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Zaxcom and AudioLTD digital wireless - the future


RadoStefanov

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I have never understood the need to dial in delays on production sound work. Not in a world that is non-linear, file-based, etc... I am sorry, but i just dont quite get it. Unless it is important to the provision of a preview recording in the form of video assist/or to camera(s) OR in direct-to-camera (single-system) work... 
 

 

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1 hour ago, RadoStefanov said:

So if we are sticklers for accuracy a shoot with camera 2 meters from a talent should ideally be delayed by approximately 5-6ms to match the realism of sound traveling through air.

 

that is an interesting idea, but cuts somewhat short IMO.

if we're watching a film on a 40" screen TV from 3 meters away we'll get 9ms delay anyway from the playback arrangement, which means it feels like the sound originates at the TV screen (which it does!) and also matches the impression of watching a person talking 3meters away, even if the lav is placed right next to the mouth and there's zero recording delay.

similarly in a cinema, if you're  10 meters away from the screen and the speakers are right behind the screen, we'll get 30ms delay from the room which gives the feeling the sound originates at the screen and kinda feels like the person talking is 10meters away too - which the picture actually is, even if it's a close up shot from 1meter distance (all this is ignoring modern surround sound technology).

so in a way, the delay gets added anyway in playback. all that said, an additional 5ms is usually going to be unnoticeable (it's hard to spot a 1 frame out of sync dialogue for normal viewers, which is a full 40ms)

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1 hour ago, soundtrane said:

I have never understood the need to dial in delays on production sound work. Not in a world that is non-linear, file-based, etc... I am sorry, but i just dont quite get it. Unless it is important to the provision of a preview recording in the form of video assist/or to camera(s) OR in direct-to-camera (single-system) work... 
 

 

My analogue microphones come into the mixer with 0ms delay.  The radio mics I also use have a 3.5ms delay.  That's fine most of the time, but if you have to mix things down to mono, it's going to sound pretty bad when those sources are in close proximity (say two wired lapels and two radio'd lapels in a discussion).  So I delay the analogue sources so that everything hits the mix at the same time.  Of course you may have the iso tracks, but even so it's one less job for the editor - especially considering that you know how much delay is required, whereas he mightn't.

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On ‎4‎/‎24‎/‎2016 at 7:19 PM, Jamie Tongue said:

http://www.audiocheck.net/blindtests_frequency.php

Can you hear anything above 16,000 Hz anyway? No, no you (probably) cannot.

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.

6 hours ago, John Blankenship said:

In a movie theater, sound is delayed by approximately a frame for every twelve rows further back you sit (figuring about four feet per row).  

So, for instance, if you sit in the back row of an eighteen row theater, your sound is delayed by more than sixty milliseconds.

Put another way, if two laved actors are standing six feet apart, the sound is delayed roughly five milliseconds from one mic to the other.

Want a perfect, frequency-aligned world?  Make all actors occupy the exact same spot at the exact same time, use only analog gear, and have every theater goer simultaneously sit in the exact same spot and not move their head at all.

Alternative:  Live in the real world and don't sweat the small stuff.

 

 

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.

 

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2 hours ago, Werner Althaus said:

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.

Nobody that I know of is recording at 32khz sample rate. The industry standard (motion pictures and television) at this point is 48K - 24 bit files --- that is what we're all doing. Prior to file based recording, 48K - 16 bit was standard with DAT. The rest of the world, mp3, Internet streaming, satellite transmission, etc., I have no idea what goes on there. The issue of Zaxcom digital transmitters sampling at 32K does not causer any of us to record our tracks at 32K, so any "adjustments" that you may fear you are having to make in post, don't even give it a second thought. The tracks that you will deal with in post will be industry standard 48K - 24 bit wave files.

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5 hours ago, Jeff Wexler said:

Nobody that I know of is recording at 32khz sample rate. The industry standard (motion pictures and television) at this point is 48K - 24 bit files --- that is what we're all doing. Prior to file based recording, 48K - 16 bit was standard with DAT. The rest of the world, mp3, Internet streaming, satellite transmission, etc., I have no idea what goes on there. The issue of Zaxcom digital transmitters sampling at 32K does not causer any of us to record our tracks at 32K, so any "adjustments" that you may fear you are having to make in post, don't even give it a second thought. The tracks that you will deal with in post will be industry standard 48K - 24 bit wave files.

 

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?

