Posts by WillB

    Have you ever heard of anyone having a bad audio clock?

    I am clueless on the subject but I vaguely remembered reading an article on the subject a few years ago and I found it:…need-digital-master-clock

    Hence the question I posed, per the Per post, bringing the subject up initially earlier in this thread.

    RME manual:
    But word clock is not only the 'great problem solver', it also has some disadvantages. The word clock is based on a fraction of the really needed clock. For example SPDIF: 44.1 kHz word clock (a simple square wave signal) has to be multiplied by 256 inside the device using a special PLL (to about 11.2 MHz). This signal then replaces the one from the quartz crystal. Big disadvantage: because of the high multiplication factor the reconstructed clock will have great deviations called jitter. The jitter of a word clock is typically 15 times higher as when using a quartz based clock.

    The end of these problems should have been the so called Superclock, which uses 256 times the word clock frequency. This equals the internal quartz frequency, so no PLL for multiplying is needed and the clock can be used directly. But reality was different, the Superclock proved to be much more critical than word clock. A square wave signal of 11 MHz distributed to several devices - this simply means to fight with high frequency technology. Reflections, cable quality, capacitive loads - at 44.1 kHz these factors may be ignored, at 11 MHz they are the end of the clock network. Additionally it was found that a PLL not only generates jitter, but also rejects disturbances. The slow PLL works like a filter for induced and modulated frequencies above several kHz. As the Superclock is used without any filtering such a kind of jitter and noise suppression is missing.

    The actual end of these problems is offered by the SteadyClock technology of the Fireface UFX. Combining the advantages of modern and fastest digital technology with analog filter techniques, re-gaining a low jitter clock signal of 22 MHz from a slow word clock of 44.1 kHz is no problem anymore. Additionally, jitter on the input signal is highly rejected, so that even in real world usage the re-gained clock signal is of highest quality.




    "Most signals used by EBU Members in studios are expected to be sampled at 48 kHz. However 44.1 kHz may be used instead for some applications, such as recordings intended as masters for CDs. If signals are to be fed to transmission equipment, the frequency used may be 32 kHz."

    I don't think anyone is saying here that the SP-DIF is necessarily better or worse or even different sounding at 44.1 or 48k.
    The discussion from my perspective is your overall project sample rate. Since it's necessary to use the KPA as the master clock in order to even use the SP-DIF out some would say they couldn't use SP-DIF in the past due to it's previous 44.1k clock only and they were using rates higher than that.

    Now that has changed and your KPA can now provide a master clock setting up to 96k. All great news.

    My initial point was that if you are recording acoustic instruments in an acoustical space (that is contributing to the overall sound and it's perspective) then a rate higher than 44.1 is desirable in capturing the overall musical soundfield.

    If your project consists of 12 tracks of KPA and Steve Slate drum samples then 44.1k is probably "good enough".

    One of the statements by Per (on page two?) mentioned clock quality. Since the KPA clock is now running the show for some using SP-DIF how is the KPA clock quality?

    I'm using the KPA at 48k as master clock. I thought I saw 88.2 & 96k as an option already.

    I've thought about making a statement such as "if you can't hear the difference between 44.1k and XXk...then you..."
    But I haven't said that. I've tried to express my hearing results diplomatically.

    If KPA supports higher rates there really is no need to continue to defend 44.1 as the - all that is necessary - bullet point.

    The engineer was referring to bandwidth in an analog(ue) circuit.

    Dear Dahla,

    I mentioned those specific instruments ("Golden age" and "War era" Martins) not because they are old. I specifically mentioned that era and manufacturer, somewhat tongue in cheek, because the general consensus among musicians and luthiers is that those instruments represent some of the finest instruments ever made due to their materials, design, and the craftsman/artists who created them. I thought about using the example "Stradivarius violin" in my somewhat sarcastic example just so people with the "old" hang-up might give it a pass. I didn't. You didn't.

