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Zaxcom and AudioLTD digital wireless - the future


RadoStefanov

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9 hours ago, daniel said:

Eh? Phase switch doesn't delay an input, it inverts the wave, mainly for MS use. 

This is an exceptionally purist thread :)

But to introduce some dirt from the field ie. trial and error, reversing the phase can sometimes give a more acceptable result for mixing sources when one is a little delayed eg. analog radio mics mixed with analog boom when the boom is forced out further etc. or two radio mics separated by some distance, which is why I had a breakout box made for my Fusion with switches for phase reversal to give me instant mix comparison without resorting to clumsy menus.

Obviously not as fine tunable as dialling in a specific delay but can often be better than nothing when there's some sort of phase cancellation.

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22 minutes ago, pindrop said:

This is an exceptionally purist thread :)

But to introduce some dirt from the field ie. trial and error, reversing the phase can sometimes give a more acceptable result for mixing sources when one is a little delayed eg. analog radio mics mixed with analog boom when the boom is forced out further etc. or two radio mics separated by some distance, which is why I had a breakout box made for my Fusion with switches for phase reversal to give me instant mix comparison without resorting to clumsy menus.

Obviously not as fine tunable as dialling in a specific delay but can often be better than nothing when there's some sort of phase cancellation.

I've also worked with a deva user who would have inline phase reverse adapters on the patchbay on his cart- usually on the booms if there was an issue as it's faster and more obvious something's been changed than running through the menus 

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1 hour ago, pindrop said:

This is an exceptionally purist thread :)

But to introduce some dirt from the field ie. trial and error, reversing the phase can sometimes give a more acceptable result for mixing sources when one is a little delayed eg. analog radio mics mixed with analog boom when the boom is forced out further etc. or two radio mics separated by some distance, which is why I had a breakout box made for my Fusion with switches for phase reversal to give me instant mix comparison without resorting to clumsy menus.

Obviously not as fine tunable as dialling in a specific delay but can often be better than nothing when there's some sort of phase cancellation.

I used to try the phase switch for exactly this reason (cos I'd see some colleagues do this) and generally I concluded it still sounded phase-y but just in a different way :-) I'm not such a purist, so i'm guessing that inverting the signal just slightly changes what freqs are getting knocked out, in some cases it might have sounded marginally better (if everything stayed put) but given most bag mixers only have the function on 1 channel it wasn't much of a solution when running 2 RMs and boom. IMHO the c.3ms delay on lectro hybrids had a preferable outcome when mixing a 'loose boom' with body worn RMs.  

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On ‎5‎/‎2‎/‎2016 at 8:48 PM, daniel said:

5Eh? Phase switch doesn't delay an input, it inverts the wave, mainly for MS use. 

Please read about the golden rule 3:1 ratio and some general info on how a phase reverse works.A phase switch is an excellent tool that works great most of the time.

And lets make something clear:
 
The analog signal from 3ms analog Lectrosonics system goes through the ADC of the recorder's input that adds additional latency of between 2-5ms.
 
With Zaxcom I am not aware of any additional latency after the receiver DAC. Maybe maximum of 1ms for the SRC . So in the end both systems are equal in latency.
 
I have often worked with both lectro and Zaxcom in the bag and have never noticed any difference whatsoever!
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everything apart from digital inputs goes through the recorders ADC, so any latency added there can be ignored since it is the same for every input.
using digital outs form your QRX receivers adds 1ms to your latency for the SRC to do its thing.

in my experience, zaxcom latency is dependent on hardware and transmission mode.
for the lowest latency, use USMONO transmission into an ENG receiver.
the QRX receiver adds a little latency. using it in dual mode adds a little more.
XR mode also adds latency.

all the additional processing takes its time.

it has been a while since i last measured this in my zaxcom gear, so recent firmware updates may have changed things.
you can easily check your delays using a hard wired mic and the radio mic signal chain of your choice, and something like audiacity which will show you the sample number of your cursor, as well as a mild amount of maths.

persoanlly, i prefer to try and get my analogue and digital sources time aligned, but only to compensate for latency added by my digital radio mics. never to compensate for a distant mic, as that doesn't allow for the drop in signal over the relative distance between the two sources.

at times, i would prefer smaller adjustments than 0.1ms, as it sometimes doesn't sound absolutely perfect when adjusting. though it is still better than not.

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2 hours ago, RadoStefanov said:

Please read about the golden rule 3:1 ratio and some general info on how a phase reverse works.A phase switch is an excellent tool that works great most of the time.

And lets make something clear:
 
The analog signal from 3ms analog Lectrosonics system goes through the ADC of the recorder's input that adds additional latency of between 2-5ms.
 
With Zaxcom I am not aware of any additional latency after the receiver DAC. Maybe maximum of 1ms for the SRC . So in the end both systems are equal in latency.
 
