[FFmpeg-devel] [PATCH 1/3] omadec: fix overflows during bit rate calculation

[Paul B Mahol]

On 12/13/16, Andreas Cadhalpun <andreas.cadhalpun at googlemail.com> wrote:
> Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun at googlemail.com>
> ---
>  libavformat/omadec.c | 4 ++--
>  1 file changed, 2 insertions(+), 2 deletions(-)
>
> diff --git a/libavformat/omadec.c b/libavformat/omadec.c
> index 6e476db..e7751d0 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 /
> 128;
>
>          /* 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 / 256;
>          avpriv_set_pts_info(st, 64, 1, samplerate);
>          break;
>      case OMA_CODECID_MP3:

Shouldn't using 8LL or similar be more future-proof?