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Topic: "Lossless" AAC sample rate adjust? (Read 5620 times) previous topic - next topic
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"Lossless" AAC sample rate adjust?

I have a camera that takes videos at high FPS, and I can later use ffmpeg to slow them down for visual effect.  For the video portion, it is possible to losslessly alter the frame rate - no transcoding necessary.  Change one number from 120 to 24 and it plays exactly the same, but Slower.

Now, for audio.  What I want is to slow / down-pitch the sound at the same rate to match.  This is doable in .wav format (just tamper with the header).  The camera shoots in AAC though.  A transcode to WAV is technically "lossless", but it's also going to blow up the filesize.  I can also re-encode back to AAC, but at a quality loss.

So, my question: Is it possible to "globally" modify some AAC header and alter its playback rate, without re-encoding?  The plan would be to extract the AAC stream, twidle a couple bits to change the playback rate, and then re-embed it. seems to indicate this is possible, but are there other considerations - e.g. tables that only work at some frequencies, or LZ compression that may refer back to this byte position, or whatever?


Re: "Lossless" AAC sample rate adjust?

Reply #2
Of the tests I've seen, transcoding with AAC isn't a perceptible problem; nothing at all like the wise internet parrots make it out to be.
Is 24-bit/192kHz good enough for your lo-fi vinyl, or do you need 32/384?

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