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Sound Reinforcement - Forums for Live Sound Professionals - Your Displayed Name Must Be Your Real Full Name To Post In The Live Sound Forums => LAB: The Classic Live Audio Board => Topic started by: Airton Pereira on August 11, 2013, 08:41:10 PM
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Hi,
I saw a youtube video about a Yamaha CL5 (http://www.youtube.com/watch?v=JMJ4pKehi90).
On about 13:00 the guy says it's better to route the mains out to two matrix and then send the matrix to the output ports. I am curious about why he thinks this is better. Anyone use this configuration?
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Hi,
I saw a youtube video about a Yamaha CL5 (http://www.youtube.com/watch?v=JMJ4pKehi90).
On about 13:00 the guy says it's better to route the mains out to two matrix and then send the matrix to the output ports. I am curious about why he thinks this is better. Anyone use this configuration?
If all you are doing is a stereo mix to a stereo PA I don't see a reason for using the matrix outs. On the other hand, if you are using subgroups I would route the subgroups to the matrixes so you can change the mix of certain zones, like less band in the front fills, or even more band in a send to a remote area that isn't getting all the stage wash.
Mac
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On about 13:00 the guy says it's better to route the mains out to two matrix and then send the matrix to the output ports. I am curious about why he thinks this is better. Anyone use this configuration?
On an LS-9 we feed the PA with a matrix and have the graphic eqs on the matrix outs so our recording feed of L/R is not affected by the graph.
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If all you are doing is a stereo mix to a stereo PA I don't see a reason for using the matrix outs. On the other hand, if you are using subgroups I would route the subgroups to the matrixes so you can change the mix of certain zones, like less band in the front fills, or even more band in a send to a remote area that isn't getting all the stage wash.
Mac
That's what I thought. On this video the guy is doing a simple mix, so I don't see why route to matrix.
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On an LS-9 we feed the PA with a matrix and have the graphic eqs on the matrix outs so our recording feed of L/R is not affected by the graph.
I use a LS9 as well. I never thought of that, thanks for the tip!
I did a little test after I saw the video and a good point about sending the mains to matrix is to have separate L and R faders, so you can check them one at a time without having to mute one side on the system eq.
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Thats been my default work flow since the days of the PM3500 ( which had an awesome matrix setup for an analog board ). As others have mentioned it provides options even if you merely feed the matrix from the stereo buss. It acts as a DA with separate outputs for the main PA, the record feed, maybe a video camera for the lead singers' husband, whatever. All of this also means separate options for processing be it EQ, compression etc. as well as tailor made mixes as Mac described.
Hi,
I saw a youtube video about a Yamaha CL5 (http://www.youtube.com/watch?v=JMJ4pKehi90).
On about 13:00 the guy says it's better to route the mains out to two matrix and then send the matrix to the output ports. I am curious about why he thinks this is better. Anyone use this configuration?
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Thats been my default work flow since the days of the PM3500 ( which had an awesome matrix setup for an analog board ). As others have mentioned it provides options even if you merely feed the matrix from the stereo buss. It acts as a DA with separate outputs for the main PA, the record feed, maybe a video camera for the lead singers' husband, whatever. All of this also means separate options for processing be it EQ, compression etc. as well as tailor made mixes as Mac described.
+1.
Maybe it's a DC thing. :)
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Also for what it's worth my recording feed is another matrix that's prefade LR in case my FOH guy decides to mix it chill at -5 after soundcheck or crank the encore at +5.
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Also for what it's worth my recording feed is another matrix that's prefade LR in case my FOH guy decides to mix it chill at -5 after soundcheck or crank the encore at +5.
Also FWIW, my record feed is usually the stereo bus fed from a stereo pair of mixes. The PA is fed from matrixes fed by groups. I have group processing, matrix routing, and a dedicated stereo mix that has big meters always on the surface no matter what layer I'm on. Plus I have an easy to grab handle if the record level gets out of hand.
