*** empty log message ***

1 files changed, 51 insertions, 76 deletions

diff --git a/src/liblame_wrapper.cc b/src/liblame_wrapper.cc
index 7af9b68..90b8824 100644
--- a/src/liblame_wrapper.cc
+++ b/src/liblame_wrapper.cc
@@ -31,6 +31,9 @@
 
 /*
  * $Log$
+ * Revision 1.3  2005/07/07 12:42:19  deva
+ * *** empty log message ***
+ *
  * Revision 1.2  2005/07/05 23:15:16  deva
  * *** empty log message ***
  *
@@ -62,6 +65,18 @@ LibLAMEWrapper::LibLAMEWrapper(Info *i)
 	lame_set_mode(gfp, STEREO);
 	lame_set_brate(gfp, 192);
   lame_set_strict_ISO(gfp, 1);
+  lame_set_num_samples(gfp, SAMPLES);
+
+  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
+
+  info->info("Lame version %d", lame_get_version(gfp));
 
 	if (lame_init_params(gfp) < 0) {
     info->error("LAME parameter initialization failed.");
@@ -72,8 +87,6 @@ LibLAMEWrapper::LibLAMEWrapper(Info *i)
 
   audio_buffer[0] = new int16_t[AUDIO_BUFFER_SIZE];
   audio_buffer[1] = new int16_t[AUDIO_BUFFER_SIZE];
-  audio_buffer[2] = new int16_t[AUDIO_BUFFER_SIZE];
-  audio_buffer[3] = new int16_t[AUDIO_BUFFER_SIZE];
 
   // And now for the dv decoder!
   decoder = NULL;
@@ -85,8 +98,6 @@ LibLAMEWrapper::~LibLAMEWrapper()
 
   delete audio_buffer[0];
   delete audio_buffer[1];
-  delete audio_buffer[2];
-  delete audio_buffer[3];
 }
 
 Frame *LibLAMEWrapper::encode(Frame *dvframe)
@@ -104,66 +115,53 @@ Frame *LibLAMEWrapper::encode(Frame *dvframe)
     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 );
-  /*
-  memset(audio_buffer[0], 0, sizeof(audio_buffer[0]));
-  memset(audio_buffer[1], 0, sizeof(audio_buffer[1]));
-  */
-  /*
-    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:
- *
- * 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[].
- * 
-*/
 
+  /**
+   * 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 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 = lame_encode_buffer(gfp, buffer_l, buffer_r, nsamples, mp3buf, mp3buf_size);
+  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) {
@@ -185,29 +183,6 @@ Frame *LibLAMEWrapper::encode(Frame *dvframe)
     }
   }
 
-  /*
-  val = lame_encode_flush(gfp, mp3buf, val);
-
-  if(val < 0) {
-    switch(val) {
-    case -1:  // mp3buf was too small
-      info->error("Lame encoding failed (during flush), mp3buf was too small.");
-      break;
-    case -2:  // malloc() problem
-      info->error("Lame encoding failed (during flush), due to malloc() problem.");
-      break;
-    case -3:  // lame_init_params() not called
-      info->error("Lame encoding failed (during flush), lame_init_params() not called.");
-      break;
-    case -4:  // psycho acoustic problems 
-      info->error("Lame encoding failed (during flush), due to psycho acoustic problems.");
-      break;
-    default:
-      info->error("Lame encoding failed (during flush), due to unknown error.");
-      break;
-    }
-  }
-  */
   audio_frame->size = val;
 
   return audio_frame;