view libmpcodecs/ad_hwac3.c @ 36782:06344efeded3

configure: improve FFmpeg check. If internal FFmpeg is not available we should fall back to external automatically instead of trying to build without (which is currently broken it seems). Also we can compile without internal copy as long as the necessary headers can be found. Two are required that FFmpeg does not install: libavformat/internal.h libavutil/x86/asm.h

/*
 * DTS code based on "ac3/decode_dts.c" and "ac3/conversion.c" from "ogle 0.9"
 * (see http://www.dtek.chalmers.se/~dvd/)
 * Reference: DOCS/tech/hwac3.txt !!!!!
 *
 * This file is part of MPlayer.
 *
 * MPlayer 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.
 *
 * MPlayer 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 MPlayer; if not, write to the Free Software Foundation, Inc.,
 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA.
 */

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

#include "config.h"
#include "mp_msg.h"
#include "help_mp.h"
#include "mpbswap.h"
#include "libavutil/common.h"
#include "libavutil/intreadwrite.h"

#include "ad_internal.h"


static int isdts = -1;

static const ad_info_t info =
{
  "AC3/DTS pass-through S/PDIF",
  "hwac3",
  "Nick Kurshev/Peter Schüller",
  "???",
  ""
};

LIBAD_EXTERN(hwac3)


static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate);
static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf);


static int a52_syncinfo (uint8_t *buf, int *sample_rate, int *bit_rate)
{
    static const uint16_t rate[] = { 32,  40,  48,  56,  64,  80,  96, 112,
                                    128, 160, 192, 224, 256, 320, 384, 448,
                                    512, 576, 640};
    int frmsizecod;
    int bitrate;
    int half;

    if (buf[0] != 0x0b || buf[1] != 0x77)    /* syncword */
        return 0;

    if (buf[5] >= 0x60)                      /* bsid >= 12 */
        return 0;
    half = buf[5] >> 3;
    half = FFMAX(half - 8, 0);

    frmsizecod = buf[4] & 63;
    if (frmsizecod >= 38)
        return 0;
    bitrate = rate[frmsizecod >> 1];
    *bit_rate = (bitrate * 1000) >> half;

    switch (buf[4] & 0xc0) {
    case 0:
        *sample_rate = 48000 >> half;
        return 4 * bitrate;
    case 0x40:
        *sample_rate = 44100 >> half;
        return 2 * (320 * bitrate / 147 + (frmsizecod & 1));
    case 0x80:
        *sample_rate = 32000 >> half;
        return 6 * bitrate;
    default:
        return 0;
    }
}

static int ac3dts_fillbuff(sh_audio_t *sh_audio)
{
  int length = 0;
  int flags = 0;
  int sample_rate = 0;
  int bit_rate = 0;

  sh_audio->a_in_buffer_len = 0;
  /* sync frame:*/
  while(1)
  {
    // Original code DTS has a 10 bytes header.
    // Now max 12 bytes for 14 bits DTS header.
    while(sh_audio->a_in_buffer_len < 12)
    {
      int c = demux_getc(sh_audio->ds);
      if(c<0)
        return -1; /* EOF*/
      sh_audio->a_in_buffer[sh_audio->a_in_buffer_len++] = c;
    }

    if (sh_audio->format == 0x2001)
    {
      length = dts_syncinfo(sh_audio->a_in_buffer, &flags, &sample_rate, &bit_rate);
      if(length >= 12)
      {
        if(isdts != 1)
        {
          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to DTS, %d bps, %d Hz\n", bit_rate, sample_rate);
          isdts = 1;
        }
        break;
      }
    }
    else
    {
      length = a52_syncinfo(sh_audio->a_in_buffer, &sample_rate, &bit_rate);
      if(length >= 7 && length <= 3840)
      {
        if(isdts != 0)
        {
          mp_msg(MSGT_DECAUDIO, MSGL_STATUS, "hwac3: switched to AC3, %d bps, %d Hz\n", bit_rate, sample_rate);
          isdts = 0;
        }
        break; /* we're done.*/
      }
    }
    /* bad file => resync*/
    memcpy(sh_audio->a_in_buffer, sh_audio->a_in_buffer + 1, 11);
    --sh_audio->a_in_buffer_len;
  }
  mp_msg(MSGT_DECAUDIO, MSGL_DBG2, "ac3dts: %s len=%d  flags=0x%X  %d Hz %d bit/s\n", isdts == 1 ? "DTS" : isdts == 0 ? "AC3" : "unknown", length, flags, sample_rate, bit_rate);