Mac
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I'll try it on my next gig. So far I found a much more stable headphone volume, cause on the LS9 the headphones volume is based on the L/R fader (post-fader) and with the fader constantly at 0db helps with the headphones volume.
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I'll try it on my next gig. So far I found a much more stable headphone volume, cause on the LS9 the headphones volume is based on the L/R fader (post-fader) and with the fader constantly at 0db helps with the headphones volume.
You can choose what the headphones hear in the monitor section of the setup. It need not be LR by default. When there's a solo engaged you hear the solo(s), but the monitor line can be fed by other options too.
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Nils, I know that. I donīt use headphones that much, only to solo channels (mostly the main vocal). But it is still nice to have a steady volume when not on solo.
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I find on the yami digitals I hardly ever set up auxes as groups, I just mix to VCAs, but I still like to have Matrices as outputs because of individual EQ, levels, processing, ect. Even a sub aux or mono feed I'll send to a matrix so that I only have to look at 1 page to see all of my outputs.
With the VENUE, it's a bit easier to build separate mixes with groups, and you can really get into detailed processing and control. I like to have as much of the "system processing" from within the console so I don't have to dive over to an outboard computer to make changes. If I'm the only one mixing for the night, I'll do almost everything from within the board.
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I find on the yami digitals I hardly ever set up auxes as groups, I just mix to VCAs, but I still like to have Matrices as outputs because of individual EQ, levels, processing, ect. Even a sub aux or mono feed I'll send to a matrix so that I only have to look at 1 page to see all of my outputs.
Without groups you lose the ability to tailor the mix to the location. How do you reduce the band levels in the front fills without groups?
Mac
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I can see why on a Yamaha console you might not go via a matrix - the presence of level controls on the omni outs makes this a bit less important. As somebody who goes in to a lot of houses with their own PA, some of which have hotter inputs than others, it's nice to have somewhere between your master fader and your output to be able to match levels.
My own console is a Roland M480 and whilst all outputs are freely patchable, they take on the master gain of whatever bus you strap them across, so a matrix is necessary to attenuate or amplify your output in order to match a sensible level on the desk (one where you can actually see the meters moving, without anything clipping!) with the level in the PA that you desire.
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I find on the yami digitals I hardly ever set up auxes as groups, I just mix to VCAs, but I still like to have Matrices as outputs because of individual EQ, levels, processing, ect. Even a sub aux or mono feed I'll send to a matrix so that I only have to look at 1 page to see all of my outputs.
With the VENUE, it's a bit easier to build separate mixes with groups, and you can really get into detailed processing and control. I like to have as much of the "system processing" from within the console so I don't have to dive over to an outboard computer to make changes. If I'm the only one mixing for the night, I'll do almost everything from within the board.
On the LS9 I have all my outputs on one layer, so it's not a problem for me.
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Without groups you lose the ability to tailor the mix to the location. How do you reduce the band levels in the front fills without groups?
Mac
My front fills are almost always their own dedicated mix, usually only vocals, and acoustics/keys, post fader.
On the Venue (which I'm mixing on more often than not), I'll feed front fills via groups, but it's still generally just vocals and instruments with little/no stage volume.
IMO there is no sense at all in feeding bass/guitars/drums into the front fills, they are always plenty loud on stage. The whole point of the front fills is to fill out the sound for those few rows before he PA opens up. You should be supplementing the stage volume, which most of the time means just vocals
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IMO there is no sense at all in feeding bass/guitars/drums into the front fills, they are always plenty loud on stage. The whole point of the front fills is to fill out the sound for those few rows before he PA opens up. You should be supplementing the stage volume, which most of the time means just vocals
My point exactly.
Mac
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My fronts are always on an Aux. a quick look at sends on fader shows contributions.
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IMO there is no sense at all in feeding bass/guitars/drums into the front fills, they are always plenty loud on stage. The whole point of the front fills is to fill out the sound for those few rows before he PA opens up. You should be supplementing the stage volume, which most of the time means just vocals
One more here. But I've seen people doing the whole PA on the front fills. Then they need to delay the fills to solve phase issues between the drums on stage and the drums on the fills...
