32 kbps opus-tools-newtf.zip:
High frequencies especially synth with (fart like sound, I don't know what sample it is) I noticed a good improvement compared to Opus 1.3 Beta: the sound is less muddy, more clear.
But with this improvement I think vocal lost a bit of presence and detail (warmth). (I tested with a Rock track, not acapella)..
Sure, an optimization could further improve the quality.
Thanks for elaborating on this.
I started building my mp3 library almost ten years ago and the oldest files were last modified back then. It is stunning to be able to sort my files by these dates; mostly they correspond to the date I bought the CD and allow me to see the progress of my exploration of the classical repertoire. When I now start adding year tags to every file and they will all be last modified in 2018, there will be no way to get this sorting back. So I am very glad mp3tag allows me to keep the modification date intact.
3rd Party Plugins - (fb2k) / Re: Fixing changed filepaths in foobar2k: foo_playlist_revive is the wrong filetypeLast post by lvqcl -
The easiest way is to open File - Preferences - Components and drag-and-drop the .dll from explorer to "Installed Components" list.
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, 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?
Which means that adplug may have got updated.
Who needs to know?
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.
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.
If people really want logs, why not look at tagged git releases?
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.