  sh_audio->samplerate = sample_rate;
  sh_audio->i_bps = bit_rate / 8;
  demux_read_data(sh_audio->ds, sh_audio->a_in_buffer + 12, length - 12);
  sh_audio->a_in_buffer_len = length;

  return length;
}


static int preinit(sh_audio_t *sh)
{
  /* Dolby AC3 audio: */
  sh->audio_out_minsize = 128 * 32 * 2 * 2; // DTS seems to need more than AC3
  sh->audio_in_minsize = 8192;
  sh->channels = 2;
  sh->samplesize = 2;
  sh->sample_format = AF_FORMAT_AC3_BE;
  // HACK for DTS where useless swapping can't easily be removed
  if (sh->format == 0x2001)
    sh->sample_format = AF_FORMAT_AC3_NE;
  return 1;
}

static int init(sh_audio_t *sh_audio)
{
  /* Dolby AC3 passthrough:*/
  if(ac3dts_fillbuff(sh_audio) < 0)
  {
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "AC3/DTS sync failed\n");
    return 0;
  }
  return 1;
}

static void uninit(sh_audio_t *sh)
{
}

static int control(sh_audio_t *sh,int cmd,void* arg, ...)
{
  switch(cmd)
  {
  case ADCTRL_RESYNC_STREAM:
  case ADCTRL_SKIP_FRAME:
      ac3dts_fillbuff(sh);
      return CONTROL_TRUE;
  }
  return CONTROL_UNKNOWN;
}


static int decode_audio(sh_audio_t *sh_audio,unsigned char *buf,int minlen,int maxlen)
{
  int len = sh_audio->a_in_buffer_len;

  if(len <= 0)
    if((len = ac3dts_fillbuff(sh_audio)) <= 0)
      return len; /*EOF*/
  sh_audio->a_in_buffer_len = 0;

  if(isdts == 1)
  {
    return decode_audio_dts(sh_audio->a_in_buffer, len, buf);
  }
  else if(isdts == 0)
  {
    AV_WB16(buf,     0xF872);   // iec 61937 syncword 1
    AV_WB16(buf + 2, 0x4E1F);   // iec 61937 syncword 2
    buf[4] = sh_audio->a_in_buffer[5] & 0x7; // bsmod
    buf[5] = 0x01;              // data-type ac3
    AV_WB16(buf + 6, len << 3); // number of bits in payload
    memcpy(buf + 8, sh_audio->a_in_buffer, len);
    memset(buf + 8 + len, 0, 6144 - 8 - len);

    return 6144;
  }
  else
    return -1;
}


static const int DTS_SAMPLEFREQS[16] =
{
  0,
  8000,
  16000,
  32000,
  64000,
  128000,
  11025,
  22050,
  44100,
  88200,
  176400,
  12000,
  24000,
  48000,
  96000,
  192000
};

static const int DTS_BITRATES[30] =
{
  32000,
  56000,
  64000,
  96000,
  112000,
  128000,
  192000,
  224000,
  256000,
  320000,
  384000,
  448000,
  512000,
  576000,
  640000,
  768000,
  896000,
  1024000,
  1152000,
  1280000,
  1344000,
  1408000,
  1411200,
  1472000,
  1536000,
  1920000,
  2048000,
  3072000,
  3840000,
  4096000
};

static int dts_decode_header(uint8_t *indata_ptr, int *rate, int *nblks, int *sfreq)
{
  int ftype;
  int surp;
  int unknown_bit;
  int fsize;
  int amode;

  int word_mode;
  int le_mode;

  unsigned int first4bytes = indata_ptr[0] << 24 | indata_ptr[1] << 16
                             | indata_ptr[2] << 8 | indata_ptr[3];

  switch(first4bytes)
  {
    /* 14 bits LE */
    case 0xff1f00e8:
      /* Also make sure frame type is 1. */
      if ((indata_ptr[4]&0xf0) != 0xf0 || indata_ptr[5] != 0x07)
        return -1;
      word_mode = 0;
      le_mode = 1;
      break;
    /* 14 bits BE */
    case 0x1fffe800:
      /* Also make sure frame type is 1. */
      if (indata_ptr[4] != 0x07 || (indata_ptr[5]&0xf0) != 0xf0)
        return -1;
      word_mode = 0;
      le_mode = 0;
      break;
    /* 16 bits LE */
    case 0xfe7f0180:
      word_mode = 1;
      le_mode = 1;
      break;
    /* 16 bits BE */
    case 0x7ffe8001:
      word_mode = 1;
      le_mode = 0;
      break;
    default:
      return -1;
  }

  if(word_mode)
  {
    /* First bit after first 32 bits:
       Frame type ( 1: Normal frame; 0: Termination frame ) */
    ftype = indata_ptr[4+le_mode] >> 7;

  if(ftype != 1)
  {
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Termination frames not handled, REPORT BUG\n");
    return -1;
  }
    /* Next 5 bits: Surplus Sample Count V SURP 5 bits */
    surp = indata_ptr[4+le_mode] >> 2 & 0x1f;
    /* Number of surplus samples */
    surp = (surp + 1) % 32;