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I usually route everything through matrixes- subs, fills, zones as well as L/R. This seems to allow me the most flexibility in my work flow. I will have FF, sub and zones from an aux fed to matrixes, so I can can choose L/R to feed them or individual channels to augment or supplement those mixes.
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I usually route everything through matrixes- subs, fills, zones as well as L/R. This seems to allow me the most flexibility in my work flow. I will have FF, sub and zones from an aux fed to matrixes, so I can can choose L/R to feed them or individual channels to augment or supplement those mixes.
If you mean adding individual input channels to the matrix on top of the main stereo feed you should avoid doing that with a digital console. There will be a difference in latency between the 2 versions of the input that will cause comb filtering, which usually sounds like hollowness.
Mac
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If you mean adding individual input channels to the matrix on top of the main stereo feed you should avoid doing that with a digital console. There will be a difference in latency between the 2 versions of the input that will cause comb filtering, which usually sounds like hollowness.
I use a matrix feed to the output on my Venue for cheapy recorders like my Zoom to pad down the level from the main L/R feed. But since my venue isn't that big and the guitar stage volume warrants minimal guitar in the mains, I take a tap from the guitar subgroup and route that to the matrix as well (which does have a bus compressor inserted). With the delay compensation turned on for for Mix & Inserts, I don't get the combing. Haven't tried it with other desks though.
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My older mixer does not have matrix mixing. I have my system sent out a L/R mix into a Rane SM26 signal splitter.
That way when recording I can keep the mains up and adjust the Rane to control the house. With the church work I am able to run a my FOH and then with two more outputs from the Rane send a second line out to some Behringer Sharks for Delays and then into amps and speakers for keeping the Foh and seconds around the 96 db area and keeping the FOH from being overpowering. The Rane will allow me to keep FOH in L/R and side fills as combined mono. I save the mono output on the mixer for the video tape. The mixer is an older EV M4 32 channel.
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If you mean adding individual input channels to the matrix on top of the main stereo feed you should avoid doing that with a digital console. There will be a difference in latency between the 2 versions of the input that will cause comb filtering, which usually sounds like hollowness.
Mac
Mac, There are some digital desks that effectively compensate for bus path latency. Have you experimented with this on CL series desks? In a quick experiment I couldn't duplicate the hollowness that I'm used to demonstrating on the other Yamaha desks. I'll have to try it again sometime soon.
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Mac, There are some digital desks that effectively compensate for bus path latency. Have you experimented with this on CL series desks? In a quick experiment I couldn't duplicate the hollowness that I'm used to demonstrating on the other Yamaha desks. I'll have to try it again sometime soon.
Happens super clearly on my Si Compact, which I talked with some people around here just a little while ago. I've only ever done vocals only in my front fills, though I don't use them crazy often. Hopefully will more in the future! I've got a little touring gig coming up, might bring the desk.
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Mac, There are some digital desks that effectively compensate for bus path latency. Have you experimented with this on CL series desks? In a quick experiment I couldn't duplicate the hollowness that I'm used to demonstrating on the other Yamaha desks. I'll have to try it again sometime soon.
AFAIK no Yamaha console has latency compensation. The net result of latency compensation is that all signals are delayed to the maximum latency through the console to make up for their shorter latency than other channels.
I don't know what consoles other than Avid do this, but there are probably some, I would doubt many.
Mac
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all the midas digital desks do it
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all the midas digital desks do it
I think Midas (always-on) and Avid (selectable) are the only ones that offer automatic delay compensation.
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There will be a difference in latency between the 2 versions of the input that will cause comb filtering
Interesting. Curious where the added latency would come from though- inputs hit the A/D at roughly the same time, and once they are in the digital domain where does the latency come from? How does this factor in on a console with a near-zero total latency spec. of 0.8ms?
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Interesting. Curious where the added latency would come from though- inputs hit the A/D at roughly the same time, and once they are in the digital domain where does the latency come from? How does this factor in on a console with a near-zero total latency spec. of 0.8ms?
