text: /*
   * Bink Audio decoder
   * Copyright (c) 2007-2011 Peter Ross (pross@xvid.org)
   * Copyright (c) 2009 Daniel Verkamp (daniel@drv.nu)
   *
   * 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
   * Bink Audio decoder
   *
   * Technical details here:
   *  http://wiki.multimedia.cx/index.php?title=Bink_Audio
   */
  
  #include "libavutil/channel_layout.h"
  #include "avcodec.h"
  #define BITSTREAM_READER_LE
  #include "get_bits.h"
  #include "dct.h"
  #include "rdft.h"
  #include "fmtconvert.h"
  #include "internal.h"
  #include "wma.h"
  #include "libavutil/intfloat.h"
  
  static float quant_table[96];
  
  #define MAX_CHANNELS 2
  #define BINK_BLOCK_MAX_SIZE (MAX_CHANNELS << 11)
  
  typedef struct {
      GetBitContext gb;
      int version_b;          ///< Bink version 'b'
      int first;
      int channels;
      int frame_len;          ///< transform size (samples)
      int overlap_len;        ///< overlap size (samples)
      int block_size;
      int num_bands;
      unsigned int *bands;
      float root;
      DECLARE_ALIGNED(32, FFTSample, coeffs)[BINK_BLOCK_MAX_SIZE];
      float previous[MAX_CHANNELS][BINK_BLOCK_MAX_SIZE / 16];  ///< coeffs from previous audio block
      uint8_t *packet_buffer;
      union {
          RDFTContext rdft;
          DCTContext dct;
      } trans;
  } BinkAudioContext;
  
  
  static av_cold int decode_init(AVCodecContext *avctx)
  {
      BinkAudioContext *s = avctx->priv_data;
      int sample_rate = avctx->sample_rate;
      int sample_rate_half;
      int i;
      int frame_len_bits;
  
      /* determine frame length */
      if (avctx->sample_rate < 22050) {
          frame_len_bits = 9;
      } else if (avctx->sample_rate < 44100) {
          frame_len_bits = 10;
      } else {
          frame_len_bits = 11;
      }
  
      if (avctx->channels < 1 || avctx->channels > MAX_CHANNELS) {
          av_log(avctx, AV_LOG_ERROR, "invalid number of channels: %d\n", avctx->channels);
          return AVERROR_INVALIDDATA;
      }
      avctx->channel_layout = avctx->channels == 1 ? AV_CH_LAYOUT_MONO :
                                                     AV_CH_LAYOUT_STEREO;
  
      s->version_b = avctx->extradata_size >= 4 && avctx->extradata[3] == 'b';
  
      if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT) {
          // audio is already interleaved for the RDFT format variant
          avctx->sample_fmt = AV_SAMPLE_FMT_FLT;
          sample_rate  *= avctx->channels;
          s->channels = 1;
          if (!s->version_b)
              frame_len_bits += av_log2(avctx->channels);
      } else {
          s->channels = avctx->channels;
          avctx->sample_fmt = AV_SAMPLE_FMT_FLTP;
      }
  
      s->frame_len     = 1 << frame_len_bits;
      s->overlap_len   = s->frame_len / 16;
      s->block_size    = (s->frame_len - s->overlap_len) * s->channels;
      sample_rate_half = (sample_rate + 1) / 2;
      if (avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
          s->root = 2.0 / (sqrt(s->frame_len) * 32768.0);
      else
          s->root = s->frame_len / (sqrt(s->frame_len) * 32768.0);
      for (i = 0; i < 96; i++) {
          /* constant is result of 0.066399999/log10(M_E) */
          quant_table[i] = expf(i * 0.15289164787221953823f) * s->root;
      }
  
      /* calculate number of bands */
      for (s->num_bands = 1; s->num_bands < 25; s->num_bands++)
          if (sample_rate_half <= ff_wma_critical_freqs[s->num_bands - 1])
              break;
  
      s->bands = av_malloc((s->num_bands + 1) * sizeof(*s->bands));
      if (!s->bands)
          return AVERROR(ENOMEM);
  
