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#include "golomb.h"
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#include "mathops.h"
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#include "dsd.h"
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#define DST_MAX_CHANNELS 6
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#define DST_MAX_ELEMENTS (2 * DST_MAX_CHANNELS)
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#define DSD_FS44(sample_rate) (sample_rate * 8LL / 44100)
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#define DST_SAMPLES_PER_FRAME(sample_rate) (588 * DSD_FS44(sample_rate))
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static const int8_t fsets_code_pred_coeff[3][3] = {
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{ -8 },
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{ -16, 8 },
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{ -9, -5, 6 },
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};
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static const int8_t probs_code_pred_coeff[3][3] = {
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{ -8 },
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{ -16, 8 },
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{ -24, 24, -8 },
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};
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typedef struct ArithCoder {
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unsigned int a;
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unsigned int c;
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} ArithCoder;
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typedef struct Table {
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unsigned int elements;
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unsigned int length[DST_MAX_ELEMENTS];
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int coeff[DST_MAX_ELEMENTS][128];
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} Table;
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typedef struct DSTContext {
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AVClass *class;
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GetBitContext gb;
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ArithCoder ac;
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Table fsets, probs;
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DECLARE_ALIGNED(16, uint8_t, status)[DST_MAX_CHANNELS][16];
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DECLARE_ALIGNED(16, int16_t, filter)[DST_MAX_ELEMENTS][16][256];
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DSDContext dsdctx[DST_MAX_CHANNELS];
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} DSTContext;
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static av_cold int decode_init(AVCodecContext *avctx)
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{
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DSTContext *s = avctx->priv_data;
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int i;
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if (avctx->channels > DST_MAX_CHANNELS) {
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avpriv_request_sample(avctx, "Channel count %d", avctx->channels);
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return AVERROR_PATCHWELCOME;
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}
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// the sample rate is only allowed to be 64,128,256 * 44100 by ISO/IEC 14496-3:2005(E)
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// We are a bit more tolerant here, but this check is needed to bound the size and duration
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if (avctx->sample_rate > 512 * 44100)
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return AVERROR_INVALIDDATA;
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if (DST_SAMPLES_PER_FRAME(avctx->sample_rate) & 7) {
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return AVERROR_PATCHWELCOME;
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}
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avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
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for (i = 0; i < avctx->channels; i++)
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memset(s->dsdctx[i].buf, 0x69, sizeof(s->dsdctx[i].buf));
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ff_init_dsd_data();
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return 0;
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}
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static int read_map(GetBitContext *gb, Table *t, unsigned int map[DST_MAX_CHANNELS], int channels)
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{
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int ch;
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t->elements = 1;
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map[0] = 0;
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if (!get_bits1(gb)) {
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for (ch = 1; ch < channels; ch++) {
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int bits = av_log2(t->elements) + 1;
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map[ch] = get_bits(gb, bits);
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if (map[ch] == t->elements) {
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t->elements++;
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if (t->elements >= DST_MAX_ELEMENTS)
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return AVERROR_INVALIDDATA;
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} else if (map[ch] > t->elements) {
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return AVERROR_INVALIDDATA;
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}
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}
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} else {
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memset(map, 0, sizeof(*map) * DST_MAX_CHANNELS);
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}
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return 0;
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}
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static av_always_inline int get_sr_golomb_dst(GetBitContext *gb, unsigned int k)
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{
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int v = get_ur_golomb_jpegls(gb, k, get_bits_left(gb), 0);
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if (v && get_bits1(gb))
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v = -v;
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return v;
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}
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static void read_uncoded_coeff(GetBitContext *gb, int *dst, unsigned int elements,
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int coeff_bits, int is_signed, int offset)
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{
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int i;
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for (i = 0; i < elements; i++) {
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dst[i] = (is_signed ? get_sbits(gb, coeff_bits) : get_bits(gb, coeff_bits)) + offset;
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}
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}
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static int read_table(GetBitContext *gb, Table *t, const int8_t code_pred_coeff[3][3],
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int length_bits, int coeff_bits, int is_signed, int offset)
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{
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unsigned int i, j, k;