 

 

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13 hours ago, Lancashire soundie said:

My analogue microphones come into the mixer with 0ms delay.  The radio mics I also use have a 3.5ms delay.  That's fine most of the time, but if you have to mix things down to mono, it's going to sound pretty bad when those sources are in close proximity (say two wired lapels and two radio'd lapels in a discussion).  So I delay the analogue sources so that everything hits the mix at the same time.  Of course you may have the iso tracks, but even so it's one less job for the editor - especially considering that you know how much delay is required, whereas he mightn't.

+1.

Not everything some of us do gets much audio PP to 'sweeten' the sound (or retime tracks), it often goes no further than a picture editor. With this in mind I try to keep things simple for those downstream and if I can't, I try to be clear and informative about the complications.

Analogue mixers (going straight to camera) still have their place in many sectors and dialling in delay on these is not an option, owning several sound kits is obviously a good way to go but some of us will want expensive kit elements (RMs) to be versatile in respect of the jobs they can go on. I don't think anyone, including myself are saying a bit of latency is a problem in most cases, but it is something we need to know about and consider on a job by job basis as well when we procure kit.

IME the best bits of kit finely balance the various engineering achievements and limitations in such a way as to make them as useful as they can be for what the end user identifies as their requirements. Somewhere in this picture is the marketing (I doubt anyone here will be surprised Audio ltd state in the first paragraph of their 1010 system page the RMs have "an unrivalled end-to-end delay of just 2mS". Other times folks will ask the questions they think (rightly or wrongly) are important to them - sometimes even manufacturers ask these sorts of questions (as part of their marketing strategy?).

Maybe I'm being a bit sensitive here as I asked the question about latency and some of the responses read dismissively or even like foreclosure, on the other hand maybe others are being a bit sensitive themselves about their favourite brand. I've not read anyone really questioning the engineering brilliance of Glenn and Zaxcom's products so I am a bit bemused at what happens when a fairly innocent question gets asked, not least as this is a thread about Audio ltd's 1010 and Zaxcom's ZHD and latency is 1 of the differences. The discussion might even (at this point) be framed as 'density vs latency' and I've not yet read a comparison in this respect of the 2 systems Eg. 20 vs 60 per TV channel, 2ms vs 5ms(?).

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19 minutes ago, daniel said:

IME the best bits of kit finely balance the various engineering achievements and limitations in such a way as to make them as useful as they can be for what the end user identifies as their requirements. Somewhere in this picture is the marketing (I doubt anyone here will be surprised Audio ltd state in the first paragraph of their 1010 system page the RMs have "an unrivalled end-to-end delay of just 2mS". Other times folks will ask the questions they think (rightly or wrongly) are important to them - sometimes even manufacturers ask these sorts of questions (as part of their marketing strategy?).

Definitely agree with this- it seems that both Zaxcom and Audio Ltd have made different engineering choices here, with some which may suit certain users better than others.  It's seems to be a case that some kind of compromise has to me made in one way or another and it seem that both Sony and Sennheiser have made their own engineering decisions too.  You go with the one which fits the way you work the best.

For another example, if size, bandwidth and power usage go out the window- Neutrik's Xirium pro 5GHz systems can do 2 channels of 24/48 uncompressed in 3.6ms analogue to analogue: http://www.neutrik.co.uk/en-uk/xirium-pro-europe/nxp2tx-e
You wouldn't be able to hide that too easily, though...   

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Richard has said: "You go with the one which fits the way you work the best" and this brings up the fundamental problem with this thread that has devolved into a discussion of just 2 specifications: latency and sample rate, specifications which are being put forth in an effort to compare 2 different digital wireless systems, Zaxcom Digital Wireless and the just introduced Audio, Ltd. Digital Wireless. The discussion of latency, as it applies to another popular wireless, the Lectrosonics line of digital hybrids, seems to have disappeared (I believe as the legions of Lectro users have determined in the real world that it is just NOT an issue). The problem I see is that people may be making decisions on which wireless to use, Zaxcom or Audio, Ltd., based on this lengthy discussion of 2 specifications that are, in my opinion, basically irrelevant. It is clear that Audio, Ltd. would like people to believe that these are the two most important specs that beat the competition (Zaxcom) --- this is unquestionably a marketing strategy which hopes to undermine the perception of the value of their main competitor. 

