diff options
Diffstat (limited to 'libavcodec')
| -rw-r--r-- | libavcodec/Makefile | 9 | ||||
| -rw-r--r-- | libavcodec/allcodecs.c | 1 | ||||
| -rw-r--r-- | libavcodec/avcodec.h | 1 | ||||
| -rw-r--r-- | libavcodec/codec_desc.c | 7 | ||||
| -rw-r--r-- | libavcodec/dsd.c | 86 | ||||
| -rw-r--r-- | libavcodec/dsd.h | 53 | ||||
| -rw-r--r-- | libavcodec/dsd_tablegen.h | 18 | ||||
| -rw-r--r-- | libavcodec/dsddec.c | 63 | ||||
| -rw-r--r-- | libavcodec/dstdec.c | 374 | ||||
| -rw-r--r-- | libavcodec/utils.c | 2 |
10 files changed, 533 insertions, 81 deletions
diff --git a/libavcodec/Makefile b/libavcodec/Makefile index 998477c161..3f0ffd11b4 100644 --- a/libavcodec/Makefile +++ b/libavcodec/Makefile @@ -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 diff --git a/libavcodec/allcodecs.c b/libavcodec/allcodecs.c index d435136c84..44ebafd321 100644 --- a/libavcodec/allcodecs.c +++ b/libavcodec/allcodecs.c @@ -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); diff --git a/libavcodec/avcodec.h b/libavcodec/avcodec.h index 97b21281f8..9ec9adfb6a 100644 --- a/libavcodec/avcodec.h +++ b/libavcodec/avcodec.h @@ -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. diff --git a/libavcodec/codec_desc.c b/libavcodec/codec_desc.c index 57bd4ba5de..23d591193a 100644 --- a/libavcodec/codec_desc.c +++ b/libavcodec/codec_desc.c @@ -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 */ { diff --git a/libavcodec/dsd.c b/libavcodec/dsd.c new file mode 100644 index 0000000000..9104f38476 --- /dev/null +++ b/libavcodec/dsd.c @@ -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; +} diff --git a/libavcodec/dsd.h b/libavcodec/dsd.h new file mode 100644 index 0000000000..42a9a4872e --- /dev/null +++ b/libavcodec/dsd.h @@ -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 */ diff --git a/libavcodec/dsd_tablegen.h b/libavcodec/dsd_tablegen.h index 990d57a5cb..e5da86a1dc 100644 --- a/libavcodec/dsd_tablegen.h +++ b/libavcodec/dsd_tablegen.h @@ -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 */ diff --git a/libavcodec/dsddec.c b/libavcodec/dsddec.c index f1dfd4b231..880d69191c 100644 --- a/libavcodec/dsddec.c +++ b/libavcodec/dsddec.c @@ -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); } diff --git a/libavcodec/dstdec.c b/libavcodec/dstdec.c new file mode 100644 index 0000000000..13be24a057 --- /dev/null +++ b/libavcodec/dstdec.c @@ -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 }, +}; diff --git a/libavcodec/utils.c b/libavcodec/utils.c index 8652b17ee7..e5a832b3b4 100644 --- a/libavcodec/utils.c +++ b/libavcodec/utils.c @@ -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 */ |