    For your benefit I've attempted to conform. How's this?

    ""Take a quality acoustic guitar (preferably it was made very recently with eco-friendly sustainable materials by a fully automated computer operated machine) that has responsive attack (impulse) and rich overtones and put it and the player in a nice sounding room.

    Set up a high quality (Shure SM-57 etc) pair of microphones in a stereo configuration such as “crossed figure eights at 90 degrees to one another”; cardioid or super cardioid at 120 degrees; figure eight and cardioid in the MS configuration.

    Record the performance at 44.1k 24 bit and at 48k 24 bit. Listen to the playback of each sample rate version over properly setup quality speakers or even quality headphones.""

    You said, "with a perfected list set up like a recipe..."

    If you read and comprehend what I actually said you will see I used the word "compromise" twice.
    Everything is a case you haven't noticed yet.

    Did I miss something?

    You said, "I suggest an easier test..."

    Well, I don't see your suggestion.

    Please tell me it isn't the "if is" quote.

    Note: To skip the miscellaneous spewing just jump to the last paragraph.

    In the pre-digital days of analog I embarked (with my colleague) on a challenge of attempting to capture the sound field of acoustic instruments or group of instruments (piano, harp, violin, viola, cello, double bass etc) in an acoustic space (room, hall, church etc) and reproducing that performance later.

    Studying the research of Alan Blumlein and others in the field of psychoacoustic and the reproduction of recorded sound using two loudspeakers (stereo) we experimented with various microphone techniques, equipment required to capture the sound, and then reproducing the performance in its acoustical environment . Critical listening of the transducers, electronics and microphone techniques led us to the conclusion that within the limitations of “stereo”, the Blumlein (X-Y) and/or Mid-Side (M-S) were the best compromises. In other words no mono multi-micing and pan pots.

    Determining the least compromised recording chain (microphone, mic preamp, storage medium etc) required extensive listening tests.
    Through a audio equipment review publication I had access to a vast number (fifty or more) of preamplifiers and power amplifiers and in a controlled listening environment these circuit designs were critically evaluated through extensive listening and measuring tests.

    The conclusion was that a properly designed circuit (preamplifier and amplifier) with “wide bandwidth” offered the most “accurate” ability to pass through and amplify a signal and convincingly reproduce the original performance in its acoustic space. Textures, clarity, attacks, overtones, imaging left to right, perceived depth and room acoustics were more convincing or “accurate” with wide bandwidth designed electronics.

    There are obviously a multitude of other factors such as negative feedback, transformers etc. that I won’t go into here. By the way, the recording chain that I settled on was completely tranformerless from microphone to storage medium. The end user/listener of the music of course used a 33 1/3 rpm phonograph disc so it was necessary to test that part of the chain via listening to the mastering of the disc process (half-speed mastering etc) and the vinyl itself (TELDEC ie Telefunken Decca virgin “pure”vinyl was vastly superior to anything. Hold it up to a light bulb and you can see light through it).

    A very talented electronics designer once told me that in order to accurately reproduce audio within the human hearing range requires a circuit bandwidth of ten times the highest reproduced frequency (20kHz x 10 = 200kHz). The same applies going below 20Hz. Although we can hear up to 20kHz why go beyond? Because in order to accurately reproduce up in that range the circuit bandwidth must exceed 20kHZ significantly. The engineer/designer words, not mine.

    Pulse tests, square wave test etc bear out his theory. Feed something in and watch if the circuit can accurately reproduce the impulse without overshoot and settle down again requires bandwidth. What does a 10kHZ square wave look like on the leading edge? Did not matter what the circuit topology was/is ie. transistor, IC, tubes, FET etc.

    Moving forward in time to digital capture and reproduce and try this simple experiment:

    Take a quality acoustic guitar (preferably a Martin made in the mid 1930s through 1944) that has responsive attack (impulse) and rich overtones and put it and the player in a nice sounding room.