      /* populate bands data */
      s->bands[0] = 2;
      for (i = 1; i < s->num_bands; i++)
          s->bands[i] = (ff_wma_critical_freqs[i - 1] * s->frame_len / sample_rate_half) & ~1;
      s->bands[s->num_bands] = s->frame_len;
  
      s->first = 1;
  
      if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
          ff_rdft_init(&s->trans.rdft, frame_len_bits, DFT_C2R);
      else if (CONFIG_BINKAUDIO_DCT_DECODER)
          ff_dct_init(&s->trans.dct, frame_len_bits, DCT_III);
      else
          return -1;
  
      return 0;
  }
  
  static float get_float(GetBitContext *gb)
  {
      int power = get_bits(gb, 5);
      float f = ldexpf(get_bits_long(gb, 23), power - 23);
      if (get_bits1(gb))
          f = -f;
      return f;
  }
  
  static const uint8_t rle_length_tab[16] = {
      2, 3, 4, 5, 6, 8, 9, 10, 11, 12, 13, 14, 15, 16, 32, 64
  };
  
  /**
   * Decode Bink Audio block
   * @param[out] out Output buffer (must contain s->block_size elements)
   * @return 0 on success, negative error code on failure
   */
  static int decode_block(BinkAudioContext *s, float **out, int use_dct)
  {
      int ch, i, j, k;
      float q, quant[25];
      int width, coeff;
      GetBitContext *gb = &s->gb;
  
      if (use_dct)
          skip_bits(gb, 2);
  
      for (ch = 0; ch < s->channels; ch++) {
          FFTSample *coeffs = out[ch];
  
          if (s->version_b) {
              if (get_bits_left(gb) < 64)
                  return AVERROR_INVALIDDATA;
              coeffs[0] = av_int2float(get_bits_long(gb, 32)) * s->root;
              coeffs[1] = av_int2float(get_bits_long(gb, 32)) * s->root;
          } else {
              if (get_bits_left(gb) < 58)
                  return AVERROR_INVALIDDATA;
              coeffs[0] = get_float(gb) * s->root;
              coeffs[1] = get_float(gb) * s->root;
          }
  
          if (get_bits_left(gb) < s->num_bands * 8)
              return AVERROR_INVALIDDATA;
          for (i = 0; i < s->num_bands; i++) {
              int value = get_bits(gb, 8);
              quant[i]  = quant_table[FFMIN(value, 95)];
          }
  
          k = 0;
          q = quant[0];
  
          // parse coefficients
          i = 2;
          while (i < s->frame_len) {
              if (s->version_b) {
                  j = i + 16;
              } else {
                  int v = get_bits1(gb);
                  if (v) {
                      v = get_bits(gb, 4);
                      j = i + rle_length_tab[v] * 8;
                  } else {
                      j = i + 8;
                  }
              }
  
              j = FFMIN(j, s->frame_len);
  
              width = get_bits(gb, 4);
              if (width == 0) {
                  memset(coeffs + i, 0, (j - i) * sizeof(*coeffs));
                  i = j;
                  while (s->bands[k] < i)
                      q = quant[k++];
              } else {
                  while (i < j) {
                      if (s->bands[k] == i)
                          q = quant[k++];
                      coeff = get_bits(gb, width);
                      if (coeff) {
                          int v;
                          v = get_bits1(gb);
                          if (v)
                              coeffs[i] = -q * coeff;
                          else
                              coeffs[i] =  q * coeff;
                      } else {
                          coeffs[i] = 0.0f;
                      }
                      i++;
                  }
              }
          }
  
          if (CONFIG_BINKAUDIO_DCT_DECODER && use_dct) {
              coeffs[0] /= 0.5;
              s->trans.dct.dct_calc(&s->trans.dct,  coeffs);
          }
          else if (CONFIG_BINKAUDIO_RDFT_DECODER)
              s->trans.rdft.rdft_calc(&s->trans.rdft, coeffs);
      }
  
