{ "numMessagesInTopic": 3, "nextInTime": 2004, "senderId": "4bvuHkshH55uj85v_7otGXfaXxfYf-JRRBs1ksGwrw3zje67aSUNlarHIUbmOaY4Mcraoe-FrFqcxUM8_0ZF69vEM6sOELQ30Ukm", "systemMessage": false, "subject": "Re: [01v96] Re: linking 01v's", "from": ""Ronny Morris" <romo1@...>", "authorName": "Ronny Morris", "msgSnippet": "... From: Carol To: 01v96@yahoogroups.com Sent: Friday, August 20, 2004 5:55 AM Subject: [01v96] Re: linking 01v s ... (slave) ... 3, ... master s ... on ... ", "msgId": 2003, "profile": "ronnymorris2001", "topicId": 2001, "spamInfo": { "reason": "0", "isSpam": false }, "replyTo": "LIST", "userId": 165376131, "messageBody": "
\n\n\n----- Original Message -----\nFrom: \n Carol \n\n \nSent: Friday, August 20, 2004 5:55 \n AM\nSubject: [01v96] Re: linking 01v's\n\n--- In 01v96@yahoogroups.com, "matt_bang" \n <matt_bang@y...> wrote:
> Hi I \n was referred by Ronny from the 01v list and my question is for
> Doug. \n My question is about linking two 01v's (not 01v96, sorry just
> to make \n sure) so here is the original post: (sorry it is a bit long)
>
> \n It is for live sound,
> I need more than 12 inputs (with preamps), but \n if i link them with
> the digital coax it is like running one mixer \n with a sidecar
(slave)
> witch is not good because the aux sends of \n the side car are not
> useable because only the stereo mix of the side \n car is sent to the
> other one and in live sound the aux sends are not \n used for effects
> (100% true for the 01v because of built-in effects) \n but mainly for
> monitor mixes on stage.
>
> The \n only way I found for doing this is to link them together with
> the \n digital coax and then take the omni outs (witch are aux 1, 2,
3,
> \n 4) of the slave and link them to inputs 13/14, 15/16 witch are not
> \n fed to the mix but to aux 1,2,3,4 of the master and of course you
> put \n the levels to 0db on the master. This way you can send aux
> \n 1,2,3,4 of the slave to the master at the level you like and also \n
> your stereo inputs are not lost because you still have them on the \n
> slave.
> Next point, still talking about digital link between \n the 2 01V's, I
> have read in the manual that you have to put a delay \n on the
master's
> inputs witch is perfectly normal due to the AD \n conversion time so
> you will be in phase. Now my question is, is there \n a delay to put
on
> inputs 13/14 and 15/16 of the master, in the \n first example we have
> AD conversion on the slave an then DA \n conversion on the master but
> for these inputs we have ADDA in the \n slave and then AD again in
the
> master ? Just to be in \n phase. For sound quality I have not tried
yet
> but we'll \n see.
>
> But all of this is not very user friendly, if you \n know how it
works,
> fine, but sometimes when doing a live gig \n (production wise), you
are
> not the person who mixes.
> \n
>
> Thank you very much !
Matt
I have a friend \n who has an older Promix so I will have to give this a
try first. I am not \n sure.
First with the coax link I don't see where adding delay to the \n master
is needed although I will look into it. Lets say both units have \n 1.2
ms latency through each one. Half of that going from a-d in , half \n
from going d-a out. In the slave 1/2 the time would be used going \n
digital.Now in the digital format it is sent to the master. The
master \n will cause 1/2 of the latency time putting it back to analog.
The same \n thing?
I "think" there is no second adda conversion it the coax digital \n
link correct?
Now for your 4 aux sends.
A signal goes through \n the slave and comes out its aux analog. 1.2ms
latency.
This goes to the \n master.
This puts the slave 1.2 ms "Behind" the master.
Adding delay to \n 13-16 will only make it even more behind the master.
Here is how to look at \n it.
If you had one git on stage and put a mic on it with a Y \n cord.
One side went to the master and one to the slave.
The slaves send \n is sent back to the master via aux's .In just this
case the Aux's are sent \n to the masters l/R instead of monitor out
sends. When it goes out the \n masters l/r with the masters same mic'ed
signal it will be 1.2 ms behind \n the masters signal.
The only work around I see is this.
On the \n master outs L/R add 1.2 ms delay and go to a analog mixer.
Instead of \n linking the Slave to the Master go out the slave's L/r and
go into the \n analog mixer as well. This would put the Master and Slave
in better \n alignment. Of course you are looking at another analog
stage and you would \n want to fine tune the delay times with something
like smaart.
If you \n keep the Aux's from the slave just as monitor sends out the
master I can't \n see where the 1.2 ms or so would be a problem.
Now for the big \n question. Is 1.2ms latency enough to worry about? We
are talking about \n just alittle over a foot. Two mic's 6 feet apart on
stage cause more phase \n problems than that. I would try to put the
drums on one unit(the slave?) \n to keep things tight.
