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avcodec: add Direct Stream Transfer (DST) decoder

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Signed-off-by: Paul B Mahol <onemda@gmail.com>
This commit is contained in:
Peter Ross 2016-05-05 21:21:27 +02:00 committed by Paul B Mahol
parent 365b0c13e4
commit 86e493a6ff
11 changed files with 611 additions and 88 deletions
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9
libavcodec/Makefile
View File

@ -239,13 +239,14 @@ OBJS-$(CONFIG_DNXHD_DECODER) += dnxhddec.o dnxhddata.o
OBJS-$(CONFIG_DNXHD_ENCODER) += dnxhdenc.o dnxhddata.o
OBJS-$(CONFIG_DPX_DECODER) += dpx.o
OBJS-$(CONFIG_DPX_ENCODER) += dpxenc.o
OBJS-$(CONFIG_DSD_LSBF_DECODER) += dsddec.o
OBJS-$(CONFIG_DSD_MSBF_DECODER) += dsddec.o
OBJS-$(CONFIG_DSD_LSBF_PLANAR_DECODER) += dsddec.o
OBJS-$(CONFIG_DSD_MSBF_PLANAR_DECODER) += dsddec.o
OBJS-$(CONFIG_DSD_LSBF_DECODER) += dsddec.o dsd.o
OBJS-$(CONFIG_DSD_MSBF_DECODER) += dsddec.o dsd.o
OBJS-$(CONFIG_DSD_LSBF_PLANAR_DECODER) += dsddec.o dsd.o
OBJS-$(CONFIG_DSD_MSBF_PLANAR_DECODER) += dsddec.o dsd.o
OBJS-$(CONFIG_DSICINAUDIO_DECODER) += dsicinaudio.o
OBJS-$(CONFIG_DSICINVIDEO_DECODER) += dsicinvideo.o
OBJS-$(CONFIG_DSS_SP_DECODER) += dss_sp.o
OBJS-$(CONFIG_DST_DECODER) += dstdec.o dsd.o
OBJS-$(CONFIG_DVBSUB_DECODER) += dvbsubdec.o
OBJS-$(CONFIG_DVBSUB_ENCODER) += dvbsub.o
OBJS-$(CONFIG_DVDSUB_DECODER) += dvdsubdec.o

1
libavcodec/allcodecs.c
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@ -395,6 +395,7 @@ void avcodec_register_all(void)
REGISTER_DECODER(DSD_MSBF_PLANAR, dsd_msbf_planar);
REGISTER_DECODER(DSICINAUDIO, dsicinaudio);
REGISTER_DECODER(DSS_SP, dss_sp);
REGISTER_DECODER(DST, dst);
REGISTER_ENCDEC (EAC3, eac3);
REGISTER_DECODER(EVRC, evrc);
REGISTER_DECODER(FFWAVESYNTH, ffwavesynth);

1
libavcodec/avcodec.h
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@ -591,6 +591,7 @@ enum AVCodecID {
AV_CODEC_ID_INTERPLAY_ACM,
AV_CODEC_ID_XMA1,
AV_CODEC_ID_XMA2,
AV_CODEC_ID_DST,
/* subtitle codecs */
AV_CODEC_ID_FIRST_SUBTITLE = 0x17000, ///< A dummy ID pointing at the start of subtitle codecs.

7
libavcodec/codec_desc.c
View File

@ -2683,6 +2683,13 @@ static const AVCodecDescriptor codec_descriptors[] = {
.long_name = NULL_IF_CONFIG_SMALL("Xbox Media Audio 2"),
.props = AV_CODEC_PROP_LOSSY,
},
{
.id = AV_CODEC_ID_DST,
.type = AVMEDIA_TYPE_AUDIO,
.name = "dst",
.long_name = NULL_IF_CONFIG_SMALL("DST (Direct Stream Transfer)"),
.props = AV_CODEC_PROP_LOSSLESS,
},
/* subtitle codecs */
{

