Added some audiocode.

Moved libfame code out of mov_encoder

1 files changed, 158 insertions, 0 deletions

diff --git a/src/liblame_wrapper.cc b/src/liblame_wrapper.cc
new file mode 100644
index 0000000..751c41b
--- /dev/null
+++ b/src/liblame_wrapper.cc
@@ -0,0 +1,158 @@
+/* -*- 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;
+}