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7th January 2015, 08:19 | #1 | Link |
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Join Date: Mar 2014
Posts: 13
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Overflows when decoding DTS to wav with ffmpeg
Hi,
when decoding lossy DTS to wav ffmpeg always produce overflows. The last file had a sample peak of 22,1db (!). Are these artifacts from the lossy process which I can ignore? What do you think about setting the advanced limiter when converting DTS with foobar2000 to avoid the problem? I want to encode to AC3 with Soundforge. Thanks |
13th January 2015, 10:31 | #3 | Link | ||
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