text: /*
   * 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(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;
      }
  
      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_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 < 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 },
  };
  