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AAC - General / "Lossless" AAC sample rate adjust?
Last post by hornpipe2 -
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?
3rd Party Plugins - (fb2k) / Re: AdPlug
Last post by mudlord -
Which means that adplug may have got updated.


Who needs to know?
3rd Party Plugins - (fb2k) / Re: AdPlug
Last post by deus-ex -
For one, most users are not software developers themselves and most probably would not know their way around that. Secondly, in order for this suggestion to work the official foo_input_adplug git needs to be up to date to begin with. At the time of this writing the last update is from 11 month ago.
3rd Party Plugins - (fb2k) / Re: OpenMPT Module Decoder
Last post by deus-ex -
I was asked to rename my component, because they got pissed off that the auto updater replaced their component with mine, especially since it's based on "unstable" "pre-release" code. I tend to think that bleeding edge is sufficient for daily use, but will be force-deprecating these 0.40-pre plugins when a new stable release appears.
Are you telling us you wouldn't be pissed off if it was the other way around? I was under the impression that this is a amicable coop between you and the OpenMPT developers, but apparently that assumption was wrong. If so I recommend you to take a step back and rethink your point of view.
you could use kode54's normalizer R128 DSP which applies normalization/ReplayGain in a non destructive manner in realtime on files, without them being tagged as such. You can even do so on Internet streams or emulated content.
3rd Party Plugins - (fb2k) / Re: AdPlug
Last post by mudlord -
If people really want logs, why not look at tagged git releases?
3rd Party Plugins - (fb2k) / Re: OpenMPT Module Decoder
Last post by mudlord -
17:02 <+mudlord> i found a tagging bug
17:02 <+mudlord> with titleformat syntax
17:03 <+mudlord> i thought anything in [ ] is to be cleared if its not present
17:03 <+mudlord> so adding a - in there makes the - present even when it shouldnt be
17:05 <+mudlord> no
17:05 <+mudlord> its a openmpt bug

Simple repro : [%artist% -] %title%
The "-" is present when that field should be completely NULL.
3rd Party Plugins - (fb2k) / Re: AdPlug
Last post by deus-ex -
If you re-read your own reply, maybe you will notice how underwhelming and narrow minded it is.
Because then emulated formats will suffer due to needing to decode up to that point.

I tested the current component and it works great on USFs. Though my CPU is a i7 4790.
thank you!