[FFmpeg-devel] [PATCH 1/3] omadec: fix overflows during bit rate calculation
Paul B Mahol
onemda at gmail.com
Tue Dec 13 09:11:32 EET 2016
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?
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