Werner Althaus

Constantin, I'd like to comment on that assessment but not before mentioning that I am not in favor of ultrahigh SRs like 192 or even 96KHz. yes it sounds different than 48 but not for the reasons proponents imagine, quite the contrary, it usually sounds worse. I'm saying this so you know that I'm not coming at this from some esoteric audiophile POV. The discussions about SRs at the dawn of digital mostly hinted at an optimum SR in 60 to 70KHz range but for practical reasons lower SRs were chosen. Now back to the hearing test: I think that going to an audiologist will tell you something about your hearing threshold at various frequencies but it doesn't tell the whole story. How do you think Bob Clearmountain or Bruce Swedien were able to mix great sounding records when their age would indicate an upper limit of 10K tops? Many top mixers are technically too old to hear "up there" but good luck trying to trick them by cutting off significantly below 20K. It's a feel thing and mixers learn to adjust to their diminishing hearing capabilities. Otherwise they couldn't do their jobs, something that George Martin understood when he really felt that it was no longer possible for him to do his work he enlisted help. I know this is outside of the realm of what this forum is about but so forgive the sidetrack.

Constantin said: “Just be prepared to encounter stronger than usual opposition and be accused of an anti Zaxcom conspiracy when you just hint a critique towards Zaxcom. Oh wait a minute, that already happened.” I guess I walked right into that one, LOL Jeff said: “….Regarding my recording of the output of my wireless receiver, the file will be 24-bit - 48K based on sound being transmitted from the transmitter with a frequency response of 20 to 16K. So, we are back to the all important LISTENING test. Is there a significant audio quality difference for our purposes between 20 to 16K and 20 to 20K?” Assuming that you used a lav with a frequency response of up to 20KHz I’d stand by my statement that the RF system doesn’t deliver the full response of the mic due to the SR but you’re right, you did faithfully capture the analog output at 48/24, I think we can agree on that. As a post mixer I might be wondering why that mic captured at 48/24 cuts off drastically at 15KHz but I wouldn’t loose sleep over it either. Does it matter? Depends on the type work you do. I do everything from OB, mixing live regional sports where fidelity isn’t much of a concern (those wireless parabs or other sideline wireless mics would be fine with TXs at 22KHz or less ) to music recording of all genres and of course loads of EFP. When I record and mix English language Opera where diction is all important ( no subtitles in the final program) I tend to use my DPA miniature mics on the principals to blend with the 5 to 8 Schoeps Supercardiods spread across the stage. This helps the audience to understand the lyrics to a greater degree. I would want the response of those DPA mics to be as good as a cabled connection both in terms of frequency response as well as dynamic range because I might have to rely on them exclusively should there be too much external noise from coughing audience members or footsteps or whatever, etc. been there, done that. So yes, I dare to say that it matters under certain circumstances. But let me again say that I agree with listening tests superseding any theoretical discussion. Thanks.

Thanks for asking, No, I didn't have the pleasure to try them either. Currently I use Lectro 200 and 400 series, Sennheiser evolution100, 300s, 500's, and my favs SK250's into 3000 series receiver. But for location work it's usually Lectros only.

Jeff said: "….. you are making this case to cast doubt on the performance of Zaxcom digital vs. the new Audio, Ltd. offering, plain and simple.” I’m sorry if I created that impression because nothing could be further from the truth. I admire the company and think they’re on the cutting edge of technology. I’ve never had the pleasure to use one in the field myself but from all I’ve heard, read and seen their products appear to be great. I just wanted to have a meaningful discussion about their decision to utilize a sample rate for the RF that differs from the standard 48KHz. What ensued was, with all due respect , a lot of vague responses that didn’t address my concerns, instead I got such wisdom as “nobody can hear 16KHz anyway, voices don’t need that kind of upper frequency extension, irrelevant when compared to real world issues such as......, etc”. All things being equal If I were to decide between 60 mics on a single TV channel at 32KHz or a lesser number at a 44.1 or 48KHz Sample rate then I would tend to prefer higher audio fidelity over channel-count unless I was convinced that there are no drawbacks to going the higher channel count route. I can imagine others feeling the same. It might even be conceivable that a manufacturer like Zaxcom lets the user decide between lower bandwidth, higher mic count per TV channel vs higher bandwidth, lower mic count per TV channel. Jeff said: “ In my workflow, and that of most everyone else, our recording is a true 24-bit 48K industry standard broadcast wave file and was NOT created by any sample rate conversion. Regarding my use of wireless microphones, when I take the output of the wireless receiver (whether it is an old Vega or a brand new pure digital unit from Zaxcom or Audio, Ltd.) I typically input that analog signal from the receiver at line level into the analog preamp of the recorder” Fine, I didn’t know whether you took an analog signal into your mixer or a digital signal, you got me there but that analog signal still has the audio frequency response of a 32KHz sample-rate if that’s what the digital RF signal is utilizing. What would you say if I claimed that SFXs recorded on a Zoom set to mp3 format were “true 24-bit 48K industry standard broadcast wave files” just because I plugged the Zoom’s analog output into my recorder set to those specs and hit record? That'd be a numbers game IMO. Jeff said: “Feel free to "question" but you will only arrive at any true conclusion when you spend some time working with the Zaxcom digital wireless and the Audio, Ltd. wireless. Continuing to play the numbers game will lead you nowhere in my opinion. Encouraging others to question the differences utilizing the numbers only will never tell the whole story. “ With this I agree, I’ll definitely will give both systems a listen when the time comes but I really don’t understand the tone of this exchange. Why would you or anybody assume I wanted to badmouth a product just because I have questions about ONE spec. and what the possible ramifications of it could be. (I never voiced concerns about latency, only sample rate) Whatever it is that I said that gave the impression that I was out to cast doubt on Zaxcom’s system or impress people with tech talk or cause malcontent, I apologize and assure you it was not my intend at all. I will refrain from posting more in this thread unless I’m asked a direct question, sound fair? Respectfully

I didn’t join the discussion to start an argument but I’m a bit surprised by some of the responses. Someone earlier noted a detectable sound difference between the AudioLTD system and the Zaxcom, preferring its “clinical” quality over the warmth the AudioLTD offers. The issue of sample-rates was discussed and one person stated that audio LTD might have a leg up on this issue offering a true 48KHz path. I took note when it was suggested that hardly anybody can hear above 16KHz anyway. While I’d never suggest that SR is a dealbreaker I am puzzeled by some of the responses directed at my comments. Case in point. Yes , you are recording 48KHz files BUT if that samplerate is a result of SRC then the limitations of the lowest sample-rate in the chain apply and you might introduce artifacts and latency as a result of the SRC. I realize that going from 32 to 48 isn’t much of a problem, neither is that extra millisecond or two but SRC is hardly ever fully transparent. Yes , 32KHz can theoretically reproduce up to 16KHz audio BUT just as in the case of 44.1 or 48KHz, the filters employed must cut off at ultra-steep slopes in order to keep 16,001Hz out. The result is that in a 48KHz system the roll-off starts way before the theoretical limit of 24KHz. Fact is that the upper range of the audio spectrum of a digital system can and does suffer from the filter used to “cut off at the Nyquist point”. Going back to the 32KHz example, it’s not like that will give you a perfect response up to 16KHz, it’s much more likely to affect the response of anything between 12K and 15K by attenuating it slightly (the filters will roll off steep but not infinitely steep) and possibly by ringing that plagues many filter at the roll-off frequency. Filter design and sample rate both affect the quality of HF reproduction. IMHO it’s one of the reasons why so many of today videocameras feature audio that is being judged as harsh and unpleasant not just by audio mixers like myself but videographers as well, once they hear a comparison recording done on my 664. Yes, dialogue may not have much information above 16KHz and applying a LPF may be common practice (I do it all the time to keep digital garbage out) BUT not at slopes in the 96dB/octave range. And yes, mic pre, mixer, converter all affect the frequency response/ content of the system but that’s not to say that a wireless systems’ audio bandwidth isn’t important. I stared in this business on Vega and Sony analog systems, then lectro187’s, 195’s, 200 series and 400 series. The way to evaluate a wireless for me haven’t changed much. It’s range and reliability, presence or absence of companding artifacts, soundquality, in that order. Companding is no longer an issue, range and reliability seem to be there so Soundquality is next. The goal was and is to get as close to a cabled mic as possible. Whenever I worked with Sennheiser 5000series gear I felt it was as good as it gets with lectros and other ENG style systems being a far distant. Once the 400 series came out I felt that the gap has closed significantly due to the advantages of digital technology. Now we have fully digital systems but our useable spectrum is disappearing. I realize the need for design compromise but I understand the specs and their significance enough to question the assumption that 32 vs 48KHz SR won’t make any difference in the fidelity of the system.