86
libavcodec/dsd.c Normal file
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@ -0,0 +1,86 @@
/*
* Direct Stream Digital (DSD) decoder
* based on BSD licensed dsd2pcm by Sebastian Gesemann
* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
* Copyright (c) 2014 Peter Ross
*
* 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
*/
#include "libavcodec/internal.h"
#include "libavcodec/mathops.h"
#include "avcodec.h"
#include "dsd_tablegen.h"
#include "dsd.h"
static av_cold void dsd_ctables_tableinit(void)
{
int t, e, m, sign;
double acc[CTABLES];
for (e = 0; e < 256; ++e) {
memset(acc, 0, sizeof(acc));
for (m = 0; m < 8; ++m) {
sign = (((e >> (7 - m)) & 1) * 2 - 1);
for (t = 0; t < CTABLES; ++t)
acc[t] += sign * htaps[t * 8 + m];
}
for (t = 0; t < CTABLES; ++t)
ctables[CTABLES - 1 - t][e] = acc[t];
}
}
av_cold void ff_init_dsd_data(void)
{
static int done = 0;
if (done)
return;
dsd_ctables_tableinit();
done = 1;
}
void ff_dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
const unsigned char *src, ptrdiff_t src_stride,
float *dst, ptrdiff_t dst_stride)
{
unsigned pos, i;
unsigned char* p;
double sum;
pos = s->pos;
while (samples-- > 0) {
s->buf[pos] = lsbf ? ff_reverse[*src] : *src;
src += src_stride;
p = s->buf + ((pos - CTABLES) & FIFOMASK);
*p = ff_reverse[*p];
sum = 0.0;
for (i = 0; i < CTABLES; i++) {
unsigned char a = s->buf[(pos - i) & FIFOMASK];
unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK];
sum += ctables[i][a] + ctables[i][b];
}
*dst = (float)sum;
dst += dst_stride;
pos = (pos + 1) & FIFOMASK;
}
s->pos = pos;
}

53
libavcodec/dsd.h Normal file
View File

@ -0,0 +1,53 @@
/*
* Direct Stream Digital (DSD) decoder
* based on BSD licensed dsd2pcm by Sebastian Gesemann
* Copyright (c) 2009, 2011 Sebastian Gesemann. All rights reserved.
* Copyright (c) 2014 Peter Ross
*
* 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
*/
#ifndef AVCODEC_DSD_H
#define AVCODEC_DSD_H
#include "libavcodec/internal.h"
#include "libavcodec/mathops.h"
#include "avcodec.h"
#include "dsd_tablegen.h"
#define HTAPS 48 /** number of FIR constants */
#define FIFOSIZE 16 /** must be a power of two */
#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */
#if FIFOSIZE * 8 < HTAPS * 2
#error "FIFOSIZE too small"
#endif
/**
* Per-channel buffer
*/
typedef struct DSDContext {
unsigned char buf[FIFOSIZE];
unsigned pos;
} DSDContext;
void ff_init_dsd_data(void);
void ff_dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
const unsigned char *src, ptrdiff_t src_stride,
float *dst, ptrdiff_t dst_stride);
#endif /* AVCODEC_DSD_H */

18
libavcodec/dsd_tablegen.h
View File

@ -25,6 +25,7 @@
#include <stdint.h>
#include "libavutil/attributes.h"
#include "dsd.h"
#define HTAPS 48 /** number of FIR constants */
#define CTABLES ((HTAPS + 7) / 8) /** number of "8 MACs" lookup tables */
@ -71,21 +72,4 @@ static const double htaps[HTAPS] = {
};
static float ctables[CTABLES][256];
static av_cold void dsd_ctables_tableinit(void)
{
int t, e, m, sign;
double acc[CTABLES];
for (e = 0; e < 256; ++e) {
memset(acc, 0, sizeof(acc));
for (m = 0; m < 8; ++m) {
sign = (((e >> (7 - m)) & 1) * 2 - 1);
for (t = 0; t < CTABLES; ++t)
acc[t] += sign * htaps[t * 8 + m];
}
for (t = 0; t < CTABLES; ++t)
ctables[CTABLES - 1 - t][e] = acc[t];
}
}
#endif /* AVCODEC_DSD_TABLEGEN_H */

63
libavcodec/dsddec.c
View File

@ -29,71 +29,14 @@
#include "libavcodec/internal.h"
#include "libavcodec/mathops.h"
#include "avcodec.h"
#include "dsd_tablegen.h"
#define FIFOSIZE 16 /** must be a power of two */
#define FIFOMASK (FIFOSIZE - 1) /** bit mask for FIFO offsets */
#if FIFOSIZE * 8 < HTAPS * 2
#error "FIFOSIZE too small"
#endif
/**
* Per-channel buffer
*/
typedef struct {
unsigned char buf[FIFOSIZE];
unsigned pos;
} DSDContext;
static void dsd2pcm_translate(DSDContext* s, size_t samples, int lsbf,
const unsigned char *src, ptrdiff_t src_stride,
float *dst, ptrdiff_t dst_stride)
{
unsigned pos, i;
unsigned char* p;
double sum;
pos = s->pos;
while (samples-- > 0) {
s->buf[pos] = lsbf ? ff_reverse[*src] : *src;
src += src_stride;
p = s->buf + ((pos - CTABLES) & FIFOMASK);
*p = ff_reverse[*p];
sum = 0.0;
for (i = 0; i < CTABLES; i++) {
unsigned char a = s->buf[(pos - i) & FIFOMASK];
unsigned char b = s->buf[(pos - (CTABLES*2 - 1) + i) & FIFOMASK];
sum += ctables[i][a] + ctables[i][b];
}
*dst = (float)sum;
dst += dst_stride;
pos = (pos + 1) & FIFOMASK;
}
s->pos = pos;
}
static av_cold void init_static_data(void)
{
static int done = 0;
if (done)
return;
dsd_ctables_tableinit();
done = 1;
}
#include "dsd.h"
static av_cold int decode_init(AVCodecContext *avctx)
{
DSDContext * s;
int i;
init_static_data();
ff_init_dsd_data();
s = av_malloc_array(sizeof(DSDContext), avctx->channels);
if (!s)
@ -140,7 +83,7 @@ static int decode_frame(AVCodecContext *avctx, void *data,
for (i = 0; i < avctx->channels; i++) {
float * dst = ((float **)frame->extended_data)[i];
dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
ff_dsd2pcm_translate(&s[i], frame->nb_samples, lsbf,
avpkt->data + i * src_next, src_stride,
dst, 1);
}

