mkvtoolnix/p_dts.cpp

/*
  mkvmerge -- utility for splicing together matroska files
      from component media subtypes

  p_dts.h

  Written by Moritz Bunkus <moritz@bunkus.org>

  Distributed under the GPL
  see the file COPYING for details
  or visit http://www.gnu.org/copyleft/gpl.html
*/

/*!
    \file
    \version $Id$
    \brief DTS output module
    \author Moritz Bunkus <moritz@bunkus.org>
*/

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

#include "pr_generic.h"
#include "dts_common.h"
#include "p_dts.h"
#include "matroska.h"

using namespace LIBMATROSKA_NAMESPACE;

bool operator!=(const dts_header_t &l, const dts_header_t &r) {
  //if (l.frametype != r.frametype) return true;
  //if (l.deficit_sample_count != r.deficit_sample_count) return true;
  if (l.crc_present != r.crc_present)
    return true;
  if (l.num_pcm_sample_blocks != r.num_pcm_sample_blocks)
    return true;
  if (l.frame_byte_size != r.frame_byte_size)
    return true;
  if (l.audio_channels != r.audio_channels)
    return true;
  if (l.core_sampling_frequency != r.core_sampling_frequency)
    return true;
  if (l.transmission_bitrate != r.transmission_bitrate)
    return true;
  if (l.embedded_down_mix != r.embedded_down_mix)
    return true;
  if (l.embedded_dynamic_range != r.embedded_dynamic_range)
    return true;
  if (l.embedded_time_stamp != r.embedded_time_stamp)
    return true;
  if (l.auxiliary_data != r.auxiliary_data)
    return true;
  if (l.hdcd_master != r.hdcd_master)
    return true;
  if (l.extension_audio_descriptor != r.extension_audio_descriptor)
    return true;
  if (l.extended_coding != r.extended_coding)
    return true;
  if (l.audio_sync_word_in_sub_sub != r.audio_sync_word_in_sub_sub)
    return true;
  if (l.lfe_type != r.lfe_type)
    return true;
  if (l.predictor_history_flag != r.predictor_history_flag)
    return true;
  if (l.multirate_interpolator != r.multirate_interpolator)
    return true;
  if (l.encoder_software_revision != r.encoder_software_revision)
    return true;
  if (l.copy_history != r.copy_history)
    return true;
  if (l.source_pcm_resolution != r.source_pcm_resolution)
    return true;
  if (l.source_surround_in_es != r.source_surround_in_es)
    return true;
  if (l.front_sum_difference != r.front_sum_difference)
    return true;
  if (l.surround_sum_difference != r.surround_sum_difference)
    return true;
  if (l.dialog_normalization_gain != r.dialog_normalization_gain)
    return true;
  return false;
}

dts_packetizer_c::dts_packetizer_c(generic_reader_c *nreader,
                                   const dts_header_t &dtsheader,
                                   track_info_t *nti)
  throw (error_c): generic_packetizer_c(nreader, nti) {
  //packetno = 0;
  samples_written = 0;
  bytes_written = 0;

  packet_buffer = NULL;
  buffer_size = 0;
  skipping_is_normal = false;

  first_header = dtsheader;
  last_header = dtsheader;

  set_track_type(track_audio);
  duplicate_data_on_add(false);
}

dts_packetizer_c::~dts_packetizer_c() {
  fprintf(stderr,"wrote %lld bytes DTS data equivalent of %lld PCM samples at "
          "%d Hz, that is %f seconds of sound\n",
          bytes_written, samples_written, first_header.core_sampling_frequency,
          ((double)samples_written) /
          (double)first_header.core_sampling_frequency);
  safefree(packet_buffer);
}

void dts_packetizer_c::add_to_buffer(unsigned char *buf, int size) {
  unsigned char *new_buffer;

  new_buffer = (unsigned char *)saferealloc(packet_buffer, buffer_size + size);

  memcpy(new_buffer + buffer_size, buf, size);
  packet_buffer = new_buffer;
  buffer_size += size;
}

int dts_packetizer_c::dts_packet_available() {
  int pos;
  dts_header_t dtsheader;

  if (packet_buffer == NULL)
    return 0;

