Overflows when decoding DTS to wav with ffmpeg

[RRAH]

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

[tebasuna51]

Please upload a sample (select the problematic fragment with a hex editor) to see the problem.

[pandy]

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?


At first you can try to reduce level (and/or use double float).

@ffmpeg -i %1 -map 0:a:0 -c:a pcm_f64le -y %1.w64

or

@ffmpeg -i %1 -map 0:a:0 -c:a pcm_f64le -y %1.wav