Meh, not convinced Thanks for the explanation, stopping to convert feels like a break in work flow which I'm trying to avoid, although having the UI do it automatically sounds like a reasonable approach. I will look into it, I'm not bothered about inefficient block starts if the file is loaded and plays correctly, I would be happy with waiting for a little bit of parsing to get the offsets as long as playback performance isn't impacted.
Play FN and clicking
In some cases it might work. But me personally I like to use the string/index object to have as many different loops available when the patch is live. Ilike to switch between them when the patch is live. For this Axoloti would not know which files I am loading after the patch is made live, so in this scenario converting on the go wont work. Also all loops I make has to fit perfectly in table, so for this I need to do a lot of "manual labour" on the files anyway, like sample rate convert, stretch files to fit table etc. And for doing these things I have to use an audio editor. The only difference in the workflow is exporting the file as raw instead of wav. Not an issue here.
But sure, if you have an idea how to code your way around the header click, I think everyone would be interested and welcome it
(Also would be awesome to remove the "BLLLLLIP" when changing content of a table when a patch is live. To me that is a bigger deal than the raw files. So if you have an idea how to avoid that, it would be most welcome)
I haven't started using samples properly on the axoloti yet, so I don't have any use cases above what you'd generally expect as such but loading a sample should be a fairly seamless operation for the average end user.
Some of you guys will no doubt have your own work flows and methods for the ways you're doing things, maybe they'll seem more sensible when I start to use them myself.
I agree Maybe someone will come up with an easier way later on, but right now, using raw files is the best practice, both for wav player and tables.
Yeah there are many different ways of converting files to Axoloti standard. Personally I use Audacity, for stretching files to fit table, ยดcause you can do it really precisely, down to sample precision. Also for raw conversion, sample rate conversion etc. And it is free
I mostly uses Axoloti for samples, sampling etc. It is capable of some pretty awesome stuff. You'll have a lot of fun with it
Sounds like you know what you're doing, I will be asking lots of questions at some point in the near future
I've been having lots of fun with the axoloti already, even the simplest things are fantastic to play with. I made a bass synth and I woke my neighbour up by accident the other night, he said 'Can you hear that?', so I said 'hear what?', he said 'That buzzing eletrical noise, sounds like something big and electric is going to explode...', so I said 'OOOOOh, You mean this noise?' as I pressed a key on the keyboard... WUB!!
Hello! I'm trying to convert .WAV files to .RAW files for WAVE/PLAY. Could you give an example of a ffmpeg
command that converts the files correctly? I'm a little confused if I should be using big-endian / little-endian etc.
For example, say I have a 16 bit 48khz WAV named test.wav
.
ffmpeg -i test.wave .... ?
Thanks!
Sorry I cant give you an example of how to do it in coding.
But you can do it with Audacity.
And you can batch convert to raw files in a program called Switch. I use that a lot. Very easy to use and pretty cheap too
Thanks for the follow up @jaffasplaffa
I was able to convert 48k 16bit WAV -> RAW with the following command.
ffmpeg -i "033.wav" -f s16le "033.raw"
.
You can put that in a bash file and convert many files really quickly!
#!/bin/bash
# turn all wav files into raw files
for file in *.wav
do
echo "$file"
#file without extension -> "${file%%.*}"
ffmpeg -i "${file}" -f s16le "${file%%.*}.raw"
done