The above Ad will no longer appear after you Sign Up for Free!

Studiomaster Trilogy 326

Discussion in 'Sound, Music, and Intercom' started by Madeen, Mar 31, 2009.

  1. Madeen

    Madeen Member

    Messages:
    12
    Likes Received:
    0
    I'm in the process of creating a reference guide for my high school's tech booth. We have a Studiomaster Trilogy 326 sound console. Honestly, I know next to nothing about how to utilize it. If anyone can give me some basic information that be great. I'd like to go play around with it sometime in the next 2 weeks, and as of now, I can make a couple of mics turn on and off. I'd also appreciate any information on how sound boards are set up and that sort of thing. Thank you. :)
     
  2. museav

    museav CBMod CB Mods Departed Member

    Messages:
    4,017
    Likes Received:
    560
    Occupation:
    Acoustical, audio and audiovisual consultant
    Location:
    Marietta, GA
    Some of the console manufacturers offer resources on their web sites, for example Soundcraft - [Support] (the printed version can be downloaded), but it would probably be well worthwhile to beg, borrow or buy some time from someone who knows how to mix and can get to know your system so that they can present train you and others 'hands on' in reference to your system.
     
  3. hsaunier

    hsaunier Active Member

    Messages:
    158
    Likes Received:
    28
    Location:
    Northwest Ohio
    Madeen,

    Found a picture of the console, but I've had no luck finding a user manual to guide you toward. So....

    In general all channels have the same controls. A place to plug in a source eg. Mic, CD machine ect. Then you control the signal Volume. There should be a knob which says "trim" or "gain" which is the master volume control for that channel. The slider at the bottom of the strip then allows fine control of the channel volume, and a mute swich which will "mute" the channel.

    You will see a series of knobs which adjust the EQ attributes of that channel. From the picture that I can look at I think these would be the series of knobs at the top of the channel strip. By counting the knobs I think you have a 4 band EQ. You likely have a knob at the top which says "Hi" then 2 knobs for "Mid-hi" 2 knobs for "Mid-low" and one knob for "Low". The High and the Low are straight forward. Turn them up or down and the highs and lows get louder / softer. The "Mid" sections allow a little more versitility. one knob should have +/- for gain and the other will have frequency ranges eg..(80-500) (500-2K). This will allow you to give a specific weight to a given frequency in the mid ranges and either add or reduce the gain for the section of frequency.

    The other series of knobs are likely Aux output gain knobs. These would be used to send a seperate mix to anouther output such as a stage monitor. Most mixers will have a set of auxs that can be mixed as "pre fader" which means that only the master gain knob at the top of the strip has an effect on the volume. Then another set which are "post fader" which means the realtive level of that volume of that knob will change every time you move the fader. There may be a button which will change one of those sets of auxs from pre to post.

    You should also have a "pan" knob which can be used in two ways. The most obvious is to move the sound from the left output to the right output or straight up in the middle. The other way a pan knob can be used is to assign a channel to a specific sub master. Can't tell from the pictures I've found, but looks like just above the fader are a series of push buttons. They may have 1-2 and 3-4 and one which says stereo. This is how you assign a channel to an output/sub master. If you want to place a set of mics which are of a similar source such as a drum kit you can push down all of the 1-2 buttons for the drum kit and then turn the pan knobs all the way left and that will assign the drum kit to the submaster 1.

    One other button that you may be able to push is a "PFL" or "Cue" button. That will allow you to listen with a pair of headphones that specific channel.

    I may not be exact in my descriptions but this should be a starter. Again I was not able to find a manual and the picture I was using could not zoom in very close.
     
    Madeen and (deleted member) like this.
  4. gpforet

    gpforet Active Member

    Messages:
    156
    Likes Received:
    24
    Location:
    Atlanta Metro
    It has become a trend that many consider the trim/gain control to be a "master volume" for the channel. That is, in fact, incorrect. As with all other audio work, proper gain staging will have a direct impact on the quality (both in terms of headroom and noise floor) of the output.

    Additionally, improper use of the gain/trim control will have an adverse effect on the ability to properly use/control aux/effects/monitor sends associated with that channel, as well and any outboard processors being connected via the insert patch point of the channel since the insert send is taken after the grain/trim control.