I'm am going to have to give this a try before I \n can say anything for
sure.\n\nThanks Doug, I already explained to him that \n the digital delay on the master would be set to 14 samples on a soley digital \n coax link. He just needs to know the latency of the A/D and D/A conversion, \n which I had thought that you had said was 1.6ms, but I see now that you said \n 1.2ms.\n\nSo the answer is 14 samples at 44.1k \n delayed on the master on the coax send and 1.2 milliseconds more delay for \n each A/D or D/A conversion. So, if the conversion is 1.2ms than he needs to \n set the STereo buss at 14ms delay on the master and add 2.4ms for the back to \n analog conversion. So the answer would be to delay the master stereo \n input by 14 samps and 2.4 or 2.43ms [+14 samples (.3ms)] on ch \n input 14-16 on the master. 2.43ms, would be the total delay time on the master \n after the conversion and digital through put on the slave. However, the \n digital link being only 14 ms, is really inconsequential and many people have \n run the master without delaying it for a year, before I told them to delay the \n signal in the internal buss which is 14 samps and they noticed no \n difference. No surprise as with an original signal delayed by 14 samps it is \n only noticeable when you can hear the original signal and the latent signal \n out of two separate speakers. In the worst case scenario, if anyone heard \n a difference (which I can hear a .2ms delay while auditioning the original \n undelayed with the delayed signal via two L and R speakers) and this wouldn't \n be a delay difference recognition, but due to the Haas effect and only in \n stereo mode it would throw a stereo pan trajectory slightly off center \n image, but In his case, I agree that it's really not going to \n make a lot of difference even delaying the master 01v for the analog \n conversion.\n\nIn a nutshell you can not delay the \n master to reflect both the digital latency and the analog conversion latency \n with the stereo link alone, because they are two different transfer mediums \n with highly different delay times. It's like having a rack full of analog and \n digital processors, each processor is going to delay the signal, but if it's \n mono, than the total latency of all processors would be inconsequential to the \n delay between the mics themselves, like Doug mentions. The same mono \n signal is going to be latent, but all signals would be delayed close to the \n same latency. You really only have to worry about minute latency when it's \n between a L and R stereo source, which I mentioned would throw the center \n image to the side of the delayed signal and slightly skew the center image. \n However we are talking about a live app, with microphones having more distance \n between them than the 5 times the delay of the conversion and also if he's \n running mono, both PA sides would match regardless. And as I pointed out to \n Tom, to walk around the room and notice the delay that he's putting on his \n speaker array, you are going to have the evironment itself factor in WRT, \n quarter wave nulling and natural comb filtering. It's impossible to delay \n signals from a speaker delay and get the same sound in all spots in a \n concert room as the size of the room and early reflections are going to be \n different, in all spots in the room. At quarter wave nodes you are going to \n get some low mid freqeuncies that are going to attenuate, some up to -15dB. So \n you can time align the speakers, but once the signal is SPL and leaves the \n speakers, different frequencies are going to arrive at different times, at \n different spots in the room and you will hear a difference in the tone due to \n quarter wave phase nulling the low mids at some spots and not \n others. Basically you are time aligning for the sweet spot as it's \n impossible to time align all frequencies emitting from a single source that \n arrive in all spots, in a room with early reflections. \n\n\nAnyway, Matt, you can try delaying \n all channel inputs except ch's 13-16 on the master by 2.4ms (I wouldn't worry \n about the .3ms coax delayed), which would be a sample delay of 107 samples, a \n 2.43 ms delay will not be recognizable as a delay to the human ear with one \n source as it's too fast for the ear to hear, you won't hear early reflections \n delay until it passes around 15-20ms with a single source. In any case running \n analog mono FOH, and using digital outboards, I've not delayed the signal at \n all and never had any problems. You will only be getting a slight phasing if \n you were processing one slde of a stereo signal and not the other, because in \n mono all channels and signal will exit the master at close to the same time \n and under the length of recognizable delay. In a stereo system because you \n have two sources to reference it is possbile to hear pan trajectory shift by \n as small as .2ms on one side, due to the Haas effect as mentioned above, but \n again this is not recognized as delay but stereo image shift. Now the \n difference between your ears WRT to time arriving at the ears is only .5ms, \n the distance that sound travels at sea level at 72 degrees temp and only if \n you stood 90 degrees angle to the source, than you have to factor in shadow \n effect, which is the head itself between the two ears and these are \n inconesquential in a most live apps, because that's the way people normally \n hear anyway. Delaying either the digital or analog conversion by 2.4 or \n 2.43ms, might not be consequential because as Doug pointed out, two \n microphones at the frontline that are spaced by 10 feet are going to have a \n delay bleed of 8.9ms. If two singers are singing harmony than both of \n their original signals are going to be the same, if both are the same distance \n from mics, but the idiosyncrasie of their own timing between singers is going \n to be more than the phasing from all mics on a medium sized stage. In a live \n situation these bleed times are natural and that's why I tell people that when \n they go to try delaying drum mics to a tee to to reduce or eliminate phase and \n they aren't getting a very good sound, it's because they are taking \n the natural phasing and normal Haas effect that a real live drum set has, \n relative to the ear sans any PA. IOW, if you time delay all drums or live \n sends (not counting speaker arrays, just normal hearing) to be exact you are \n going to actually be making the sound unnatural as all instruments and drums \n have a normal amount of phasing due to the time that the different instrument \n signals reach the ear (Haas effect). I just don't think that you will hear a \n significant difference if you don't delay the master to accept the analog \n conversion delay times from the slave. Try the 2.4 or 2.43ms delay on 13-16 \n and than A/B with it set with no delay and tell me what you hear? If there is \n any difference, it would be very small.\n\nKeep On \n Trackin'\n\n------Ronny Morris - Digitak \n Mastering------\n
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