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ffmpeg/libavcodec/dstdec.c

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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 "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;
}
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 },
};
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