    The gain/trim control is used to adjust the level of the incoming signal so that the rest of the channel and board are operating at the appropriate voltage/gain levels. Typically, the control is adjusted by connecting an input to the channel, applying signal (be it microphone, direct box, or other source) and then adjusting (trimming) so that the signal downline of the control is operating at or near 0-bd. This can be done by either watching the peak light on the channel and increasing gain until the peak light just barely flickers when a strong signal is applied, or soloing the channel to the vu meters of the board and adjusting trim until the meter indicates 0 db during peaks. Too high a trim level and headroom is sacrificed as internal amplifiers downline clip and produce a distorted signal. Too low and the noise floor rises. With the current crop of low noise preamps available in today's consoles, increasing noise floor by operating trim at too low a level is far more preferable than overdriving the console by too high a trim level.

    This approach is applicable to both analog and digital consoles with the caveat that most tend to avoid at all costs going above 0-db due to the completely unmusical nature of digital clipping.

    Although many want to train operators by teaching just what is needed to make the mics or lights work. Far too often, these operators leave with incorrect imformation and both current a future shows suffer. Additionally, these incorrect assumptions are then often taught to the next crop of students.

    Without proper instruction regarding frequency, dynamic range, and gain staging, there is simply no way to teach someone how to properly operate a console, or sound system.
     
    Madeen and (deleted member) like this.
  5. museav

    museav CBMod CB Mods Departed Member

    Messages:
    4,017
    Likes Received:
    560
    Occupation:
    Acoustical, audio and audiovisual consultant
    Location:
    Marietta, GA
    I think this just goes to show that there is no one "right" way. Some people do set the trim to get a nominal 0 level for each input and I used to do it that way but then sometimes ended up with faders all over the place. Some set the trim so that they get the desired post fader level for each channel with the fader set at 0 or some other nominal starting point, this often results in lower trim settings. Some do either of the above and then lower the trim level to account for potential summing where signals may be common to more than one input or for gain applied via EQ. Others may have some other approach. None of them are necessarily right or wrong but you have to be aware of which approach is being used as they can all result in slight differences in the console operation.

    We all need to be careful of the use of dB in relation to consoles as you can be dealing with absolute analog levels (dBu or dBm), absolute digital levels (dBFS) or relative levels (dB with no reference). For example, a "0" level on a console meter may be indicating -20dBFS or 0dBu or +4dBu or some other value, all it is really reflecting is 0dB relative to some reference level that may vary from console to console. And then there is whether you are looking at peak or average levels. As gpforet alluded to, this can be a major shift when going from analog to digital as peak levels for analog consoles are often around +24dBu with +4dBu as a nominal operating level while once in the digital domain 0dBFS represents the absolute maximum level and -12dBFS to -20dBFS may be more typical operating levels. So 0dB related to analog may mean you still have 20-24dB of headroom where in the digital world it means that you have no headroom. And 0 on a digital console may represent a +24dBu analog signal while 0 on an analog console often represents a 0 to +4dBu signal.

    This is not meant to get overly technical but rather to show that when discussing specific settings or values one often has to be careful as what applies to one console may not apply exactly the same to another.
     
  6. gpforet

    gpforet Active Member

    Messages:
    156
    Likes Received:
    24
    Location:
    Atlanta Metro
    I understand, and I've seen quite a few engineers wrestle a board using the trim to allow them to set their fader around zero. The problem is this approach affects the drive to all the sends for that channel. So, if I adjusting trim to keep my fader at a certain location, then I am also affecting the drive to the monitor/effects/aux send for that channel. Not a big problem when I'm running a split to a separate monitor console, but if I'm driving monitors from FOH then I'm playing havok with those sends. I'm also changing the drive to my outboard compressors, meaning any time I change the trim, I have to reach over and change the threshold on the comp for that channel. I like being able to have a pretty good idea I'm sending 0 to the comp, and that setting the comp threshold to -6 is, in reality, -6 to the fader.

    Regarding the faders, I use them to my advantage to get a visual of relative loudness on the stage. If my vocal fader is pushing +3, and my guitar fader is at around -30, then I've got a pretty good idea that the stage is getting hosed down with loud guitar, which then helps me to make decisions about getting those gate threshold a little higher.

    But, bottom line, if my trim is not being used to set gain structure relative to 0, then I really will have no idea what level I'm sending further down the path, so, then I can forget about managing headroom or properly gain staging my outboard racks. Granted, lots of this is a moot point if I'm driving a digital console using all onboard dynamics processing and a stage split and separate monitor console.

    I spend much more time on analog boards, and while the majority of the time I'll have a split to the monitor beach, I'm still driving loads of outboard, with perhaps 10 or more channels of comps on inserts, and it's a lot easier for me to set my comp thresholds to 0, trim my channel inputs to 0, and the lower my thresholds as needed to squash any glare, then grab the make-up gain on the comp while soloing to bring my level back to 0 before it gets squirted out to the effects sends. That way I'm not fighting to match levels futher down the line.

