ac3.c

/*
 * Support for ATSC Digital Audio Compression Standard, Revision B
 * (AC3) audio streams.
 *
 * ***** BEGIN LICENSE BLOCK *****
 * Version: MPL 1.1
 *
 * The contents of this file are subject to the Mozilla Public License Version
 * 1.1 (the "License"); you may not use this file except in compliance with
 * the License. You may obtain a copy of the License at
 * http://www.mozilla.org/MPL/
 *
 * Software distributed under the License is distributed on an "AS IS" basis,
 * WITHOUT WARRANTY OF ANY KIND, either express or implied. See the License
 * for the specific language governing rights and limitations under the
 * License.
 *
 * The Original Code is the MPEG TS, PS and ES tools.
 *
 * The Initial Developer of the Original Code is Amino Communications Ltd.
 * Portions created by the Initial Developer are Copyright (C) 2008
 * the Initial Developer. All Rights Reserved.
 *
 * Contributor(s):
 *   Kynesim, Cambridge UK
 *
 * ***** END LICENSE BLOCK *****
 */

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

#include "compat.h"
#include "printing_fns.h"
#include "misc_fns.h"
#include "ac3_fns.h"

static const unsigned int // Table 5.18, frame sizes
l_frmsizecod[19][3] = {
  { 64, 69, 96 },
  { 80, 87, 120 },
  { 96, 104, 144 },
  { 112, 121, 168 },
  { 128, 139, 192 },
  { 160, 174, 240 },
  { 192, 208, 288 },
  { 224, 243, 336 },
  { 256, 278, 384 },
  { 320, 348, 480 },
  { 384, 417, 576 },
  { 448, 487, 672 },
  { 512, 557, 768 },
  { 640, 696, 960 },
  { 768, 835, 1152 },
  { 896, 975, 1344 },
  { 1024, 1114, 1536 },
  { 1152, 1253, 1728 },
  { 1280, 1393, 1920 }
};


/*
 * Read the next AC3 frame.
 *
 * Assumes that the input stream is synchronised - i.e., it does not
 * try to cope if the next two bytes are not '0000 1011 0111 0111'
 *
 * - `file` is the file descriptor of the AC3 file to read from
 * - `frame` is the AC3 frame that is read
 *
 * Returns 0 if all goes well, EOF if end-of-file is read, and 1 if something
 * goes wrong.
 */
#define SYNCINFO_SIZE 5

int read_next_ac3_frame(int            file,
			audio_frame_p *frame)
{
  int   i, err;
  byte  sync_info[SYNCINFO_SIZE];
  byte *data = NULL;
  int   fscod;
  int   frmsizecod;
  int   frame_length;

  offset_t posn = tell_file(file);

  err = read_bytes(file, SYNCINFO_SIZE, sync_info);
  if (err == EOF)
    return EOF;
  else if (err)
  {
    fprint_err("### Error reading syncinfo from AC3 file\n"
               "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
    return 1;
  }

  if (sync_info[0] != 0x0b || sync_info[1] != 0x77)
  {
    fprint_err("### AC3 frame does not start with 0x0b77"
               " syncword - lost synchronisation?\n"
               "    Found 0x%02x%02x instead of 0x0b77\n",
               (unsigned)sync_info[0], (unsigned)sync_info[1]);
    fprint_err("    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
    return 1;
  }

  fscod = sync_info[4] >> 6;
  if (fscod == 3)
  {
    // Bad sample rate code
    fprint_err("### Bad sample rate code in AC3 syncinfo\n"
               "    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
    return 1;
  }

  frmsizecod = sync_info[4] & 0x3f;
  if (frmsizecod > 37)
  {
    fprint_err("### Bad frame size code %d in AC3 syncinfo\n",
                frmsizecod);
    fprint_err("    (in frame starting at " OFFSET_T_FORMAT ")\n", posn);
    return 1;
  }

  frame_length = l_frmsizecod[frmsizecod >> 1][fscod];
  if (fscod == 1)
    frame_length += frmsizecod & 1;
  frame_length <<= 1;  // Convert from 16-bit words to bytes

  data = malloc(frame_length);
  if (data == NULL)
  {
    print_err("### Unable to extend data buffer for AC3 frame\n");
    return 1;
  }
  for (i = 0; i < SYNCINFO_SIZE; i++)
    data[i] = sync_info[i];

  err = read_bytes(file, frame_length - SYNCINFO_SIZE,
                   &data[SYNCINFO_SIZE]);
  if (err)
  {
    if (err == EOF)
      print_err("### Unexpected EOF reading rest of AC3 frame\n");
    else
      print_err("### Error reading rest of AC3 frame\n");
    free(data);
    return 1;
  }

  err = build_audio_frame(frame);
  if (err)
  {
    free(data);
    return 1;
  }
  (*frame)->data = data;
  (*frame)->data_len = frame_length;

  return 0;
}


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