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/* -*- 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.
 */
#include <config.h>
#include "liblame_wrapper.h"
#include "miav_config.h"

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

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

	lame_set_in_samplerate(gfp, INPUT_SAMPLE_RATE);
	lame_set_out_samplerate(gfp, OUTPUT_SAMPLE_RATE);

 	lame_set_num_channels(gfp, CHANNELS);
  lame_set_num_samples(gfp, SAMPLES);

	lame_set_quality(gfp, config->readInt("mp3_quality"));
	lame_set_mode(gfp, STEREO);
	lame_set_brate(gfp, config->readInt("mp3_bitrate"));

  lame_set_strict_ISO(gfp, 1);

  // 1 = write a Xing VBR header frame.
  lame_set_bWriteVbrTag(gfp, 0);

  // Types of VBR.  default = vbr_off = CBR
  //  lame_set_VBR(gfp, vbr_rh);

  // VBR quality level.  0=highest  9=lowest
  //  lame_set_VBR_q(gfp, 6);
  
  lame_set_copyright(gfp, 0);       // is there a copyright on the encoded data?
  lame_set_original(gfp, 1);        // is the encoded data on the original media?
  lame_set_error_protection(gfp, 0);// add 2 byte CRC protection to each frame?
  lame_set_padding_type(gfp, PAD_NO); // PAD_NO, PAD_ALL, PAD_ADJUST, PAD_MAX_INDICATOR 
                                    // 0 = do not pad frames
                                    // 1 = always pad frames
                                    // 2 = adjust padding to satisfy bit rate
  lame_set_extension(gfp, 0);       // private extension bit


	if (lame_init_params(gfp) < 0) {
    info->error("LAME parameter initialization failed.");
    return;
  }

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

  // And now for the dv decoder!
  decoder = NULL;
}

LibLAMEWrapper::~LibLAMEWrapper()
{
  lame_close(gfp);

  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;
  }

  /**
   * Decode audio using libdv
   */
  dv_decode_full_audio( decoder, dvframe->data, audio_buffer );

  /**
   * 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:
   *
   * return code     number of bytes output in mp3buffer.  can be 0 
   *                 if return code = -1:  mp3buffer was too small
   *
   * 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[1];   // 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;
  val = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size);
  // val = lame_encode_mp3_frame(gfp, buffer_l, buffer_r, mp3buf, mp3buf_size);
  
  //  info->info("Framenr: %d", lame_get_frameNum(gfp));

  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;
    }
  }

  /**
   * OPTIONAL:
   * lame_encode_flush_nogap will flush the internal mp3 buffers and pad
   * the last frame with ancillary data so it is a complete mp3 frame.
   * 
   * 'mp3buf' should be at least 7200 bytes long
   * to hold all possible emitted data.
   *
   * After a call to this routine, the outputed mp3 data is complete, but
   * you may continue to encode new PCM samples and write future mp3 data
   * to a different file.  The two mp3 files will play back with no gaps
   * if they are concatenated together.
   *
   * This routine will NOT write id3v1 tags into the bitstream.
   *
   * return code = number of bytes output to mp3buf. Can be 0
   */

  int flush_sz = lame_encode_flush_nogap(gfp,    // global context handle
                                         mp3buf + val, // pointer to encoded MP3 stream
                                         mp3buf_size - val);  // number of valid octets in this stream

  audio_frame->size = val + flush_sz;

  // Bits pr. second
  audio_frame->bitrate = config->readInt("mp3_bitrate") * 1000;

  return audio_frame;
}