    I guess maybe it is a preference, but if the goal is to optimize headroom (both in the board and thru any outboard hardware), lower the noise floor, and prevent clipping at the same time, then using trim just so my fader will line-up will not get me there.
     
  7. hsaunier

    hsaunier Active Member

    Messages:
    158
    Likes Received:
    28
    Location:
    Northwest Ohio
    Looks like my over simplified approach to explaining mixng consoles has opened an interesting line of discussion.

    So Madeen, here is how I begin to set up the gain on a console.

    1. I confirm the master faders in the center of the console are at unity or 0 and are not muted.

    2. I make sure the channel I'm going to work on IS muted prior to plugging anything in.

    3. I confrim that the channel I'm working on has been assigned to the stereo faders. This should be the buttons located just above the fader, that allow you to assign a channel to the sub group or the stereo faders.

    4. If each channel does not have it's own meter, I push the "PFL / Cue" button so the channel is assigned to a visual meter.

    5. I confirm that the trim/gain knob is turned all the way down.

    6. While still muted if have the source start to make noise. (CD, Drum, Voice)

    7. Look at the meter. If the signal is not peaking I unmute the channel and start to slide the fader up until I reach 0, if the signal is still not loud enough for the room, you can now start to turn up the trim/gain to get the desired volume. If however the volume is too loud for the room and you have not started to bring up either the fader or the trim/gain you may need to look for a "pad" button near the trim/gain knob. By activating this you can reduce the amount of signal coming into the headamp.

    Typically once sound check is done you should not use the trim/gain knobs to adjust levels. There are times it becomes neccessary, but I think at this point you should get an experienced person in to start covering all of those contingencies.

    Don't get real hung up on mixing by meters. Use your ears they are the best meters.
     
    Madeen and (deleted member) like this.
  8. Madeen

    Madeen Member

    Messages:
    12
    Likes Received:
    0
    Wow! Thanks for all the information. I'm going to be able to get into the control booth next week, and I'll mess around with the console and see if I can't figure some this out. I wish I could get down there today or tomorrow, but the 8th grade play is on the 3rd and 4th, and I don't want to screw with their setup.
     
  9. museav

    museav CBMod CB Mods Departed Member

    Messages:
    4,017
    Likes Received:
    560
    Occupation:
    Acoustical, audio and audiovisual consultant
    Location:
    Marietta, GA
    Sorry for the long post but I think it is a great example of there often being "more than one way to skin a cat."


    I disagree, but maybe it should be clarified that I am not talking about mixing with trim, I don't agree with that approach, but rather not necessarily setting the trim to have every input at 0. If having the trim set to have 0 at the input for all channels results in some channels being run at +5 on the channel faders and others at -30, then there may be some advantages to adjusting the trim level to reduce the deviation in the channel faders (and likely other send) levels.

    Your logic and approach seem to be much the same as mine once were, but after having some things pointed out by others I started to realize that there maybe it was not always the best approach. The most critical realization was that I was often adding gain to get input signals to 0 only to then attenuate that signal everywhere, which is basically the same as adding less gain at the input and less attenuation elsewhere, but with the latter allowing more headroom at the input.

    I should also clarify that I don't always set all the faders at 0, for example I may have the guitar and vocals set a bit lower to allow more room for solos. However, having the faders starting around 0 seems to have some advantages. For one, faders and pots are usually not linear over their entire range and typically the further from 0, the more change in level from the same physical movement. Keeping all the channels well intoi the linear range of the faders and you not only have more finite control but all the channels also operate similarly, making mixing easier for many. Another potential advantage occurs if you use VCAs. If you have large variances in individual channel fader levels assigned to a VCA then individual channels can hit their maximum or minimum range long before others, potentially limiting the effective range of the VCA. Sart with all the channels with a more similar setting and there is less chance of this happening.

    I also realized that higher input levels increases the chances of running into issues when you start mixing signals. Most metering occurs after faders or pots, so the level on a bus before such controls is essentially the metered level along with the control setting (e.g. a metered 0 level with the related pot or fader at -10 means the level on the bus prior to that control is +10 and not the 0 shown on the meter, which is after the control). Using your example of monitor (aux) sends, mix 16 channels with similar frequency content and the resulting signal could be up to 12dB greater than the individual channels, so with the individual channels at 0dB you might actually have quite a bit less headroom on that aux bus than you think. To compensate for this you might attenuate the level at each aux send, but that then goes back to the concept of instead making the signals lower to start with and then not having to attenuate as much.

