I'm not sure if this is the right category...I'm thinking about buying a used Octamic II which is 10 years old and in good condition (for half the price of a new one).

My question is, if it's still worth it or if relevant aspects (like the preamps or converters) has been significantly improved the last 10 years. (I would like to record drums, if this information is important...)

Thanks for your help!

Nothing I tried would force it to remember a sample rate that was:

a) useable
b) one that I wanted to use.

So I had Anitgravity write me a command line utility to set the sample rate so that I can run that once my Mac has started!

I'm sharing it in case this might be useful for someone else having similar issues.

Download here: https://drive.google.com/file/d/12-smE- … sp=sharing

It's a Univeral Binary so should run on Intel/Apple Silicon (I've only tested on my Intel Mac Pro).

It's signed and notarised, but it's not possible to staple a raw binary, so you will probably have to jump through some hoops to remove quarantine bit after downloading (ask your favourite LLM is you don't know how).

It's provided AS IS, and I offer no warranty and absolve myself of all liability if you choose to run it.  YOU USE IT AT YOUR OWN RISK.

That said it works perfectly for me and I promise it doesn't install a crypto-miner on your machine!

I cannot offer support, and I will not update it, since it does the job for me.  You can prompt Antigravity yourself if you don't trust this or need different features.

Usage:

$ ./set_rme_sample_rate
Usage: set_rme_sample_rate <DeviceName> <SampleRate>
Example: set_rme_sample_rate "Digiface USB" 44100

It may well work for other RME devices, but I cannot test that myself, and as mentioned you run this at your own risk.  If your computer or RME dies as a result of running it, that's not my problem (but honestly I think that isn't a major thing to worry about happening).

To have it run when your Mac boots, ask your favourite LLM for instructions if you don't already know how.

Enjoy.

Now,....imagine if they were one machine!

At the moment, when using the BFace, I hardly ever use the physical buttons on the unit. I use the ARC for Talkback, volume, and especially having access to the Snapshots is fantastic. (I am a TV/Film composer and find being able to switch to Music, Dialogue, FX tracks very helpful)

Just saying,....imagine if the ARC had an output and perhaps ADAT in and out too.

I'm a long-time RME user (UFX, XTC, Babyface Pro FS) and I've been building a new audio mixer app. I'm looking for testers — particularly those with higher channel-count interfaces like the UFX+ or Fireface 802 — but all RME hardware welcome.

FusionDesk is a TotalMix FX inspired desktop mixer for Windows 10/11:

• Up to 32 x 32 channel input/output routing 
• VST3 hosting 
• MP3, FLAC & WAV linear and loop recording up to 32-bit float 
• Supports ASIO, WASAPI, Per-Application Audio Capture, Dante, ASIO4ALL

It's pre-release but stable enough for real sessions. What I really want to know is whether it holds up under actual use.

Specifically: 
- Try to break it — unusual routing, edge cases, heavy VST loads 
- Tell me what feels wrong or unintuitive 
- Tell me what's missing that you'd actually want to use 
- Any comments or suggestions are very welcome

If you're on W11, W10 feedback is also valuable — WASAPI behaviour can differ and I want to cover both.

If you're up for it, DM me or drop an email and I'll send you a build.

Cheers, 
Richard 
Testing@AudioArt.uk

I know you can get Thunderbolt chassis units from Sonnet and have had them in the past in my iMac Pro days, but as someone with 3 AES cards (one also using the daughter card) and an AIO Pro card in my Mac Pro M2 Ultra (and loving it aside from the age), I am not looking forward to moving to a Mac Studio and multiple external units to get back what I already have with the Mac Pro in terms of PCIe connectivity built right in and no, I am not willing to use a PC and Windows.

It would be really cool to see RME make a dedicated Thunderbolt 5 unit with the same AES I/O that the AES card has now that Apple seems to be forever done with making computers with PCIe slots.

UFX 3  ( USB 3.0 ) Madi w/ ARC- Ferrofish 16 Madi
Newer ASUS 14G Laptop Windows 11
Reaper 7 newest version

TIA

I recently bought a used ADI-2 Pro FS R BE and began to convert my vinyl collection into digital formats.

In short my chain:

turntable -> phono preamp (MC load 100Ω) -> converter -> mac over usb (Ref Level for converting is set to +13dBu):

After the conversion of some new LP’s and normalizing at -23dB target level I noticed a slight hiss on passages with silence after one or two seconds.

A good example is the song „Waiting Room“ from FUGAZI or „La Grange“ from ZZ Top at the beginning where only the sticks play.

I’ve noted down some noise levels from the analog input indicated on the display of the converter in USB mode.

converter alone : -102dB

phone preamp & converter: -77dB

turntable & phone preamp & converter: between -70dB and -65dB

Before I go over with manipulation on the DAW:

Are these levels good enough considering the hardware only?

Is there a reasonable way to bring down the noise below -70dB or am I chasing unicorns?

Maybe you know more.

Thanks in advance

Dynamys

The channel balance control (often just called the "balance" knob) on a hi-fi amplifier adjusts the relative volume between the left and right stereo channels.
What it actually does

At the center position (usually with a detent/click), both left and right channels output at equal volume.
Turning it toward the left increases the left channel's level while decreasing the right channel's (or vice versa when turning right).
It does not boost the overall volume or add power — it simply attenuates (turns down) one channel relative to the other to restore proper left-right balance at your listening position.

This keeps the total perceived loudness roughly similar while shifting the stereo image (the phantom center and soundstage) left or right.

Thank you

The generic recording signal chain modeled in this application includes:

    microphone →
       optional attenuator →
           optional preamplifier →
               optional attenuator → 
                  RME Mic-Pre A/D converter

You'd first use a pulldown menu to select the genre of what you wish to record.  The tool will optionally provide researched typical values for the ranges of SPL levels expected at the microphones, desired headroom, and the THD quality standard.  Once you also enter your venue's noise floor, you're all set to optimize gains to achieve one of three possible goals:

Maximum SNR       Sets gain as high as possible within your headroom target.  Best for quiet sources (choral, chamber,
                           bioacoustics) where the noise floor is the primary concern.

Minimum THD        Sets gain as low as possible while achieving the active THD quality standard and maintaining at least 40 dB of
                            usable SNR. Targets the specific THD level you have selected rather than simply minimizing gain.

Genre balance       Uses your selected recording genre to set a minimum acceptable SNR floor, then finds the lowest gain
                            satisfying that floor. Balances noise and distortion for your specific application.
                            Also grades the result against your selected THD quality standard.

You can access the web tool at my GitHub site, here:
   https://herbarcher3.github.io/gain-stag … r_v21.html

A detailed user manual is available here:
   https://herbarcher3.github.io/gain-stag … er_v21.pdf

This tool is under active development. Specifications are sourced from manufacturer datasheets, published measurements, and topology-based estimates. Results are intended as as a notional check on your own experienced setup plans for the recordings you do.  You would, of course, want to rely on your own ears and meter observations to refine these settings if needed during a performance.

Is it possible to achieve these for this much money?

Which brings me to the question:

Are wired AirPods Max 2 useful for music production or not (delay)?

Is seems they have USB cable support now https://www.youtube.com/watch?v=iU-RBiW7EYc

What is the fellow RME musician perspective on it? Are they suitable for semipro, enthusiasts or not?

Reason why they are interesting compared to other headphones is:
- they are widley used, so people are looking to have their music play well on AirPods Max sound profile
- can be used for other stuff in daily life

https://www.roundhouse.org.uk/whats-on/sigrid/

Going to lend her my Baby Face Pro