  pos = find_dts_header(packet_buffer, buffer_size, &dtsheader);
  if (pos < 0)
    return 0;

  return 1;
}

void dts_packetizer_c::remove_dts_packet(int pos, int framesize) {
  int new_size;
  unsigned char *temp_buf;

  new_size = buffer_size - (pos + framesize);
  if (new_size != 0)
    temp_buf = (unsigned char *)safememdup(&packet_buffer[pos + framesize],
                                           new_size);
  else
    temp_buf = NULL;
  safefree(packet_buffer);
  packet_buffer = temp_buf;
  buffer_size = new_size;
}

unsigned char *dts_packetizer_c::get_dts_packet(dts_header_t &dtsheader) {
  int pos;
  unsigned char *buf;
  double pims;

  if (packet_buffer == NULL)
    return 0;
  pos = find_dts_header(packet_buffer, buffer_size, &dtsheader);
  if (pos < 0)
    return 0;
  if ((pos + dtsheader.frame_byte_size) > buffer_size)
    return 0;

  if (dtsheader != last_header) {
    fprintf(stderr,"DTS header information changed! - New format:\n");
    print_dts_header(&dtsheader);
    last_header = dtsheader;
  }


  pims = get_dts_packet_length_in_milliseconds(&dtsheader);

  if (ti->async.displacement < 0) {
    /*
     * DTS audio synchronization. displacement < 0 means skipping an
     * appropriate number of packets at the beginning.
     */
    ti->async.displacement += (int)pims;
    if (ti->async.displacement > -(pims / 2))
      ti->async.displacement = 0;

    remove_dts_packet(pos, dtsheader.frame_byte_size);

    return 0;
  }

  if (verbose && (pos > 0) && !skipping_is_normal)
    fprintf(stdout, "dts_packetizer: skipping %d bytes (no valid DTS header "
            "found). This might make audio/video go out of sync, but this "
            "stream is damaged.\n", pos);

  buf = (unsigned char *)safememdup(packet_buffer + pos,
                                    dtsheader.frame_byte_size);

  if (ti->async.displacement > 0) {
    /*
     * DTS audio synchronization. displacement > 0 is solved by duplicating
     * the very first DTS packet as often as necessary. I cannot create
     * a packet with total silence because I don't know how, and simply
     * settings the packet's values to 0 does not work as the DTS header
     * contains a CRC of its data.
     */
    ti->async.displacement -= (int)pims;
    if (ti->async.displacement < (pims / 2))
      ti->async.displacement = 0;

    return buf;
  }

  remove_dts_packet(pos, dtsheader.frame_byte_size);

  return buf;
}

void dts_packetizer_c::set_headers() {
  set_codec_id(MKV_A_DTS);
  set_audio_sampling_freq((float)first_header.core_sampling_frequency);
  set_audio_channels(first_header.audio_channels);

  generic_packetizer_c::set_headers();
}

int dts_packetizer_c::process(unsigned char *buf, int size,
                              int64_t timecode, int64_t, int64_t, int64_t) {

  int64_t my_timecode;

  debug_enter("dts_packetizer_c::process");

  if (timecode != -1)
    my_timecode = timecode;

  add_to_buffer(buf, size);

  dts_header_t dtsheader;
  unsigned char *packet;
  while ((packet = get_dts_packet(dtsheader)) != NULL) {
    int64_t packet_len_in_ms =
      (int64_t)get_dts_packet_length_in_milliseconds(&dtsheader);

    if (timecode == -1)
      my_timecode = (int64_t)(((double)samples_written*1000.0) /
                              ((double)dtsheader.core_sampling_frequency));

//     fprintf(stderr,"DTS packet timecode %lld  len %lld\n", my_timecode,
//             packet_len_in_ms);

    add_packet(packet, dtsheader.frame_byte_size, my_timecode,
               packet_len_in_ms);

    bytes_written += dtsheader.frame_byte_size;
    samples_written += get_dts_packet_length_in_core_samples(&dtsheader);
  }

  debug_leave("dts_packetizer_c::process");

  return EMOREDATA;
}

void dts_packetizer_c::dump_debug_info() {
  fprintf(stderr, "DBG> dts_packetizer_c: queue: %d\n", packet_queue.size());
}