| /* |
| * 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]; |
| 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 }, |
| }; |