Daniel has said: "Maybe I'm being a bit sensitive here as I asked the question about latency and some of the responses read dismissively or even like foreclosure, on the other hand maybe others are being a bit sensitive themselves about their favourite brand" --- there is no problem asking questions about latency, but there seems to be an inordinate amount of concern for something which has just not posed any problems for any of us (and for me that is almost a 12 year history of using digital wireless with latency). Has anyone here actually had long term real world experience with the just announced Audio, Ltd. digital wireless? Other than the perception that a lower number is better, has anyone found that the slightly lower latency spec with the Audio, Ltd. had proved itself to be a big advantage over other wireless units (Zaxcom and Lectro) that may have a slightly higher latency? Also, Zaxcom didn't become my "favourite brand" over night --- the Zaxcom Digital Wireless became my choice because it is the best sounding wireless I have ever used (an opinion that has been shared with me on many occasions from people in post that have dealt with my wireless tracks vs. any of the other most commonly used wireless). So, who am I to question which of the specs with Zaxcom differ from the specs of the others? I plan on testing the new units from Audio, Ltd. to develop my own opinion --- I do not plan on getting entangled in a "technical discussion" or the numbers game regarding latency or sample rate. I will put the Audio, Ltd. through its paces and who knows, maybe Audio, Ltd. will become my favorite brand...  or not.

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7 minutes ago, John Blankenship said:

The answer lies in the signature that Jeff often  digital systrm my mic determines tge frenses.

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.

 

Edited by Werner Althaus
keyboard acting up
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If you can hear a diff between the high end wireless systems transmitted audio 32K SR vs 48k or whatever then congrats on your well-preserved hearing.  To me the mounting of the lav mic on talent, their exact wardrobe, head and body position while speaking, the sound of the set BG noise and the lav mic in play are far more interesting topics for discussion anymore.  Coming from the era of fixed freq, noisy, short-range, clippy wireless I would put the technical aspects of wireless mic design in the SOLVED bin, esp with this new generation of gear.  The issues now are what I mentioned above plus the "Spectrum-Eating FCC Monster".

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34 minutes ago, Jeff Wexler said:

Richard has said: "You go with the one which fits the way you work the best" and this brings up the fundamental problem with this thread that has devolved into a discussion of just 2 specifications: latency and sample rate, specifications which are being put forth in an effort to compare 2 different digital wireless systems, Zaxcom Digital Wireless and the just introduced Audio, Ltd. Digital Wireless. The discussion of latency, as it applies to another popular wireless, the Lectrosonics line of digital hybrids, seems to have disappeared (I believe as the legions of Lectro users have determined in the real world that it is just NOT an issue). The problem I see is that people may be making decisions on which wireless to use, Zaxcom or Audio, Ltd., based on this lengthy discussion of 2 specifications that are, in my opinion, basically irrelevant. It is clear that Audio, Ltd. would like people to believe that these are the two most important specs that beat the competition (Zaxcom) --- this is unquestionably a marketing strategy which hopes to undermine the perception of the value of their main competitor. 

Daniel has said: "Maybe I'm being a bit sensitive here as I asked the question about latency and some of the responses read dismissively or even like foreclosure, on the other hand maybe others are being a bit sensitive themselves about their favourite brand" --- there is no problem asking questions about latency, but there seems to be an inordinate amount of concern for something which has just not posed any problems for any of us (and for me that is almost a 12 year history of using digital wireless with latency). Has anyone here actually had long term real world experience with the just announced Audio, Ltd. digital wireless? Other than the perception that a lower number is better, has anyone found that the slightly lower latency spec with the Audio, Ltd. had proved itself to be a big advantage over other wireless units (Zaxcom and Lectro) that may have a slightly higher latency? Also, Zaxcom didn't become my "favourite brand" over night --- the Zaxcom Digital Wireless became my choice because it is the best sounding wireless I have ever used (an opinion that has been shared with me on many occasions from people in post that have dealt with my wireless tracks vs. any of the other most commonly used wireless). So, who am I to question which of the specs with Zaxcom differ from the specs of the others? I plan on testing the new units from Audio, Ltd. to develop my own opinion --- I do not plan on getting entangled in a "technical discussion" or the numbers game regarding latency or sample rate. I will put the Audio, Ltd. through its paces and who knows, maybe Audio, Ltd. will become my favorite brand...  or not.

The first discussion of latency is 4 pages into this thread (you could say that's a bit of latency in itself :-).

I was long time 411 user and worked with latency in that system quite well, I have even suggested in the past it was advantageous.

I don't think we've had "inordinate" discussion about latency or sample rate (raised by Werner).

High density and range are other aspects to this thread and have been given plenty of air time (and are areas in which the ZHD excel), all these things will be more or less important to each of us to varying levels. I just think it fair to consider all aspects of these new systems in this thread and as a Richard suggest manufacturers may compromised on 1 aspect to gain advantage in another.

Not everyone who reads and contributes to this forum will need to fit 60 RMs into 1 tv channel or need to record someone speaking 1000 metres away, some people may want their RMs to play along with analogue sources and a low 'draw' analogue mixer. From what I've read so far the high density, long range of ZHD has been at the 'expense' of an increase in latency over their current and other digital systems (which is not necessarily a bad thing - but something to consider when buying).