    Set up a high quality (Schoeps etc) condenser pair of microphones in a stereo configuration such as “crossed figure eights at 90 degrees to one another”; cardioid or super cardioid at 120 degrees; figure eight and cardioid in the MS configuration.

    Record the performance at 44.1k 24 bit and at 48k 24 bit. Listen to the playback of each sample rate version over properly setup quality speakers or even quality headphones.

    What does this have to do with an electric guitar plugged into a limited bandwidth tube amplifier fed into a transformer and then into an even more limited bandwidth speaker (or that same signal chain only substitute the KPA after the electric guitar)?

    Accuracy is not a goal with most sound recordings. The Neumann U-47 or M-49 are considered by some to be the “best” vocal mics ever made and they were made sixty to seventy years ago. They really are not from a technical specification accurate but their euphonic colorations offer a desirable “quality” that many preferred both back in the days of Frank Sinatra and the Beatles through today even when thrown into the final end user’s listening experience ie. 128k (or less) lossy Mp3 and lastly the “in ear” “headphones”.

    Long way of saying:
    The KPA now allowing 48k (or higher) as the external master clock is a benefit to me because other acoustic instruments and vocals in the 48k 24 bit DAW project can/will sound better at the higher rate and the SPDIF out sounds better to my ears.

    Having listened carefully to KPA SPDIF to DAW at 44.1k in the past and now at 48k (all my projects are 48k) I don't hear a difference but since I find the SPDIF out sounds better I am grateful I can finally use it exclusively.
    I just spent the morning using KPA SPDIF to Cubase and had a blast.
    Thanks again.

    I always record 48k 24b.
    I've done this subjective listening test a few times before, comparing SP-DIF vs Analog Out when 44.1 was the only option.
    I tried it again today now with 48k available.
    Clean Amp and Stratocaster and Simultaneous Recording and volume matched:
    Track 1=SPDIF out.
    Track 2=Analog out.
    RME UFX to two Cubase 9.5.20 tracks.
    The results I'm hearing are the same as I've previously found.
    SP-DIF has less blossoming lower notes of guitar and more extended top-end without the subtle but present raspy, harshness that the analog track has.
    Blind A-B testing/track switching and it is easy to pick which is which consistently.
    I'm not making a definitive statement. Obviously the convertors in the UFX are a factor.

    Thanks Team Kemper

    Today while updating the latest beta OS via Rig Manager to my KPA I encountered a (what I thought was a serious) problem. After trying various things I finally capitulated and filled out a Kemper Service ticket explaining my non-functioning KPA's issue.

    Within forty-five minutes I was carrying out the suggestions that Hans-Jörg Scheffler from the Kemper Amps Support Team sent to me and I was then back up and running.

    Thank you for excellent support.

    I'm not sure if this was the cure but here is what I just did to get it installed:

    I had Rig Exchange Unticked (OFF). Turned it on and then inputted and tested credentials. Credentials good but then RM crashed.

    Restarted RM and retested credentials with RMExchange now on. Then checked for update via HELP/Software Update. Seemed like nothing happened so I looked "behind" RM Windows. Update Window was there and 4.2 Beta shown as available, clicked install. Currently running through the last step of checking rigs on my KPA so I am assuming it installed.

    And yes, it did the install and now, of course, the new pure cab (2) function is in CAB Module. Also, I doubt if turning on Rig Exchange had anything to do with it. Probably just putting in credentials which I had in there previous version of RM but were blank when I updated to this new beta version. After update the Tuner initially didn't work but reboot fixed it.

    If it was downloaded by RM, you'll find it in a folder located in the same parent folder where it stores your Rigs etc.

    It didn't download it I guess because I checked where you suggested and I did a search of the whole drive for kaos.bin.

    Also, in RM when I check for software updates from the HELP tab the response is:
    "Rig Manager and Profiler Software Update: This software is up-to-date."
    In Preferences I have Auto check for software updates and beta releases TICKED.