I came across this problem while converting hi-res *.wav tracks into 24/48 ALAC files. This is what I found: encoding an audio file into ALAC 24bit results in a reversed waveform polarity, which comes back to normal once converted back to *.wav, *.flac or even ALAC 16bit. I have been experimenting with both dBPowerAmp and DVD Audio Extractor. The software doesn't change the result.
Do you know what is causing this? I don't think it's normal.
Do you know what is causing this? I don't think it's normal.
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