/* -*- Mode: C++; tab-width: 2; indent-tabs-mode: nil; c-basic-offset: 2 -*- */
/***************************************************************************
 *            liblame_wrapper.cc
 *
 *  Sat Jul  2 11:11:34 CEST 2005
 *  Copyright  2005 Bent Bisballe
 *  deva@aasimon.org
 ****************************************************************************/

/*
 *  This file is part of MIaV.
 *
 *  MIaV 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.
 *
 *  MIaV 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 MIaV; if not, write to the Free Software
 *  Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA  02111-1307  USA.
 */

/*
 * $Id$
 */

/*
 * $Log$
 * Revision 1.1  2005/07/02 11:39:51  deva
 * Added some audiocode.
 * Moved libfame code out of mov_encoder
 *
 */

#include <config.h>
#include "liblame_wrapper.h"

LibLAMEWrapper::LibLAMEWrapper(Info *i)
{
  info = i;

  // Init library.
  lamegf = lame_init();
  if(!lamegf) {
    info->error("LAME initialization failed (due to malloc failure!)");
  }

  lame_init_params(lamegf);

  decoder = NULL;

  audio_buffer[0] = new int16_t[AUDIO_BUFFER_SIZE];
  audio_buffer[1] = new int16_t[AUDIO_BUFFER_SIZE];

}

LibLAMEWrapper::~LibLAMEWrapper()
{
  delete audio_buffer[0];
  delete audio_buffer[1];

}

Frame *LibLAMEWrapper::encode(Frame *dvframe)
{

  if(!decoder) {
    decoder = dv_decoder_new(FALSE/*this value is unused*/, FALSE, FALSE);
    decoder->quality = DV_QUALITY_BEST;
    
    dv_parse_header(decoder, dvframe->data);
  
    decoder->system = e_dv_system_625_50;  // PAL lines, PAL framerate
    decoder->sampling = e_dv_sample_422;  // 4 bytes y, 2 bytes u, 2 bytes v
    decoder->std = e_dv_std_iec_61834;
    decoder->num_dif_seqs = 12;
  }

  // See
  // http://www.koders.com/cpp/fidE614E999154E2B4A813DA272C4421633063C78CA.aspx
  // line 769

  /**
   * Decode audio using libdv
   */
  //  int n, i;
  //  int16_t* s = ( int16_t * ) sound;
  dv_decode_full_audio( decoder, dvframe->data, audio_buffer );
  /*
  for ( n = 0; n < SAMPLES; ++n )
    for ( i = 0; i < CHANNELS; i++ )
      *s++ = audio_buffer[ i ][ n ];
      */

/*
 * input pcm data, output (maybe) mp3 frames.
 * This routine handles all buffering, resampling and filtering for you.
 * 
 * The required mp3buf_size can be computed from num_samples, 
 * samplerate and encoding rate, but here is a worst case estimate:
 *
 * mp3buf_size in bytes = 1.25*num_samples + 7200
 *
 * I think a tighter bound could be:  (mt, March 2000)
 * MPEG1:
 *    num_samples*(bitrate/8)/samplerate + 4*1152*(bitrate/8)/samplerate + 512
 * MPEG2:
 *    num_samples*(bitrate/8)/samplerate + 4*576*(bitrate/8)/samplerate + 256
 *
 * but test first if you use that!
 *
 * set mp3buf_size = 0 and LAME will not check if mp3buf_size is
 * large enough.
 *
 * NOTE:
 * if gfp->num_channels=2, but gfp->mode = 3 (mono), the L & R channels
 * will be averaged into the L channel before encoding only the L channel
 * This will overwrite the data in buffer_l[] and buffer_r[].
 * 
*/

  Frame* audio_frame = new Frame(NULL, (int)(1.25 * SAMPLES + 7200));

  const short int     *buffer_l = audio_buffer[0];   // PCM data for left channel
  const short int     *buffer_r = audio_buffer[0];   // PCM data for right channel
  const int           nsamples = SAMPLES;      // number of samples per channel
  unsigned char*      mp3buf = audio_frame->data;        // pointer to encoded MP3 stream
  const int           mp3buf_size = audio_frame->size;   // number of valid octets in this

  int val = lame_encode_buffer(lamegf, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size);

  if(val < 0) {
    switch(val) {
    case -1:  // mp3buf was too small
      info->error("Lame encoding failed, mp3buf was too small.");
      break;
    case -2:  // malloc() problem
      info->error("Lame encoding failed, due to malloc() problem.");
      break;
    case -3:  // lame_init_params() not called
      info->error("Lame encoding failed, lame_init_params() not called.");
      break;
    case -4:  // psycho acoustic problems 
      info->error("Lame encoding failed, due to psycho acoustic problems.");
      break;
    default:
      info->error("Lame encoding failed, due to unknown error.");
      break;
    }
  }

  return audio_frame;
}