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FFmpeg/libavcodec/dstdec.c
Andreas Rheinhardt a247ac640d avcodec: Constify AVCodecs 
Given that the AVCodec.next pointer has now been removed, most of the
AVCodecs are not modified at all any more and can therefore be made
const (as this patch does); the only exceptions are the very few codecs
for external libraries that have a init_static_data callback.

Signed-off-by: Andreas Rheinhardt <andreas.rheinhardt@gmail.com>
Signed-off-by: James Almer <jamrial@gmail.com>
2021-04-27 10:43:15 -03:00

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