1

has been asked before, I am still unsure.
what is the best sample format (mono/stereo, sampling rate) in terms of

  • fast transmission via sysex
  • disk space on AR

thanks.

2

The lower the sampling rate, the smaller the file, the faster the transmission, but the worse the quality.

From what I understand the Rytm will perform sample rate conversion on any sample thats not 48K, as that is what the Rytm uses internally.

3

From the Analog Rytm FAQ:

The Analog Rytm uses 16 bit/48kHz/mono samples. Samples with other bit depths and sample rates are possible to use, but will be internally converted to 16 bit/48kHz.

So this means that anything higher/better then 16 bit/48kHz/mono is a waste. It’s just gonna take longer time to transfer.

Saving +Drive space by using lower quality samples is an interesting idea but it all comes down to the sound quality just like kwtsh said. In some cases samples sound “better” when they have lower bit and sample rate but that boils down to individual samples I would say.

There was some speculation that the +Drive keeps the original format and size but that’s not the case. It’s obvious now that the sample folders show the file size. For example a 24bit/48kHz/stereo sample that’s 1.44mb in size on the computer turns into a 0.45mb file on the +Drive.

4

The Rytm’s native format is 16bit 48khz. Anything else will be converted by C6, and because most software leaves artifacts doing this I would recommend getting a good sample rate converter. Check out http://src.infinitewave.ca/ for a list of comparisons. Audacity is good, especially because it’s free but it looks like the best sound quality wise are Adobe Audition and Izotope SRC which is in products like Sound Forge Pro and Sample Manager.

5

I don’t have any problems loading in 16 bit/44.1 khz mono. If it works, it works. :slight_smile:

6

I’d recommend chilling out about this. Even though the current sample dump standard is insanely slow, the RYTM seems to do a decent job at sample rate conversion.

Sure, there might be tools who’ll do a slightly better job, and you might get a slightly better result converting to mono, 16bit, 48kHz before transfer, but the reality is that you won’t hear ANY difference once you use your samples in the context of whatever it is you’re producing, especially if you start using layering and the filter and the distortion and whatever other kind of sound-sculping stuff this machine has to offer.

Seriously, the only time you should maybe start worrying about sound quality or introducing artifacts is when you’re converting a final mix between different bit depths or sample rates.

7

thanks for the explanation, very helpful.
i did experience conversion artefacts once. got a noise hiss on a 808 kick that was not audible via the computer. dont remember the original sample bit & sample rate.

8

I have written a python script to slice up audio files (in batch if so desired) for the Analog Elektron Rytm (or any other thing that can play audio files).

It is available for free download here:

This python script slices audio files (in batch if so desired) along the length of the file until it reaches the end of the file.
By default it will slice a file into 2-second blocks, with each block starting at the end of the next block, and each block output as a separate file (into the folder containing the input file; file output names as per input but with the position in the original file added to the output file name). The default format of the output slices is 16-bit, 48kHz, mono. The user can crush the sample to 8-bit width or have it in medium (16-bit) or high-quality (32-bit). Sample rate can be anywhere from low quality (11025 Hz) to high quality (48000 Hz) – in fact, sample rate can be whatever you want, but your computer may not know how to deal those non-standard rates (e.g., I tested it with 1 Hz, and iTunes died when trying to play it – see the help menu for standard/accepted options [python SliceAudio.py -h] ). The user can also alter the sample slice length and the overlap slide on the previous slice (e.g., you could slice into 10 second windows with each subsequent window sliding along 1 second to overlap the previous window by 1 second. NB. time is measured in milliseconds, so multiply x-seconds by 1000 to get the desired slice length in seconds). There is an option for stereo output. The script can input and output any format that is supported by ffmpeg**.

Dependencies:

  1. gcc
  2. pydub (sudo pip install pydub), see https://github.com/jiaaro/pydub
  3. ffmpeg (brew install libav --with-libvorbis --with-sdl --with-theora)
  4. audioread (sudo pip install audioread)

NB: the 16bit/mono/44.1kHz sample rate was selected to be compatible with the Analog Elektron Rytm drum machine.

Example usage: python SliceAudio.py -i xyz.m4a -f m4a -b 2 -s 11025 -l 10000 python SliceAudio.py -h

**ffmpeg formats: [https://trac.ffmpeg.org/wiki/audio%20types](https://trac.ffmpeg.org/wiki/audio types)

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