/* * 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/intreadwrite.h" #include "libavutil/mem_internal.h" #include "libavutil/reverse.h" #include "codec_internal.h" #include "decode.h" #include "get_bits.h" #include "avcodec.h" #include "golomb.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->ch_layout.nb_channels > DST_MAX_CHANNELS) { avpriv_request_sample(avctx, "Channel count %d", avctx->ch_layout.nb_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->ch_layout.nb_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 int 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++) { int64_t v = 0; for (l = 0; l < total; l++) v += (((k >> l) & 1) * 2 - 1) * fsets->coeff[i][j * 8 + l]; if ((int16_t)v != v) return AVERROR_INVALIDDATA; table[i][j][k] = v; } } } return 0; } static int decode_frame(AVCodecContext *avctx, AVFrame *frame, 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->ch_layout.nb_channels; DSTContext *s = avctx->priv_data; GetBitContext *gb = &s->gb; ArithCoder *ac = &s->ac; 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); ret = build_filter(s->filter, &s->fsets); if (ret < 0) return ret; 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 FFCodec ff_dst_decoder = { .p.name = "dst", CODEC_LONG_NAME("DST (Digital Stream Transfer)"), .p.type = AVMEDIA_TYPE_AUDIO, .p.id = AV_CODEC_ID_DST, .priv_data_size = sizeof(DSTContext), .init = decode_init, FF_CODEC_DECODE_CB(decode_frame), .p.capabilities = AV_CODEC_CAP_DR1, .p.sample_fmts = (const enum AVSampleFormat[]) { AV_SAMPLE_FMT_FLT, AV_SAMPLE_FMT_NONE }, };