    /* One unknown bit, crc? */
    unknown_bit = indata_ptr[4+le_mode] >> 1 & 0x01;

    /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
    *nblks = (indata_ptr[4+le_mode] & 0x01) << 6 | indata_ptr[5-le_mode] >> 2;
    /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
       encoded in the current frame per channel. */
    ++(*nblks);

    /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
       (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
       FSIZE defines the byte size of the current audio frame. */
    fsize = (indata_ptr[5-le_mode] & 0x03) << 12 | indata_ptr[6+le_mode] << 4
            | indata_ptr[7-le_mode] >> 4;
    ++fsize;

    /* Audio Channel Arrangement ACC AMODE 6 bits */
    amode = (indata_ptr[7-le_mode] & 0x0f) << 2 | indata_ptr[8+le_mode] >> 6;

    /* Source Sampling rate ACC SFREQ 4 bits */
    *sfreq = indata_ptr[8+le_mode] >> 2 & 0x0f;
    /* Transmission Bit Rate ACC RATE 5 bits */
    *rate = (indata_ptr[8+le_mode] & 0x03) << 3
            | (indata_ptr[9-le_mode] >> 5 & 0x07);
  }
  else
  {
    /* in the case judgement, we assure this */
    ftype = 1;
    surp = 0;
    /* 14 bits support, every 2 bytes, & 0x3fff, got used 14 bits */
    /* Bits usage:
       32 bits: Sync code (28 + 4)      1th and 2th word, 4 bits in 3th word
       1  bits: Frame type              1 bits in 3th word
       5  bits: SURP                    5 bits in 3th word
       1  bits: crc?                    1 bits in 3th word
       7  bits: NBLKS                   3 bits in 3th word, 4 bits in 4th word
       14 bits: FSIZE                   10 bits in 4th word, 4 bits in 5th word
                                        in 14 bits mode, FSIZE = FSIZE*8/14*2
       6  bits: AMODE                   6 bits in 5th word
       4  bits: SFREQ                   4 bits in 5th word
       5  bits: RATE                    5 bits in 6th word
       total bits: 75 bits    */

    /* NBLKS 7 bits: Valid Range=5-127, Invalid Range=0-4 */
    *nblks = (indata_ptr[5-le_mode] & 0x07) << 4
             | (indata_ptr[6+le_mode] & 0x3f) >> 2;
    /* NBLKS+1 indicates the number of 32 sample PCM audio blocks per channel
       encoded in the current frame per channel. */
    ++(*nblks);

    /* Frame Byte Size V FSIZE 14 bits: 0-94=Invalid, 95-8191=Valid range-1
       (ie. 96 bytes to 8192 bytes), 8192-16383=Invalid
       FSIZE defines the byte size of the current audio frame. */
    fsize = (indata_ptr[6+le_mode] & 0x03) << 12 | indata_ptr[7-le_mode] << 4
            | (indata_ptr[8+le_mode] & 0x3f) >> 2;
    ++fsize;
    fsize = fsize * 8 / 14 * 2;

    /* Audio Channel Arrangement ACC AMODE 6 bits */
    amode = (indata_ptr[8+le_mode] & 0x03) << 4
            | (indata_ptr[9-le_mode] & 0xf0) >> 4;

    /* Source Sampling rate ACC SFREQ 4 bits */
    *sfreq = indata_ptr[9-le_mode] & 0x0f;
    /* Transmission Bit Rate ACC RATE 5 bits */
    *rate = (indata_ptr[10+le_mode] & 0x3f) >> 1;
  }
#if 0
  if(*sfreq != 13)
  {
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: Only 48kHz supported, REPORT BUG\n");
    return -1;
  }
#endif
  if((fsize > 8192) || (fsize < 96))
  {
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: fsize: %d invalid, REPORT BUG\n", fsize);
    return -1;
  }

  if(*nblks != 8 &&
    *nblks != 16 &&
    *nblks != 32 &&
    *nblks != 64 &&
    *nblks != 128 &&
    ftype == 1)
  {
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: nblks %d not valid for normal frame, REPORT BUG\n", *nblks);
    return -1;
  }

  return fsize;
}

static int dts_syncinfo(uint8_t *indata_ptr, int *flags, int *sample_rate, int *bit_rate)
{
  int nblks;
  int fsize;
  int rate;
  int sfreq;

  fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
  if(fsize >= 0)
  {
    if(rate >= 0 && rate <= 29)
      *bit_rate = DTS_BITRATES[rate];
    else
      *bit_rate = 0;
    if(sfreq >= 1 && sfreq <= 15)
      *sample_rate = DTS_SAMPLEFREQS[sfreq];
    else
      *sample_rate = 0;
  }
  return fsize;
}

static int convert_14bits_to_16bits(const unsigned char *src,
                                    unsigned char *dest,
                                    int len,
                                    int is_le)
{
  uint16_t *p = (uint16_t *)dest;
  uint16_t buf = 0;
  int spacebits = 16;
  if (len <= 0) return 0;
  while (len > 0) {
    uint16_t v;
    if (len == 1)
      v = is_le ? src[0] : src[0] << 8;
    else
      v = is_le ? src[1] << 8 | src[0] : src[0] << 8 | src[1];
    v <<= 2;
    src += 2;
    len -= 2;
    buf |= v >> (16 - spacebits);
    spacebits -= 14;
    if (spacebits < 0) {
      *p++ = buf;
      spacebits += 16;
      buf = v << (spacebits - 2);
    }
  }
  *p++ = buf;
  return (unsigned char *)p - dest;
}

static int decode_audio_dts(unsigned char *indata_ptr, int len, unsigned char *buf)
{
  int nblks;
  int fsize;
  int rate;
  int sfreq;
  int nr_samples;
  int convert_16bits = 0;
  uint16_t *buf16 = (uint16_t *)buf;

  fsize = dts_decode_header(indata_ptr, &rate, &nblks, &sfreq);
  if(fsize < 0)
    return -1;
  nr_samples = nblks * 32;

  buf16[0] = 0xf872; /* iec 61937     */
  buf16[1] = 0x4e1f; /*  syncword     */
  switch(nr_samples)
  {
  case 512:
    buf16[2] = 0x000b;      /* DTS-1 (512-sample bursts) */
    break;
  case 1024:
    buf16[2] = 0x000c;      /* DTS-2 (1024-sample bursts) */
    break;
  case 2048:
    buf16[2] = 0x000d;      /* DTS-3 (2048-sample bursts) */
    break;
  default:
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: %d-sample bursts not supported\n", nr_samples);
    buf16[2] = 0x0000;
    break;
  }

  if(fsize + 8 > nr_samples * 2 * 2)
  {
    // dts wav (14bits LE) match this condition, one way to passthrough
    // is not add iec 61937 header, decoders will notice the dts header
    // and identify the dts stream. Another way here is convert
    // the stream from 14 bits to 16 bits.
    if ((indata_ptr[0] == 0xff || indata_ptr[0] == 0x1f)
        && fsize * 14 / 16 + 8 <= nr_samples * 2 * 2) {
      // The input stream is 14 bits, we can shrink it to 16 bits
      // to save space for add the 61937 header
      fsize = convert_14bits_to_16bits(indata_ptr,
                                       &buf[8],
                                       fsize,
                                       indata_ptr[0] == 0xff /* is LE */
                                       );
      mp_msg(MSGT_DECAUDIO, MSGL_DBG3, "DTS: shrink 14 bits stream to "
             "16 bits %02x%02x%02x%02x => %02x%02x%02x%02x, new size %d.\n",
             indata_ptr[0], indata_ptr[1], indata_ptr[2], indata_ptr[3],
             buf[8], buf[9], buf[10], buf[11], fsize);
      convert_16bits = 1;
    }
    else
    mp_msg(MSGT_DECAUDIO, MSGL_ERR, "DTS: more data than fits\n");
  }

  buf16[3] = fsize << 3;

  if (!convert_16bits) {
    int be_stream = indata_ptr[0] == 0x1f || indata_ptr[0] == 0x7f;
    if (be_stream == HAVE_BIGENDIAN)
      memcpy(&buf[8], indata_ptr, fsize);
    else
    {
      int i = fsize >> 1;
      uint16_t *d = buf16 + 4;
      uint8_t *s = indata_ptr;
      while (i--) {
        *d++ = HAVE_BIGENDIAN ? AV_RL16(s) : AV_RB16(s);
        s += 2;
      }
      if (fsize & 1) {
        // treat as if there was an additional 0
        *d++ = HAVE_BIGENDIAN ? *s : *s << 8;
        fsize++;
      }
    }
  }
  memset(&buf[fsize + 8], 0, nr_samples * 2 * 2 - (fsize + 8));

  return nr_samples * 2 * 2;
}