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. Multimedia.cx seems to indicate this is possible (https://wiki.multimedia.cx/index.php/MPEG-4_Audio), 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?
Playing at half sample rate might be possible, but I've never seen anyone actually do it.
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.