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for (i = 0; i < t->elements; i++) {
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t->length[i] = get_bits(gb, length_bits) + 1;
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if (!get_bits1(gb)) {
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read_uncoded_coeff(gb, t->coeff[i], t->length[i], coeff_bits, is_signed, offset);
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} else {
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int method = get_bits(gb, 2), lsb_size;
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if (method == 3)
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return AVERROR_INVALIDDATA;
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read_uncoded_coeff(gb, t->coeff[i], method + 1, coeff_bits, is_signed, offset);
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lsb_size = get_bits(gb, 3);
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for (j = method + 1; j < t->length[i]; j++) {
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int c, x = 0;
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for (k = 0; k < method + 1; k++)
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x += code_pred_coeff[method][k] * (unsigned)t->coeff[i][j - k - 1];
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c = get_sr_golomb_dst(gb, lsb_size);
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if (x >= 0)
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c -= (x + 4) / 8;
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else
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c += (-x + 3) / 8;
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if (!is_signed) {
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if (c < offset || c >= offset + (1<<coeff_bits))
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return AVERROR_INVALIDDATA;
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}
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t->coeff[i][j] = c;
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}
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}
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}
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return 0;
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}
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static void ac_init(ArithCoder *ac, GetBitContext *gb)
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{
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ac->a = 4095;
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ac->c = get_bits(gb, 12);
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}
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static av_always_inline void ac_get(ArithCoder *ac, GetBitContext *gb, int p, int *e)
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{
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unsigned int k = (ac->a >> 8) | ((ac->a >> 7) & 1);
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unsigned int q = k * p;
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unsigned int a_q = ac->a - q;
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*e = ac->c < a_q;
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if (*e) {
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ac->a = a_q;
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} else {
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ac->a = q;
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ac->c -= a_q;
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}
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if (ac->a < 2048) {
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int n = 11 - av_log2(ac->a);
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ac->a <<= n;
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ac->c = (ac->c << n) | get_bits(gb, n);
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}
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}
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static uint8_t prob_dst_x_bit(int c)
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{
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return (ff_reverse[c & 127] >> 1) + 1;
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}
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static void build_filter(int16_t table[DST_MAX_ELEMENTS][16][256], const Table *fsets)
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{
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int i, j, k, l;
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for (i = 0; i < fsets->elements; i++) {
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int length = fsets->length[i];
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for (j = 0; j < 16; j++) {
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int total = av_clip(length - j * 8, 0, 8);
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for (k = 0; k < 256; k++) {
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int v = 0;
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for (l = 0; l < total; l++)
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v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l];
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table[i][j][k] = v;
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}
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}
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}
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}
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static int decode_frame(AVCodecContext *avctx, void *data,
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int *got_frame_ptr, AVPacket *avpkt)
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{
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unsigned samples_per_frame = DST_SAMPLES_PER_FRAME(avctx->sample_rate);
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unsigned map_ch_to_felem[DST_MAX_CHANNELS];
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unsigned map_ch_to_pelem[DST_MAX_CHANNELS];
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unsigned i, ch, same_map, dst_x_bit;
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unsigned half_prob[DST_MAX_CHANNELS];
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const int channels = avctx->channels;
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DSTContext *s = avctx->priv_data;
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GetBitContext *gb = &s->gb;
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ArithCoder *ac = &s->ac;
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AVFrame *frame = data;
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uint8_t *dsd;
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float *pcm;
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int ret;
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if (avpkt->size <= 1)
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return AVERROR_INVALIDDATA;
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frame->nb_samples = samples_per_frame / 8;
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if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
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return ret;
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dsd = frame->data[0];
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pcm = (float *)frame->data[0];
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if ((ret = init_get_bits8(gb, avpkt->data, avpkt->size)) < 0)
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return ret;
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if (!get_bits1(gb)) {
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skip_bits1(gb);
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if (get_bits(gb, 6))
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return AVERROR_INVALIDDATA;