I'll happily agree that a well IMplemented 32KHz sampling rate can sound good up to the cut off point but I'd hope that In digital systems my mic determines frequency response, not the transmitter.

Jeff, since I'm here to learn about digital wireless (I'm a longtime user of Lectros hybrid systems) may I ask why you think samplerate is irrelevant? Samplerate equals frequency response and cutting off at 16KHz (best case scenario) seems a drawback to me. Thanks in advance.

Thanks for the comment. If you’re recording the Zaxcom’s RF audio then you’re effectively recording a 32 KHz, regardless if the receiver or mixer applies SR conversion . That will not change the shortcomings of 32KHz sample rate. If I understood correctly Zaxcom is using a 32KHz sample-rate for RF. If there were any audio above 16KHz in that transmission (regardless of the transmitters internal SR) the reconstruction filter at the DA stage receiving the baseband digital audio ( I’m not 100% confident in my lingo here) will not be able to properly do it's job for those higher frequencies since Nyquist states that you need 2 datapoints per frequency. The audio used for the RF must therefore be cut off by a steep AA filter that also prevents the lower sideband intermod between SR and audio frequencies to creep back into the audio range. per Nyquist this must happen at 16 KHz or lower (or a multiple of that if oversampling is used) As a result one might be dealing with artifacts from that lower AA filter frequency in the audible range. Someone was describing a lack of warmth in the Zaxcom system, could be a symptom, I think that the reason DSLR cameras' audio sounds so bad in the upper ranges is related to poor filtering. Even at 48KHz capture on lesser cameras I still frequently have to use LPFs to deal with that issue, at 32KHz those issues have the potential to further encroach into the audible range. Is this making sense?

I don't have any test equipment handy right now but isn't it true that data compressed internet audio streaming is usually bandlimited to below 15KHz? Streaming above 16KHz content over the internet to evaluate hearing capabilities seems idiotic to me. Most mp3 and other codecs cut off at 15KHz. I can hear/ feel 16KHz just fine, so can many listeners (just get an old NTSC CRT TV set with the 15.75KHz flyback whine, turn it on and off and see if you can feel or hear the diff, many can and will) I would never record at anything less than 44.1KHz samplerate, 48 KHz being the preferred rate, no need to go higher. To say that 32KHz SR =16KHz bandwidth is also assuming that your AA filters are that steep and don't introduce ringing. IME most recording devices that cut off at 15KHz usually exhibits a rather unpleasant upper HF response due to ringing filters. As a post mixer I frequently use a LPF set to 16 or 17KHz to get rid of that stuff in files recorded at 48KHz, wouldn't want to be forced to roll of at 12 or 13KHz because someone recorded at 32KHz. Just my opinion. yes, it is delayed but your eye is also further away from the screen so the brain will connect the dots. latency that conforms to physical reality is never a problem, latency that originates in the virtual world can be.

Nice video. I thought that Neil's "critique" of digital was actually rather constrained and I could almost accept it. At least he didn't bore us with "theories" about missing parts in the audio due to "stairsteps" or the need for ultrahigh sample-rates. I thought the universe analogy makes sense not scientifically but on an intuitive level. And being mindful of the fact that you might as well digitize at the point of analog capture (like the CLASP system) is a welcome surprise too for a guy who has spewed some pretty nonsensical stuff over the years. I always admired him as an artist though, one of the best. Thanks for posting. He does stuff like that, like claiming that the stage-power voltage is not exactly what it should be (he says he can hear it in the way his "59 Fender tweed Deluxe amp distorts), he might even venture a guess as to by how much it's off. usually nobody challenges his assertions, so good for you.

