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author | Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> | 2016-12-13 00:35:12 +0100 |
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committer | Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com> | 2017-01-06 19:58:42 +0100 |
commit | d74c471a39db2919a0e1db9666df725cbcb83d86 (patch) | |
tree | 8b3db9648a1792a627ffe341f4347a4bdd9d968a | |
parent | 09905c412ddb2dc3dd24e5741f64a59c7610e8ea (diff) | |
download | ffmpeg-d74c471a39db2919a0e1db9666df725cbcb83d86.tar.gz |
omadec: fix overflows during bit rate calculation
Signed-off-by: Andreas Cadhalpun <Andreas.Cadhalpun@googlemail.com>
-rw-r--r-- | libavformat/omadec.c | 4 |
1 files changed, 2 insertions, 2 deletions
diff --git a/libavformat/omadec.c b/libavformat/omadec.c index 6e476dbf25..757ae5371a 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: |