libhb: Fix issue with silence insertion when using FLAC encoder

When we insert silence to maintain sync, we typically need to insert about
70ms.  But the default flac frame duration is very large (>=96ms).  This is
causing us to overshoot the target and we immediately drop a frame.  Dropping
a frame undershoots and we insert a frame... ad infinitum.

This patch changes the behavior of silence insertion for non-passthru audio.
Instead of inserting full frames of silence, the last "frame" inserted may be
a partial.  This lets us insert a finer grained silence duration and eliminates
the problem with large flac frames.


git-svn-id: svn://svn.handbrake.fr/HandBrake/trunk@5188 b64f7644-9d1e-0410-96f1-a4d463321fa5

1 files changed, 20 insertions, 5 deletions

diff --git a/libhb/sync.c b/libhb/sync.c
index 5ad5bfedb..444f2f98b 100644
--- a/libhb/sync.c
+++ b/libhb/sync.c
@@ -1162,7 +1162,7 @@ static void InsertSilence( hb_work_object_t * w, int64_t duration )
     hb_sync_audio_t *sync = &pv->type.audio;
     hb_buffer_t     *buf;
     hb_fifo_t       *fifo;
-    int frame_dur, frame_count;
+    int frame_dur;
 
     // to keep pass-thru and regular audio in sync we generate silence in
     // frame-sized units. If the silence duration isn't an integer multiple
@@ -1178,9 +1178,8 @@ static void InsertSilence( hb_work_object_t * w, int64_t duration )
         frame_dur = ( 90000 * w->audio->config.out.samples_per_frame ) /
                                             w->audio->config.in.samplerate;
     }
-    frame_count = ( duration + (frame_dur >> 1) ) / frame_dur;
 
-    while ( --frame_count >= 0 )
+    while (duration >= frame_dur >> 2)
     {
         if( w->audio->config.out.codec & HB_ACODEC_PASS_FLAG )
         {
@@ -1189,15 +1188,31 @@ static void InsertSilence( hb_work_object_t * w, int64_t duration )
             buf->s.stop  = buf->s.start + frame_dur;
             memcpy( buf->data, sync->silence_buf, buf->size );
             fifo = w->audio->priv.fifo_out;
+            duration -= frame_dur;
         }
         else
         {
-            buf = hb_buffer_init( sizeof( float ) * w->audio->config.out.samples_per_frame *
-                                  hb_mixdown_get_discrete_channel_count( w->audio->config.out.mixdown ) );
+            int channel_count = hb_mixdown_get_discrete_channel_count( w->audio->config.out.mixdown );
+            int size = sizeof( float ) *
+                       w->audio->config.out.samples_per_frame *
+                       channel_count;
+            if (frame_dur > duration)
+            {
+                int samples = duration * w->audio->config.in.samplerate / 90000;
+                if (samples == 0)
+                {
+                    break;
+                }
+                size = sizeof(float) * samples * channel_count;
+                frame_dur = (90000 * samples) / w->audio->config.in.samplerate;
+            }
+            buf = hb_buffer_init(size);
             buf->s.start = sync->next_start;
+            buf->s.duration = frame_dur;
             buf->s.stop  = buf->s.start + frame_dur;
             memset( buf->data, 0, buf->size );
             fifo = w->audio->priv.fifo_sync;
+            duration -= frame_dur;
         }
         buf = OutputAudioFrame( w->audio, buf, sync );
         hb_fifo_push( fifo, buf );