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Just wondering about how to handle the 60 seconds of tone (-20dBFS/ roughly -20LKFS) and 30 sec of silence before the program starts when measuring/ correcting loudness to a target value of -24LKFS. The 1770-3 standard gates for silence but doesn't ignore tone so the integrated loudness will be skewed by that tone if included in the file to be measured/ corrected. I'm mixing close to -24LKFS by ear, doing what I've always done but I want to include a render for loudness to get as accurate as possible using RX5's loudness tool. If I include tone in the file (like I always did in the past) it'll turn down the audio because there's 1 minute of "content" (tone) that is 4 dB too loud over time, not good. On the other hand, the guys doing the layback want tone included in the file. There has to be a standard way of dealing with this by now. Thoughts?

No, it doesn't. The 664 meters use an analog scale with 0VU = -20dBfs. They should have done 2 things, IMO. They should have designed the I/O to handle +24dBu at minimum and they should allow the digital reference to be selectable (at least include -18dBfs as an option) A VU style meter doesn't show dBu, it shows dB relative to a reference level with 0VU =reference . That reference level used to be -10dBv for consumer gear and, +4dBu or +8dBu for prolevel gear, then Mackie and some others started this 0dBu=0VU thing years ago. It might be a costcutting measure since it brings the requirements on the analog I/O closer to the range of consumer gear. In the end it makes sense to preserve 20dB of "headroom" by moving the reference level to a point that's 20dB below the maximum analog output but then it's not really a reference level anymore, I mean, why not use -6dBu as =0VU on an XLR output? I used to not even consider equipment purchases where, at +4dBu, the analog I/O doesn't equal or exceed the artificial digital headroom set by the agreed upon digital reference of -20dBfs (-18dBfs in Europe) but my guess is that those days are over.

Heard back from Matt@ Sound Devices. "Let me step through a few things on the 664:-The maximum analog output level is +20dBu. -This corresponds to 0dBFS, and is not adjustable. -This of course implies that -20dBFS = 0dBu. Because of this, it is not possible for -20dBFS to correspond to +4dBu; this would imply a maximum analog output level of +24dBu, which is beyond what the voltage rails of the 664 will allow. We have for quite some time now shipped all of our units with defaults of 0dBu = 0VU, as this was requested over and over again from customers. It somewhat flies in the face of convention from years ago of having 0VU = +4dBu or +8dBu, but then again, as electronics have gotten quieter in real world situations this arbitrary reference has become less relevant". Not sure I like what I hear but at least I know what's going on. Also asked about the 0dB pan-law and got this: "The panning really comes from how most of our customers use the product. Most are not recording music, and don't use the pans like "real" pans - rather they hard pan L, C, or R simply as a bus assign really. In those situations, it is easier for them to deal with the gain not changing as they pan." Makes me wonder why they didn't just use a routing switch (LCR) and sae on real estate. Other than that it a fabulous mixer. I took it out for a 6 microphone multicamera shoot yesterday and the results were great, really like the preamps, huge improvement over 442, and the unit is so intuitive it hurts:)

I just took posession of a 664 and am a bit confused about the menu options. In the 442 and 552 that I use I have the option to calibrate the tone level (defaults to 0dBu for reasons unkown to me) AND the meter reference level (also defauts to 0dBu=0VU).I set all our mixers to tonelevel =+4dBu, meter reference level 0VU=+4dBu. With the 664 this seems impossible. The tech support person @‌ SD didn't really underrstand my concerns so I was wondering how to best deal with this. Recalibrate all out SD mixers to 0dBu = tone= 0VU? I know the output of the 552 is a max of +20dBu but we've never had any issues with the output limiters set to +17 dBu (honestly I thought they were kicking in at +17VU which would be +21dBu in our setup) Again, in my world -20dBFS = 0VU = +4dBu = 1.23Volts