374
libavcodec/dstdec.c Normal file
View File

@ -0,0 +1,374 @@
/*
* 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 * 8 / 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(64, 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;
}
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(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] * 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;
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 * avctx->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, avctx->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, avctx->channels)) < 0)
return ret;
}
/* Half Probability (10.10) */
for (ch = 0; ch < avctx->channels; ch++)
half_prob[ch] = get_bits1(gb);
/* Filter Coef Sets (10.12) */
read_table(gb, &s->fsets, fsets_code_pred_coeff, 7, 9, 1, 0);
/* Probability Tables (10.13) */
read_table(gb, &s->probs, probs_code_pred_coeff, 6, 7, 0, 1);
/* 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 * avctx->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];
const 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_WN64A(status + 8, (AV_RN64A(status + 8) << 1) | ((AV_RN64A(status) >> 63) & 1));
AV_WN64A(status, (AV_RN64A(status) << 1) | v);
}
}
dsd:
for (i = 0; i < avctx->channels; i++) {
ff_dsd2pcm_translate(&s->dsdctx[i], frame->nb_samples, 0,
frame->data[0] + i * 4,
avctx->channels * 4, pcm + i, avctx->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 },
};

2
libavcodec/utils.c
View File

@ -3490,6 +3490,8 @@ static int get_audio_frame_duration(enum AVCodecID id, int sr, int ch, int ba,
/* calc from sample rate */
if (id == AV_CODEC_ID_TTA)
return 256 * sr / 245;
else if (id == AV_CODEC_ID_DST)
return 588 * sr / 44100;
if (ch > 0) {
/* calc from sample rate and channels */