    As far as outboard effects, inserts and direct outputs, I understand what you are saying but again, I am not suggesting mixing with the trim or even adjusting it once it is set. Also keep in mind that maximum input and output levels of the external devices and console may not always match and you may not want what is effectively unity gain from the console to an outboard device and back. Another approach to devices on inserts would be to set the console trim to give you the desired level into the outboard devices but then adjust the output level on those devices to provide a return level that allows the channel fader's nominal position to be around 0.

    The point is that both approaches have their pros and cons. It is understanding how they work and the advantages and disadvantages of each that is important ao that they can be properly applied depending upon the specific application.
     
  10. Madeen

    Madeen Member

    Messages:
    12
    Likes Received:
    0
    Hsaunier: Okay so, I took the stuff you wrote down with me. You were right about most of it. And now I have follow up questions! So, bear with me.

    If I have a 4 band EQ why did you say there is 1 Hi, 1 Lo, 2 Mid-Hi’s, and 2 Mid-Lo’s.
    The set-up looks like this:

    [​IMG]

    So, doesn’t it only have 1 Hi, 1 Lo, 1 Mid-Hi, and 1 Mid-Lo?

    Then just above the fader the push buttons are mute, L-R, 1-2, 3-4. There wasn’t a stereo button, unless that is the L-R. If not, what is that?

    You said that I push the PFL button it will let me listen with headphones. It looks like that’s true because it says Phones next to it, but when I hit it the music was still coming from the speakers. Do the headphones actually need to be hooked up for it to work like that? I couldn’t find any in the control booth that fit it.

    Okay now this is the part that I’m having the most trouble with. Warning: My auditorium has a rep as having a bad sound system because when it was renovated they put the material for sound amplification in all the walls. So, that may be more my problem than anything else. That being said, everything on your list of steps to setup gain worked except number 7. That part of the board looks like this:

    [​IMG]

    The signal light turns on, but the peak light does not. I tried turning the gain and moving the slider, but nothing worked. Additionally, if you move the slider more than a few millimeters, the sound becomes almost too loud, not to mention moving it to 0. You mentioned a pad button I did not see anything labeled that. There was a button that said something like 100 hertz, I think. I forgot to write it down. But it didn’t seem to have an effect.

    Sorry for the length. Thanks for the help.
     
  11. hsaunier

    hsaunier Active Member

    Messages:
    158
    Likes Received:
    28
    Location:
    Northwest Ohio
    OK Madeen,

    Answer to your question about the channel EQ.

    Since there are only 4 knobs it is likely that you have 1 mid range sweep which means that you have a 3 band Eq. There should be one knob in the "mid" area that is "Gain" which should have a -0+ at the 12 oclock position. then -15 at 7 o'clock and +15 at 5 o'clock. The other knob should have frequency numbers aound it ranging from 100hz - 5Khz.

    As for adjusting the gain in step 7. The reason I look at the peak light prior to unmuting is to make sure that I'm not going to blast the speakers with sound when I unmute the channel. Typically you don't want to see the peak light come on because that is an indication that too much signal is passing through the channel. You do want to see activity on the green signal light. If there is no pad button just make sure that the trim/gain knob is turned all the way down prior to unmuting the channel. Slowly take the fader up toward 0. if you get to -30 and the sound is right for the room then just stop there.

    Feel free to give me a call if you need further help.
     
    Madeen and (deleted member) like this.
  12. Madeen

    Madeen Member

    Messages:
    12
    Likes Received:
    0
    What exactly is a graphic equalizer, and what is it used for?
    We have an Ashly 4.24G 4 channel graphic equalizer/ processor.

    We also have a studio effects processor, and I'm wondering if anyone else has and actually uses one on community productions like the ones we'd have here.
     
  13. museav

    museav CBMod CB Mods Departed Member

    Messages:
    4,017
    Likes Received:
    560
    Occupation:
    Acoustical, audio and audiovisual consultant
    Location:
    Marietta, GA
    Try the handy ControlBooth glossary, http://www.controlbooth.com/forums/glossary/11046-graphic-equalizer.html. Your unit is a little different in that it is a four channel digital unit that in addition to 28 band, 1/3 octave equalization for each channel also provides a compressor/limiter, delay and 24dB/octave high and low pass filters for each channel. This could be very useful in many systems and might often be used between the console outputs and the amplifier inputs, we'd have to know more about your system and its configuration to know where it might be most effective for you.

    An effects processor can be handy to have. You might use it for vocals, perhaps inserted on a group or aux since you only have one, or for special effects.
     

Share This Page