FFmpeg/libavcodec/dtsdec.c

/*
 * dtsdec.c : free DTS Coherent Acoustics stream decoder.
 * Copyright (C) 2004 Benjamin Zores <ben@geexbox.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 General Public License as published by
 * the Free Software Foundation; either version 2 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 General Public License for more details.
 *
 * You should have received a copy of the GNU General Public License
 * along with FFmpeg; if not, write to the Free Software
 * Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
 */

#ifdef HAVE_AV_CONFIG_H
#undef HAVE_AV_CONFIG_H
#endif

#include "avcodec.h"
#include <dts.h>

#include <stdlib.h>
#include <string.h>

#ifdef HAVE_MALLOC_H
#include <malloc.h>
#endif

#define BUFFER_SIZE 18726
#define HEADER_SIZE 14

#ifdef LIBDTS_FIXED
#define CONVERT_LEVEL (1 << 26)
#define CONVERT_BIAS 0
#else
#define CONVERT_LEVEL 1
#define CONVERT_BIAS 384
#endif

static inline
int16_t convert (int32_t i)
{
#ifdef LIBDTS_FIXED
    i >>= 15;
#else
    i -= 0x43c00000;
#endif
    return (i > 32767) ? 32767 : ((i < -32768) ? -32768 : i);
}

static void
convert2s16_2 (sample_t * _f, int16_t * s16)
{
  int i;
  int32_t * f = (int32_t *) _f;

  for (i = 0; i < 256; i++)
    {
      s16[2*i] = convert (f[i]);
      s16[2*i+1] = convert (f[i+256]);
    }
}

static void
convert2s16_4 (sample_t * _f, int16_t * s16)
{
  int i;
  int32_t * f = (int32_t *) _f;

  for (i = 0; i < 256; i++)
    {
      s16[4*i] = convert (f[i]);
      s16[4*i+1] = convert (f[i+256]);
      s16[4*i+2] = convert (f[i+512]);
      s16[4*i+3] = convert (f[i+768]);
    }
}

static void
convert2s16_5 (sample_t * _f, int16_t * s16)
{
  int i;
  int32_t * f = (int32_t *) _f;

  for (i = 0; i < 256; i++)
    {
      s16[5*i] = convert (f[i]);
      s16[5*i+1] = convert (f[i+256]);
      s16[5*i+2] = convert (f[i+512]);
      s16[5*i+3] = convert (f[i+768]);
      s16[5*i+4] = convert (f[i+1024]);
    }
}

static void
convert2s16_multi (sample_t * _f, int16_t * s16, int flags)
{
  int i;
  int32_t * f = (int32_t *) _f;

  switch (flags)
    {
    case DTS_MONO:
      for (i = 0; i < 256; i++)
        {
          s16[5*i] = s16[5*i+1] = s16[5*i+2] = s16[5*i+3] = 0;
          s16[5*i+4] = convert (f[i]);
        }
      break;
    case DTS_CHANNEL:
    case DTS_STEREO:
    case DTS_DOLBY:
      convert2s16_2 (_f, s16);
      break;
    case DTS_3F:
      for (i = 0; i < 256; i++)
        {
          s16[5*i] = convert (f[i]);
          s16[5*i+1] = convert (f[i+512]);
          s16[5*i+2] = s16[5*i+3] = 0;
          s16[5*i+4] = convert (f[i+256]);
        }
      break;
    case DTS_2F2R:
      convert2s16_4 (_f, s16);
      break;
    case DTS_3F2R:
      convert2s16_5 (_f, s16);
      break;
    case DTS_MONO | DTS_LFE:
      for (i = 0; i < 256; i++)
        {
          s16[6*i] = s16[6*i+1] = s16[6*i+2] = s16[6*i+3] = 0;
          s16[6*i+4] = convert (f[i+256]);
          s16[6*i+5] = convert (f[i]);
        }
      break;
    case DTS_CHANNEL | DTS_LFE:
    case DTS_STEREO | DTS_LFE:
    case DTS_DOLBY | DTS_LFE:
      for (i = 0; i < 256; i++)
        {
          s16[6*i] = convert (f[i+256]);
          s16[6*i+1] = convert (f[i+512]);
          s16[6*i+2] = s16[6*i+3] = s16[6*i+4] = 0;
          s16[6*i+5] = convert (f[i]);
        }
      break;
    case DTS_3F | DTS_LFE:
      for (i = 0; i < 256; i++)
        {
          s16[6*i] = convert (f[i+256]);
          s16[6*i+1] = convert (f[i+768]);
          s16[6*i+2] = s16[6*i+3] = 0;
          s16[6*i+4] = convert (f[i+512]);
          s16[6*i+5] = convert (f[i]);
        }
      break;
    case DTS_2F2R | DTS_LFE:
      for (i = 0; i < 256; i++)
        {
          s16[6*i] = convert (f[i+256]);
          s16[6*i+1] = convert (f[i+512]);
          s16[6*i+2] = convert (f[i+768]);
          s16[6*i+3] = convert (f[i+1024]);
          s16[6*i+4] = 0;
          s16[6*i+5] = convert (f[i]);
        }
      break;
    case DTS_3F2R | DTS_LFE:
      for (i = 0; i < 256; i++)
        {
          s16[6*i] = convert (f[i+256]);
          s16[6*i+1] = convert (f[i+768]);
          s16[6*i+2] = convert (f[i+1024]);
          s16[6*i+3] = convert (f[i+1280]);
          s16[6*i+4] = convert (f[i+512]);
          s16[6*i+5] = convert (f[i]);
        }
      break;
    }
}

