[FFmpeg-cvslog] omadec: fix overflows during bit rate calculation

[Andreas Cadhalpun]

ffmpeg | branch: master | Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com> | Tue Dec 13 00:35:12 2016 +0100| [d74c471a39db2919a0e1db9666df725cbcb83d86] | committer: Andreas Cadhalpun

omadec: fix overflows during bit rate calculation

Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>

> http://git.videolan.org/gitweb.cgi/ffmpeg.git/?a=commit;h=d74c471a39db2919a0e1db9666df725cbcb83d86
---

 libavformat/omadec.c | 4 ++--
 1 file changed, 2 insertions(+), 2 deletions(-)

diff --git a/libavformat/omadec.c b/libavformat/omadec.c
index 6e476db..757ae53 100644
--- a/libavformat/omadec.c
+++ b/libavformat/omadec.c
@@ -365,7 +365,7 @@ static int oma_read_header(AVFormatContext *s)
         st->codecpar->channels    = 2;
         st->codecpar->channel_layout = AV_CH_LAYOUT_STEREO;
         st->codecpar->sample_rate = samplerate;
-        st->codecpar->bit_rate    = st->codecpar->sample_rate * framesize * 8 / 1024;
+        st->codecpar->bit_rate    = st->codecpar->sample_rate * framesize / (1024 / 8);
 
         /* fake the ATRAC3 extradata
          * (wav format, makes stream copy to wav work) */
@@ -398,7 +398,7 @@ static int oma_read_header(AVFormatContext *s)
             return AVERROR_INVALIDDATA;
         }
         st->codecpar->sample_rate = samplerate;
-        st->codecpar->bit_rate    = samplerate * framesize * 8 / 2048;
+        st->codecpar->bit_rate    = samplerate * framesize / (2048 / 8);
         avpriv_set_pts_info(st, 64, 1, samplerate);
         break;
     case OMA_CODECID_MP3: