I know this isn't exactly the place to ask, but Sound Canvas VA doesn't work in the latest MIDI plug in for Foobar. Gives me a Parameter File1 Read Error. Is there a way to fix this?
Ah silly me. Thanks for replying even though this isn't that type of thread.
EDIT: Uh, sorry again. I don't know where the install folder is. Where is it installed to by default? I can't seem to locate it in Program Files. Only a folder called Activator.
This is a stand-alone utility, designed to rip individual chip channels directly to .wav files, for mixing. Obviously, this is not usable for xSF formats, where it is next to useless, since most of the xSF formats employ dynamic channel allocation on playback, so individual sound channels get scattered across all the hardware channels in a seemingly random fashion.
Also, it obviously can't work for vgmstream. What you'd want for that is the Channel Mixer DSP, and downmix individual tracks with the channels you want, either at playback time, or convert them to some other format. It could probably use a settings file to allow describing downmixing settings for individual tracks, such as split a 6 channel track into 3 stereo subsongs.
That's another thing vgmstream needs: subsong support, for the few formats that may actually benefit from it. Maybe in the future...
There are also a few audio drivers for SNES that employ dynamic channel allocation on playback. I guess the best solution for these and most xSF formats would be tracking the usage of specific samples/instruments (or more bluntly soloing one sample/instrument after another) instead taking the hardware channels verbatim. Though I'm sure that's a big-ish change for most if not all players' playback code.
The numbers of times a song is looped is only relevant to VGM format and GYM format, not any of the others, which rely on manual timing tags to determine the length and fade times.
When editing the timing tags for SPC files using the foobar plugin - only whole second increments seem to be possible, not millisecond adjustments (which appear necessary to get accurate end times). Am I doing something wrong?
@Kode 15.99 logs the same as 15, only giving 15 seconds of audio. I can just set it longer and manually add cut/fade to the song. I likely wouldn't be able to find the exact value I need (not sure how other people have managed to do it)
@UltimateKoopa
You can make a batch file to do this. The one I made created a folder for each subsong and put the relevant files in each folder. It helps if you know the song names beforehand as you can have the batch file name the folders for you as opposed to having just numbers.
It would be nice to use the multidumper to set length, loops count, fade time of what gets logged
You could use Bleep! to log audio from .gbs files. This program allows you to choose song length and fade time. You could then write an autohotkey (.ahk) script that automatically logs each channel to WAV and puts them together into one folder for each song.
So, logging them one at a time? Isn't that what this tool was designed to avoid? Also, I'm not touching Bleep! because it's written in assembly language, the language of people who don't want other people to mess with their code.
Logging multiple files at once, as opposed to manually logging each one separately, isn't so much of an issue. But having them log for longer than 3 minutes can be a big issue.