yuv4mpeg: Correctly round chroma up for odd luma sizes

[ffmpeg] / libavformat / audiointerleave.c

diff --git a/libavformat/audiointerleave.c b/libavformat/audiointerleave.c
index 3c235c069e7df59ecad081a4575edca4777829c3..e4cde9d0aab619a2db1aa6fe12e53fc0194c0bde 100644 (file)
--- a/libavformat/audiointerleave.c
+++ b/libavformat/audiointerleave.c
@@ -3,24 +3,25 @@
  *
  * Copyright (c) 2009 Baptiste Coudurier <baptiste dot coudurier at gmail dot com>
  *
- * This file is part of FFmpeg.
+ * This file is part of Libav.
  *
- * FFmpeg is free software; you can redistribute it and/or
+ * Libav is free software; you can redistribute it and/or
  * modify it under the terms of the GNU Lesser General Public
  * License as published by the Free Software Foundation; either
  * version 2.1 of the License, or (at your option) any later version.
  *
- * FFmpeg is distributed in the hope that it will be useful,
+ * Libav 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
  * Lesser General Public License for more details.
  *
  * You should have received a copy of the GNU Lesser General Public
- * License along with FFmpeg; if not, write to the Free Software
+ * License along with Libav; if not, write to the Free Software
  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
  */
 
 #include "libavutil/fifo.h"
+#include "libavutil/mathematics.h"
 #include "avformat.h"
 #include "audiointerleave.h"
 #include "internal.h"
@@ -69,8 +70,8 @@ int ff_audio_interleave_init(AVFormatContext *s,
     return 0;
 }
 
-static int ff_interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
-                                   int stream_index, int flush)
+static int interleave_new_audio_packet(AVFormatContext *s, AVPacket *pkt,
+                                       int stream_index, int flush)
 {
     AVStream *st = s->streams[stream_index];
     AudioInterleaveContext *aic = st->priv_data;
@@ -124,10 +125,10 @@ int ff_audio_rechunk_interleave(AVFormatContext *s, AVPacket *out, AVPacket *pkt
         AVStream *st = s->streams[i];
         if (st->codec->codec_type == AVMEDIA_TYPE_AUDIO) {
             AVPacket new_pkt;
-            while (ff_interleave_new_audio_packet(s, &new_pkt, i, flush))
+            while (interleave_new_audio_packet(s, &new_pkt, i, flush))
                 ff_interleave_add_packet(s, &new_pkt, compare_ts);
         }
     }
 
-    return get_packet(s, out, pkt, flush);
+    return get_packet(s, out, NULL, flush);
 }