18 minutes ago, Philip Perkins said:

If you can hear a diff between the high end wireless systems transmitted audio 32K SR vs 48k or whatever then congrats on your well-preserved hearing.  To me the mounting of the lav mic on talent, their exact wardrobe, head and body position while speaking, the sound of the set BG noise and the lav mic in play are far more interesting topics for discussion anymore.  Coming from the era of fixed freq, noisy, short-range, clippy wireless I would put the technical aspects of wireless mic design in the SOLVED bin, esp with this new generation of gear.  The issues now are what I mentioned above plus the "Spectrum-Eating FCC Monster".

I generally trust the manufacturers of pro gear to make stuff that sounds good and I'm more concerned with the logistical and practical considerations of owning and operating kit. It's a pity the sennheiser ek6042 is not available yet as I would like to see this thrown into this mix as I believe it will have some logistical and practical advantages for many owner operators in europe if not america. It's touting 'average' latency, a long range digital mode and analogue RM compatibility.

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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.

 

Digital or not the frequency response or rather, the frequency content, will always be influenced by the preamp, the mixer, and, if applicable, the a-d converter. In a wireless situation this is even truer, the influence by non-mic components higher. The mic is only a part of the entire chain. If you don't like that, don't go wireless.

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Werner says: "Samplerate equals frequency response and cutting off at 16KHz (best case scenario) seems a drawback to me" --- at first, it would seem to be a drawback, if one only considers the number. So, I will amend my statement about sample rate being irrelevant: it IS relevant but through real world experience (with analog wireless, analog hybrid and pure digital) it is a number (32K sample rate with Zaxcom) that does not concern me. Echoing what Philip Perkins and others have said, it is a number that is way down the list of significant factors affecting how a microphone sounds when it reaches my recorder. I look forward to testing the Audio, Ltd. Digital wireless (the only other viable competitive digital set for our sort of work) and I will then be able to determine if the higher sample rate and/or lower latency provides superior sound. For me, the numbers do not matter, unless of course they do...  we'll see (rather, we'll hear).

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For most videos the dialog will be rolled off below 16k anyway, so a 32k sample rate is sufficient, and will not affect the quality of the voice in the final product.  If you were planning to use the wireless to record SFX that will be heavily manipulated, then it could be an issue, but I would argue in that case it's not the right tool for the job.

There is a point that latency (and frequency response) do become an issue, but I think all of the units available today are well within a necessary spec to not be a problem.  It is good to publish and be aware of these specs, but it is as necessary to be aware of the targets these specs should meet, otherwise the specs are meaningless.

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On ‎4‎/‎29‎/‎2016 at 11:55 AM, RadoStefanov said:

.

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.

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Proper blind real world tests you'd never be able to discern the difference between 32K and 48K, because as Philip eloquently pointed out, there's a number of much more important factors at work.

Bet blind tests would also expose the supposed warmth of one over another is a figment in the head of the listener.
Plus who wouldn't want 1000 meters range, because it means in adverse circumstances the range still remains something very usable.

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"Someone earlier noted a detectable sound difference between the AudioLTD system and the Zaxcom, preferring its “clinical” quality over the warmth the AudioLTD offers."

Not surprising that they sound different — Audio, Ltd. is obviously doing several things differently but to attribute the difference to sample rate is most probably not the correct conclusion. Someone should explain why the method Audio, Ltd. is using produces harsh noises when reaching the usable range limit or why the RF seems more prone to drop outs when confronted with obstacles in the line of sight. I don't think these differences can be attributed to sample rate.

"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."

First of all, let's be clear that the Audio, Ltd. does not offer "a true 48Khz path" I think the sample rate is 44.1. Of course, I regret even pointing this out because I still maintain that you are playing a numbers game, not because you want to dazzle everyone with your knowledge of sampling rates, SRC, Nyqvist, flyback whine and anti-aliasing artifacts, and all the things you will have to do in post to correct for what you characterize as a serious compromise with the Zaxcom digital wireless, you are making this case to cast doubt on the performance of Zaxcom digital vs. the new Audio, Ltd. offering, plain and simple.

"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."

The above statement indicates a complete lack of understanding of how many of us work. 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 — from there it goes to the 24-bit A to D and is recorded as a pure 24-bit 48K file — no sample rate conversion, no additional filtering, no chance of introducing artifacts, etc.

"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."

As I said, no SRC is employed in my recording.

"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."

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. 

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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

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