X-Git-Url: http://git.osdn.jp/view?a=blobdiff_plain;f=libhb%2Fenclame.c;h=9692eef29e31a59511835583e270344ff58c7f96;hb=93b82f5fb7457e2e07dcf88e8d23e1739a9b0afe;hp=a3f4a388282233ef5ffb06757b3e69369f7386e2;hpb=48741d715b5fee3a4b232f01de78d9519513a617;p=handbrake-jp%2Fhandbrake-jp-git.git

diff --git a/libhb/enclame.c b/libhb/enclame.c
index a3f4a388..9692eef2 100644
--- a/libhb/enclame.c
+++ b/libhb/enclame.c
@@ -29,6 +29,7 @@ struct hb_work_private_s
     lame_global_flags * lame;
 
     int             done;
+    int             out_discrete_channels;
     unsigned long   input_samples;
     unsigned long   output_bytes;
     uint8_t       * buf;
@@ -41,6 +42,7 @@ int enclameInit( hb_work_object_t * w, hb_job_t * job )
 {
     hb_work_private_t * pv = calloc( 1, sizeof( hb_work_private_t ) );
     hb_audio_t * audio = w->audio;
+
     w->private_data = pv;
 
     pv->job   = job;
@@ -53,16 +55,25 @@ int enclameInit( hb_work_object_t * w, hb_job_t * job )
     lame_set_VBR_mean_bitrate_kbps( pv->lame, audio->config.out.bitrate );
     lame_set_in_samplerate( pv->lame, audio->config.out.samplerate );
     lame_set_out_samplerate( pv->lame, audio->config.out.samplerate );
-    lame_init_params( pv->lame );
+
+    pv->out_discrete_channels = HB_AMIXDOWN_GET_DISCRETE_CHANNEL_COUNT(audio->config.out.mixdown);
     // Lame's default encoding mode is JOINT_STEREO.  This subtracts signal
     // that is "common" to left and right (within some threshold) and encodes
     // it separately.  This improves quality at low bitrates, but hurts 
     // imaging (channel separation) at higher bitrates.  So if the bitrate
     // is suffeciently high, use regular STEREO mode.
-    if ( audio->config.out.bitrate >= 128 )
+    if ( pv->out_discrete_channels == 1 )
+    {
+        lame_set_mode( pv->lame, MONO );
+        lame_set_num_channels( pv->lame, 1 );
+    }
+    else if ( audio->config.out.bitrate >= 128 )
+    {
         lame_set_mode( pv->lame, STEREO );
+    }
+    lame_init_params( pv->lame );
 
-    pv->input_samples = 1152 * 2;
+    pv->input_samples = 1152 * pv->out_discrete_channels;
     pv->output_bytes = LAME_MAXMP3BUFFER;
     pv->buf  = malloc( pv->input_samples * sizeof( float ) );
 
@@ -98,9 +109,9 @@ static hb_buffer_t * Encode( hb_work_object_t * w )
     hb_work_private_t * pv = w->private_data;
     hb_audio_t * audio = w->audio;
     hb_buffer_t * buf;
-    int16_t samples_s16[1152 * 2];
+    float samples[2][1152];
     uint64_t pts, pos;
-	int      i;
+    int      i, j;
 
     if( hb_list_bytes( pv->list ) < pv->input_samples * sizeof( float ) )
     {
@@ -110,16 +121,22 @@ static hb_buffer_t * Encode( hb_work_object_t * w )
     hb_list_getbytes( pv->list, pv->buf, pv->input_samples * sizeof( float ),
                       &pts, &pos);
 
-    for( i = 0; i < 1152 * 2; i++ )
+    memset(samples, 0, 2*1152*sizeof(float));
+    for( i = 0; i < 1152; i++ )
     {
-        samples_s16[i] = ((float*) pv->buf)[i];
+        for( j = 0; j < pv->out_discrete_channels; j++ )
+        {
+            samples[j][i] = ((float *) pv->buf)[(pv->out_discrete_channels * i + j)];
+        }
     }
 
     buf        = hb_buffer_init( pv->output_bytes );
-    buf->start = pts + 90000 * pos / 2 / sizeof( float ) / audio->config.out.samplerate;
+    buf->start = pts + 90000 * pos / pv->out_discrete_channels / sizeof( float ) / audio->config.out.samplerate;
     buf->stop  = buf->start + 90000 * 1152 / audio->config.out.samplerate;
-    buf->size  = lame_encode_buffer_interleaved( pv->lame, samples_s16,
+    buf->size  = lame_encode_buffer_float( 
+            pv->lame, samples[0], samples[1],
             1152, buf->data, LAME_MAXMP3BUFFER );
+
     buf->frametype   = HB_FRAME_AUDIO;
 
     if( !buf->size )