I have often worked with both lectro and Zaxcom in the bag and have never noticed any difference whatsoever!

The context for my statements has been analogue mixers where the only AD conversion in the chain after the RMs is the camera and NOT digital mixer/recorders.

Yes, 3:1 ratio helps with phase by virtue of the difference in levels between the mic placed at 3 feet and the mic placed at 1 foot from the source, because the principle applies when the placements concerned are 3 metres to 1 metre or 6 yards to 2 yards all of which would have quite different timing differences but (in theory) would yield a 10db difference in signal strength of the 'spill' to primary source so when combined, the wave amplitude of the 'spill' is insufficient to have much impact on the wave amplitude of the primary source of each mic. Maybe you need to go read up on the 3:1 ratio again :-)

IME the phase switch is just a bit of a trick on the mind - it sounds different from before because the 'flanges' have shifted and not because you have got rid of them.

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BTW, I am actually really sorry to have derailed this thread. It seemed to me some things were being ignored or conflated and I was trying to make myself and a context clear because, shocking as it sounds, some people do still use analogue mixers straight to camera and as lovely as both these new RM systems are i wanted to discuss how either may impact a workflow on an analogue mixer. It was not my intention to turn a very interesting thread, with contributors far more expert than I, into some kind of 'ding dong'

d r 

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13 hours ago, soundslikejustin said:

That is completely compensated for by the Tv. 

Not on LG, Samsung or Panasonic LEDs. Every CES I bother their techs with TV latency issues.

They consider it a non issue because most people go through an AV receivers first. The AV receivers does not know how much delay should be added to the sound. 

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The modern HDMI spec sends both picture and audio to the AV receiver, and then back to the television, so that the two are synced correctly.

For that to work they would need to be in sync before being converted to HDMI. The HDMI may keep them in their original sync, but it won't improve sync that's gone off

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hasn't this gone bit off-topic?
i think we all agree that delay on (multichannel-) recording and on playback are two completely different issues..

but back to topic about the future of wireless:
for somebody like me who has no understanding of all the engeneering involved, it's puzzling that it's so hard to do 48/24 digital with low latency these days, and that quality wireless systems are still so expensive. since all manufacturers struggle with it it seems to be difficult as hell, but still - a teradeck video transmitter can send 100 times the data at 1ms latency for half the money. (arguably it has less range, eats lots more power and it would probably be hard to operate more then a few next to each other, so that might have something to do with it : )

but i can imagine that in the 5-10 years full digital transmission will become the standard simply because technology advancement will make it cheaper and  easier to do then with analog hybrids. i even dream of systems with built in recorder and timecode that equal the quality of todays high-end systems for under 1000USD, but i guess we'll have to wait and see.

chris

 

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"hasn't this gone bit off-topic?
i think we all agree that delay on (multichannel-) recording and on playback are two completely different issues.."

Yes, it has gone very much off topic but sometimes that's the way things go. I feel the original topic went somewhat astray even early on when for most of us the issue of latency and/or sample rate with digital wireless is essentially a non-issue.

"but back to topic about the future of wireless:
for somebody like me who has no understanding of all the engeneering involved, it's puzzling that it's so hard to do 48/24 digital with low latency these days, and that quality wireless systems are still so expensive. since all manufacturers struggle with it it seems to be difficult"

I don't know that any manufacturer is "struggling" with digital modulation for our wireless mics --- it is difficult, it obviously involves fairly sophisticated engineering and creative understanding of digital audio, but Zaxcom has of course, figured out most of this and has had outstanding performance with a line of pure digital wireless for over 10 years. The other manufacturers, I'm sure, are working on these things, with only Audio, Ltd. at this time having come up with a viable digital wireless (the subject of this original topic). It remains to be seen how much of this Audio, Ltd. has gotten right before we can know any competitive advantage they may have over the existing Zaxcom units. Of course we are also very interested in what Lectrosonics will come up with since by all accounts, Lectro is the dominate player in our world.

"a teradeck video transmitter can send 100 times the data at 1ms latency for half the money. (arguably it has less range, eats lots more power and it would probably be hard to operate more then a few next to each other, so that might have something to do with it : )"

You bet that has something to do with it --- would you want to be the first to try and put a Teradeck transmitter on an actor (and asking them by the way do you mind sticking this 10Ah battery block in your pocket).

"but i can imagine that in the 5-10 years full digital transmission will become the standard simply because technology advancement will make it cheaper and  easier to do then with analog hybrids."

Digital modulation will become the standard but not because of technological advancement --- it will become the standard because it is the only way to effectively deal with the shrinking RF landscape and provide the features and functions that we need in production. For me personally, and a whole lot of others, digital transmission with total remote control, little to no intermod, wideband and high density, local transmitter recording, this has already become standard for us.