static int
channels_multi (int flags)
{
  if (flags & DTS_LFE)
    return 6;
  else if (flags & 1)   /* center channel */
    return 5;
  else if ((flags & DTS_CHANNEL_MASK) == DTS_2F2R)
    return 4;
  else
    return 2;
}

static int
dts_decode_frame (AVCodecContext *avctx, void *data, int *data_size,
                  uint8_t *buff, int buff_size)
{
  uint8_t * start = buff;
  uint8_t * end = buff + buff_size;
  static uint8_t buf[BUFFER_SIZE];
  static uint8_t * bufptr = buf;
  static uint8_t * bufpos = buf + HEADER_SIZE;

  static int sample_rate;
  static int frame_length;
  static int flags;
  int bit_rate;
  int len;
  dts_state_t *state = avctx->priv_data;

  *data_size = 0;

  while (1)
    {
      len = end - start;
      if (!len)
        break;
      if (len > bufpos - bufptr)
        len = bufpos - bufptr;
      memcpy (bufptr, start, len);
      bufptr += len;
      start += len;
      if (bufptr != bufpos)
          return start - buff;
      if (bufpos != buf + HEADER_SIZE)
          break;

            {
              int length;

              length = dts_syncinfo (state, buf, &flags, &sample_rate,
                                     &bit_rate, &frame_length);
              if (!length)
                {
                  av_log (NULL, AV_LOG_INFO, "skip\n");
                  for (bufptr = buf; bufptr < buf + HEADER_SIZE-1; bufptr++)
                    bufptr[0] = bufptr[1];
                  continue;
                }
              bufpos = buf + length;
            }
    }

            {
              level_t level;
              sample_t bias;
              int i;

              flags = 2; /* ???????????? */
              level = CONVERT_LEVEL;
              bias = CONVERT_BIAS;

              flags |= DTS_ADJUST_LEVEL;
              if (dts_frame (state, buf, &flags, &level, bias))
                goto error;
              avctx->sample_rate = sample_rate;
              avctx->channels = channels_multi (flags);
              avctx->bit_rate = bit_rate;
              for (i = 0; i < dts_blocks_num (state); i++)
                {
                  if (dts_block (state))
                    goto error;
                  {
                    int chans;
                    chans = channels_multi (flags);
                    convert2s16_multi (dts_samples (state), data,
                                       flags & (DTS_CHANNEL_MASK | DTS_LFE));

                    data += 256 * sizeof (int16_t) * chans;
                    *data_size += 256 * sizeof (int16_t) * chans;
                  }
                }
              bufptr = buf;
              bufpos = buf + HEADER_SIZE;
              return start-buff;
            error:
              av_log (NULL, AV_LOG_ERROR, "error\n");
              bufptr = buf;
              bufpos = buf + HEADER_SIZE;
            }

  return start-buff;
}

static int
dts_decode_init (AVCodecContext *avctx)
{
  avctx->priv_data = dts_init (0);
  if (avctx->priv_data == NULL)
    return -1;

  return 0;
}

static int
dts_decode_end (AVCodecContext *s)
{
  return 0;
}

AVCodec dts_decoder = {
  "dts",
  CODEC_TYPE_AUDIO,
  CODEC_ID_DTS,
  sizeof (dts_state_t *),
  dts_decode_init,
  NULL,
  dts_decode_end,
  dts_decode_frame,
};