85
libavformat/iff.c
View File

@ -59,6 +59,10 @@
#define ID_RGB8 MKTAG('R','G','B','8')
#define ID_RGBN MKTAG('R','G','B','N')
#define ID_DSD MKTAG('D','S','D',' ')
#define ID_DST MKTAG('D','S','T',' ')
#define ID_DSTC MKTAG('D','S','T','C')
#define ID_DSTF MKTAG('D','S','T','F')
#define ID_FRTE MKTAG('F','R','T','E')
#define ID_ANIM MKTAG('A','N','I','M')
#define ID_ANHD MKTAG('A','N','H','D')
#define ID_DLTA MKTAG('D','L','T','A')
@ -159,6 +163,7 @@ static int iff_probe(AVProbeData *p)
static const AVCodecTag dsd_codec_tags[] = {
{ AV_CODEC_ID_DSD_MSBF, ID_DSD },
{ AV_CODEC_ID_DST, ID_DST },
{ AV_CODEC_ID_NONE, 0 },
};
@ -287,7 +292,7 @@ static int parse_dsd_prop(AVFormatContext *s, AVStream *st, uint64_t eof)
case MKTAG('C','M','P','R'):
if (size < 4)
return AVERROR_INVALIDDATA;
tag = avio_rl32(pb);
st->codecpar->codec_tag = tag = avio_rl32(pb);
st->codecpar->codec_id = ff_codec_get_id(dsd_codec_tags, tag);
if (!st->codecpar->codec_id) {
av_log(s, AV_LOG_ERROR, "'%c%c%c%c' compression is not supported\n",
@ -338,6 +343,63 @@ static int parse_dsd_prop(AVFormatContext *s, AVStream *st, uint64_t eof)
return 0;
}
static int read_dst_frame(AVFormatContext *s, AVPacket *pkt)
{
IffDemuxContext *iff = s->priv_data;
AVIOContext *pb = s->pb;
uint32_t chunk_id;
uint64_t chunk_pos, data_pos, data_size;
int ret = AVERROR_EOF;
while (!avio_feof(pb)) {
chunk_pos = avio_tell(pb);
if (chunk_pos >= iff->body_end)
return AVERROR_EOF;
chunk_id = avio_rl32(pb);
data_size = iff->is_64bit ? avio_rb64(pb) : avio_rb32(pb);
data_pos = avio_tell(pb);
if (data_size < 1)
return AVERROR_INVALIDDATA;
switch (chunk_id) {
case ID_DSTF:
if (!pkt) {
iff->body_pos = avio_tell(pb) - (iff->is_64bit ? 12 : 8);
iff->body_size = iff->body_end - iff->body_pos;
return 0;
}
ret = av_get_packet(pb, pkt, data_size);
if (ret < 0)
return ret;
if (data_size & 1)
avio_skip(pb, 1);
pkt->flags |= AV_PKT_FLAG_KEY;
pkt->stream_index = 0;
pkt->duration = 588 * s->streams[0]->codecpar->sample_rate / 44100;
pkt->pos = chunk_pos;
chunk_pos = avio_tell(pb);
if (chunk_pos >= iff->body_end)
return 0;
avio_seek(pb, chunk_pos, SEEK_SET);
return 0;
case ID_FRTE:
if (data_size < 4)
return AVERROR_INVALIDDATA;
s->streams[0]->duration = avio_rb32(pb) * 588LL * s->streams[0]->codecpar->sample_rate / 44100;
break;
}
avio_skip(pb, data_size - (avio_tell(pb) - data_pos) + (data_size & 1));
}
return ret;
}
static const uint8_t deep_rgb24[] = {0, 0, 0, 3, 0, 1, 0, 8, 0, 2, 0, 8, 0, 3, 0, 8};
static const uint8_t deep_rgba[] = {0, 0, 0, 4, 0, 1, 0, 8, 0, 2, 0, 8, 0, 3, 0, 8};
static const uint8_t deep_bgra[] = {0, 0, 0, 4, 0, 3, 0, 8, 0, 2, 0, 8, 0, 1, 0, 8};
@ -425,10 +487,16 @@ static int iff_read_header(AVFormatContext *s)
case ID_BODY:
case ID_DBOD:
case ID_DSD:
case ID_DST:
case ID_MDAT:
iff->body_pos = avio_tell(pb);
iff->body_end = iff->body_pos + data_size;
iff->body_size = data_size;
if (chunk_id == ID_DST) {
int ret = read_dst_frame(s, NULL);
if (ret < 0)
return ret;
}
break;
case ID_CHAN:
@ -654,7 +722,8 @@ static int iff_read_header(AVFormatContext *s)
avpriv_request_sample(s, "compression %d and bit depth %d", iff->maud_compression, iff->maud_bits);
return AVERROR_PATCHWELCOME;
}
} else if (st->codecpar->codec_tag != ID_DSD) {
} else if (st->codecpar->codec_tag != ID_DSD &&
st->codecpar->codec_tag != ID_DST) {
switch (iff->svx8_compression) {
case COMP_NONE:
st->codecpar->codec_id = AV_CODEC_ID_PCM_S8_PLANAR;
@ -675,6 +744,8 @@ static int iff_read_header(AVFormatContext *s)
st->codecpar->bits_per_coded_sample = av_get_bits_per_sample(st->codecpar->codec_id);
st->codecpar->bit_rate = st->codecpar->channels * st->codecpar->sample_rate * st->codecpar->bits_per_coded_sample;
st->codecpar->block_align = st->codecpar->channels * st->codecpar->bits_per_coded_sample;
if (st->codecpar->codec_tag == ID_DSD && st->codecpar->block_align <= 0)
return AVERROR_INVALIDDATA;
break;
case AVMEDIA_TYPE_VIDEO:
@ -745,16 +816,16 @@ static int iff_read_packet(AVFormatContext *s,
int ret;
int64_t pos = avio_tell(pb);
if (st->codecpar->codec_tag == ID_ANIM) {
if (avio_feof(pb))
return AVERROR_EOF;
} else if (pos >= iff->body_end) {
if (avio_feof(pb))
return AVERROR_EOF;
if (st->codecpar->codec_tag != ID_ANIM && pos >= iff->body_end)
return AVERROR_EOF;
}
if (st->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) {
if (st->codecpar->codec_tag == ID_DSD || st->codecpar->codec_tag == ID_MAUD) {
ret = av_get_packet(pb, pkt, FFMIN(iff->body_end - pos, 1024 * st->codecpar->block_align));
} else if (st->codecpar->codec_tag == ID_DST) {
return read_dst_frame(s, pkt);
} else {
if (iff->body_size > INT_MAX)
return AVERROR_INVALIDDATA;