"i even dream of systems with built in recorder and timecode that equal the quality of todays high-end systems for under 1000USD, but i guess we'll have to wait and see."

You will only have to "wait and see" if the most important thing for you is the cost --- a system with "built in recorder and timecode" (and quite a bit more) is already available and in constant use by those who own Zaxcom wireless. Whether the cost will come down in the future, from any of the manufactuers, who knows --- I'm not going to hold my breath for a full digital recording wireless from Azden for under $1000. that meets or exceeds my demands for wireless performance and audio quality.

 

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8 hours ago, Lancashire soundie said:

The modern HDMI spec sends both picture and audio to the AV receiver, and then back to the television, so that the two are synced correctly.

None of this fixes the picture latency of a LED or LCD. If that was the case plasmas will be 7-8ms early from sound. HDMI does not know how much the TV lag is.

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Off course the 7-8ms value of modern LCD and LEDs is without picture post processing turned on. It is just how long it takes the led panel to refresh. With post processing the additional lag gets from 18-80ms depending on models. While this is important for gamers I don't see how it makes any difference for watching TV or movies

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2 hours ago, Jeff Wexler said:

for somebody like me who has no understanding of all the engeneering involved, it's puzzling that it's so hard to do 48/24 digital with low latency these days, and that quality wireless systems are still so expensive. since all manufacturers struggle with it it seems to be difficult"

I don't know that any manufacturer is "struggling" with digital modulation for our wireless mics

well, seems to me that they are struggling with 48/24 with 1ms latency - if it was easy to implement they would be offering such a product, no? if it's needed or not is another question and people might disagree on their personal needs and preferences.

but i'm pretty sure if (or rather when) zaxcom brings out a system with 1ms latency or 48/24 sampling they will label it clearly so and announce it as an improvement.

 

2 hours ago, Jeff Wexler said:

"i even dream of systems with built in recorder and timecode that equal the quality of todays high-end systems for under 1000USD, but i guess we'll have to wait and see."

You will only have to "wait and see" if the most important thing for you is the cost

yes, unfortunately for my needs cost is an important issue atm.
if i had 4000-5000USD per channel i agree todays solutions are perfectly fine. 

 

 

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4 minutes ago, daniel said:

Out of interest, can any of the Zaxcom aficionados tell me what the channel density is of the current (not ZHD) is?

i run them at 0.7MHz spacing. have used 0.6MHz in the past which has been ok.

assuming thats what you were asking.

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The older non ZHD modulation was roughly 4 times as wide, recommend spacing was about 700 KHz and the signal was for the most part resistant to intermodulation the possibility for intermodulation did exist.

ZHD is 50 KHz wide, spacing can be as narrow as 100KHz and the signal shows no intermodulation.

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So the channel density of the older zaxcom system is about the same as an analogue (best case scenario) system (but less than Sony's current (but not available in North America?) DWX digital radio mic system (16 channels per 8Mhz) - which also happens to be 24bit/48khz with 3.6ms latency [non of which matters of course, especially if you've recorded the audio on a Zoom in mp3 mode while playing a video game on Rado's LED TV :-]).

Obviously the new ZHD digital system with 100khz, intermod free spacing is truly an incredibly awesome engineering achievement whatever the latency or sample rate and if you're working in a highly congested spectrum then a high density RM system is the only way to go from the sound of it.

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2 minutes ago, daniel said:

So the channel density of the older zaxcom system is about the same as an analogue (best case scenario) system (but less than Sony's current (but not available in North America?) DWX digital radio mic system (16 channels per 8Mhz) - which also happens to be 24bit/48khz with 3.6ms latency [non of which matters of course, especially if you've recorded the audio on a Zoom in mp3 mode while playing a video game on Rado's LED TV :-]).

Obviously the new ZHD digital system with 100khz, intermod free spacing is truly an incredibly awesome engineering achievement whatever the latency or sample rate and if you're working in a highly congested spectrum then a high density RM system is the only way to go from the sound of it.

whilst you may have a similar channel count with the current XR mode within a 6 or 8MHz channel as an analogue setup, once you need more frequencies, the digital system shows its advantage. 

in order to avoid intermod, the space you require for more analogue channels increases dramatically, whereas with digital, just keep adding your minimum spacing. so you can get say 16 digital systems within 2 tv channels, you would need about twice that bandwidth to get the same number of analogue channels running.

Audio Ltd, last time I saw a unit, has its preset spacing at 0.4MHz - so can get more channels than the Sony in an 8MHz window. with 2ms delay but 44.1 KHz sample rate. but I will get a program that will batch convert the sample rate of the recordings if I need to use them, and rely on either analogue outputs or sample rate converters to deal with everything else.

 

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