      for (ch = 0; ch < s->channels; ch++) {
          int j;
          int count = s->overlap_len * s->channels;
          if (!s->first) {
              j = ch;
              for (i = 0; i < s->overlap_len; i++, j += s->channels)
                  out[ch][i] = (s->previous[ch][i] * (count - j) +
                                        out[ch][i] *          j) / count;
          }
          memcpy(s->previous[ch], &out[ch][s->frame_len - s->overlap_len],
                 s->overlap_len * sizeof(*s->previous[ch]));
      }
  
      s->first = 0;
  
      return 0;
  }
  
  static av_cold int decode_end(AVCodecContext *avctx)
  {
      BinkAudioContext * s = avctx->priv_data;
      av_freep(&s->bands);
      av_freep(&s->packet_buffer);
      if (CONFIG_BINKAUDIO_RDFT_DECODER && avctx->codec->id == AV_CODEC_ID_BINKAUDIO_RDFT)
          ff_rdft_end(&s->trans.rdft);
      else if (CONFIG_BINKAUDIO_DCT_DECODER)
          ff_dct_end(&s->trans.dct);
  
      return 0;
  }
  
  static void get_bits_align32(GetBitContext *s)
  {
      int n = (-get_bits_count(s)) & 31;
      if (n) skip_bits(s, n);
  }
  
  static int decode_frame(AVCodecContext *avctx, void *data,
                          int *got_frame_ptr, AVPacket *avpkt)
  {
      BinkAudioContext *s = avctx->priv_data;
      AVFrame *frame      = data;
      GetBitContext *gb = &s->gb;
      int ret, consumed = 0;
  
      if (!get_bits_left(gb)) {
          uint8_t *buf;
          /* handle end-of-stream */
          if (!avpkt->size) {
              *got_frame_ptr = 0;
              return 0;
          }
          if (avpkt->size < 4) {
              av_log(avctx, AV_LOG_ERROR, "Packet is too small\n");
              return AVERROR_INVALIDDATA;
          }
          buf = av_realloc(s->packet_buffer, avpkt->size + FF_INPUT_BUFFER_PADDING_SIZE);
          if (!buf)
              return AVERROR(ENOMEM);
          s->packet_buffer = buf;
          memcpy(s->packet_buffer, avpkt->data, avpkt->size);
          init_get_bits(gb, s->packet_buffer, avpkt->size * 8);
          consumed = avpkt->size;
  
          /* skip reported size */
          skip_bits_long(gb, 32);
      }
  
      /* get output buffer */
      frame->nb_samples = s->frame_len;
      if ((ret = ff_get_buffer(avctx, frame, 0)) < 0)
          return ret;
  
      if (decode_block(s, (float **)frame->extended_data,
                       avctx->codec->id == AV_CODEC_ID_BINKAUDIO_DCT)) {
          av_log(avctx, AV_LOG_ERROR, "Incomplete packet\n");
          return AVERROR_INVALIDDATA;
      }
      get_bits_align32(gb);
  
      frame->nb_samples = s->block_size / avctx->channels;
      *got_frame_ptr    = 1;
  
      return consumed;
  }
  
  AVCodec ff_binkaudio_rdft_decoder = {
      .name           = "binkaudio_rdft",
      .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (RDFT)"),
      .type           = AVMEDIA_TYPE_AUDIO,
      .id             = AV_CODEC_ID_BINKAUDIO_RDFT,
      .priv_data_size = sizeof(BinkAudioContext),
      .init           = decode_init,
      .close          = decode_end,
      .decode         = decode_frame,
      .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,
  };
  
  AVCodec ff_binkaudio_dct_decoder = {
      .name           = "binkaudio_dct",
      .long_name      = NULL_IF_CONFIG_SMALL("Bink Audio (DCT)"),
      .type           = AVMEDIA_TYPE_AUDIO,
      .id             = AV_CODEC_ID_BINKAUDIO_DCT,
      .priv_data_size = sizeof(BinkAudioContext),
      .init           = decode_init,
      .close          = decode_end,
      .decode         = decode_frame,
      .capabilities   = CODEC_CAP_DELAY | CODEC_CAP_DR1,
  };
  