Channels and busses typically have their own EQ, gates, comps,... each of these algorithms consumes additional clock cycles to process the data stream, this processing time is beyond the latency of the AD and DA conversion. Internal FX units (verb, delays, multi-band comps, ...) take even more clock cycles to process.
One can design a console to have uniform delay through any and all paths, but then this built-in 'delay compensation' has to be set to the maximum processing time for the worst case scenario, which increases the overall latency of the console.
How this is addressed is a design choice. I suspect most consoles will always include the delay of the channel strips whether they are engaged or not. Others may also include the bus 'channel strips' processing time, so that whether you go through a bus or not you get the same delay,....
As an example, using an internal effect unit for parallel compression (on drums, guitars, i.e two busses, one with an internal FX unit comp inserted and one bus straight thorugh) is a likely scenario for comb filtering. A better choice might be to use a single bus and a compressor that has a built in mix ratio, so the compressed and uncompressed signal get the same delay.
The takeaway from this: You need to understand, for any particular console you are going to use, how routing and adding effects will change the latency of the data stream. Each console is potentially different so a routing scheme that is safe (= 'comb filter free') in one console might not be safe in another console. Sucks, but thats life in the digital world.
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Interesting. Curious where the added latency would come from though- inputs hit the A/D at roughly the same time, and once they are in the digital domain where does the latency come from? How does this factor in on a console with a near-zero total latency spec. of 0.8ms?
This has been discussed at length over the past few years. the different latency comes from the different digital path length and different processing on each path.
A latency of 0.8ms may be near zero, but it is also non zero. Have you measured the latency on a console and found it to be 0.8ms? Most of the consoles I am familiar with are in the 2ms-2.5ms range, analog in to analog out.
Mac
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What is a quick and dirty way to test latency? Send a signal generator out of a pro tools rig, then back in, measure the latency, then send it out through the mixer and back in, then subtract?
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What is a quick and dirty way to test latency? Send a signal generator out of a pro tools rig, then back in, measure the latency, then send it out through the mixer and back in, then subtract?
That'll work, but don't use the signal generator, use a hand drawn single sample spike. Expect it to come back 'smeared' due to the DAD conversion but pick the first break. This is actually good to know for your ProTools setup by itself. Some DAWS do an automatic round trip latency compensation (i.e. when you are overdubbing this is desirable.) However, they base the compensation on what the audio driver reports which I have found to be inaccurate on many cases. Cubase has a place to enter the 'extra latency'.
Measure your console channel strip direct to LR, Auxes, channel thru a buss, with effects applied as inserts in various places (Ideally insert and then disable he effect, its hard to pick the first break on some effects like verb)....
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I think Midas (always-on) and Avid (selectable) are the only ones that offer automatic delay compensation.
AFAIK Digico also have it. They have a different way if implementing it though. All the processing in all channels/groups is always on and the signal has the same latency regardless of whether the processing is on or not.
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AFAIK Digico also have it. They have a different way if implementing it though. All the processing in all channels/groups is always on and the signal has the same latency regardless of whether the processing is on or not.
I don't know what or how Digico has as far as delay compensation, but having all processing in all the time is not uncommon, and does not compensate for different path length.
Mac
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Whoops! Forgot to mention that the console compensates for different path lengths too! I found this out while talking to one of the digico reps that had come here for a training seminar.
I was specifically interested in this as i wanted to use parallel compression on vocals.
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What is a quick and dirty way to test latency? Send a signal generator out of a pro tools rig, then back in, measure the latency, then send it out through the mixer and back in, then subtract?
Or split the signal and loop one copy directly back into PT while the other one goes through the console. If you have enough inputs to PT you can capture various console paths/outputs at once.
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Or split the signal and loop one copy directly back into PT while the other one goes through the console. If you have enough inputs to PT you can capture various console paths/outputs at once.
I'll try that, doesn't sound too hard.
Easy enough to find out about Digico - I'll try on an SD9 this weekend.