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memcpy(frame->data[0], avpkt->data + 1, FFMIN(avpkt->size - 1, frame->nb_samples * channels));
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goto dsd;
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}
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/* Segmentation (10.4, 10.5, 10.6) */
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if (!get_bits1(gb)) {
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avpriv_request_sample(avctx, "Not Same Segmentation");
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return AVERROR_PATCHWELCOME;
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}
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if (!get_bits1(gb)) {
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avpriv_request_sample(avctx, "Not Same Segmentation For All Channels");
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return AVERROR_PATCHWELCOME;
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}
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if (!get_bits1(gb)) {
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avpriv_request_sample(avctx, "Not End Of Channel Segmentation");
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return AVERROR_PATCHWELCOME;
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}
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/* Mapping (10.7, 10.8, 10.9) */
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same_map = get_bits1(gb);
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if ((ret = read_map(gb, &s->fsets, map_ch_to_felem, channels)) < 0)
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return ret;
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if (same_map) {
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s->probs.elements = s->fsets.elements;
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memcpy(map_ch_to_pelem, map_ch_to_felem, sizeof(map_ch_to_felem));
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} else {
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avpriv_request_sample(avctx, "Not Same Mapping");
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if ((ret = read_map(gb, &s->probs, map_ch_to_pelem, channels)) < 0)
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return ret;
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}
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/* Half Probability (10.10) */
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for (ch = 0; ch < channels; ch++)
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half_prob[ch] = get_bits1(gb);
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/* Filter Coef Sets (10.12) */
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ret = read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
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if (ret < 0)
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return ret;
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/* Probability Tables (10.13) */
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ret = read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
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if (ret < 0)
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return ret;
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/* Arithmetic Coded Data (10.11) */
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if (get_bits1(gb))
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return AVERROR_INVALIDDATA;
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ac_init(ac, gb);
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build_filter(s->filter, &s->fsets);
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memset(s->status, 0xAA, sizeof(s->status));
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memset(dsd, 0, frame->nb_samples * 4 * channels);
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ac_get(ac, gb, prob_dst_x_bit(s->fsets.coeff[0][0]), &dst_x_bit);
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for (i = 0; i < samples_per_frame; i++) {
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for (ch = 0; ch < channels; ch++) {
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const unsigned felem = map_ch_to_felem[ch];
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int16_t (*filter)[256] = s->filter[felem];
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uint8_t *status = s->status[ch];
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int prob, residual, v;
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#define F(x) filter[(x)][status[(x)]]
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const int16_t predict = F( 0) + F( 1) + F( 2) + F( 3) +
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F( 4) + F( 5) + F( 6) + F( 7) +
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F( 8) + F( 9) + F(10) + F(11) +
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F(12) + F(13) + F(14) + F(15);
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#undef F
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if (!half_prob[ch] || i >= s->fsets.length[felem]) {
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unsigned pelem = map_ch_to_pelem[ch];
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unsigned index = FFABS(predict) >> 3;
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prob = s->probs.coeff[pelem][FFMIN(index, s->probs.length[pelem] - 1)];
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} else {
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prob = 128;
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}
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ac_get(ac, gb, prob, &residual);
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v = ((predict >> 15) ^ residual) & 1;
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dsd[((i >> 3) * channels + ch) << 2] |= v << (7 - (i & 0x7 ));
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AV_WL64A(status + 8, (AV_RL64A(status + 8) << 1) | ((AV_RL64A(status) >> 63) & 1));
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AV_WL64A(status, (AV_RL64A(status) << 1) | v);
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}
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}
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dsd:
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for (i = 0; i < channels; i++) {
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ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
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frame->data[0] + i * 4,
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channels * 4, pcm + i, channels);
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}
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*got_frame_ptr = 1;
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return avpkt->size;
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}
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AVCodec ff_dst_decoder = {
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.name = "dst",
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.long_name = NULL_IF_CONFIG_SMALL("DST (Digital Stream Transfer)"),
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.type = AVMEDIA_TYPE_AUDIO,
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.id = AV_CODEC_ID_DST,
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.priv_data_size = sizeof(DSTContext),
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.init = decode_init,
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.decode = decode_frame,
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.capabilities = AV_CODEC_CAP_DR1,
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.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT,
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AV_SAMPLE_FMT_NONE },
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};
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