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Author Topic: Automatic Mixers part two the results.  (Read 3668 times)

Bob L. Wilson

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Automatic Mixers part two the results.
« on: June 06, 2012, 11:26:37 am »

Idea was to see what readily available automixer provided the best performance in a subsystem to be used for smaller events in our sanctuary.

Last post on the old thread: Given no clear consensus I think I am going to buy an IRP, a Shure and a Lectrosonics and compare them heads up in our room with our mics and our sources. I think I can buy all three for less than $500 total.

I actually ended up installing and testing three units in detail: IRP VM4083, Shure SCM810, and a Lectrosonics DM84. System setup was as follows a Whirlwind mic splitter with Jensen transformers and a PS48 phantom powersupply were used to split eight microphone inputs out of the existing larger group. The eight consist of two EV RE-1 wireless, two Countryman Isomax goosenecks on the pulpit and lectern, two Crown PZM piano microphones, and two floor jacks. The eight were then routed to both our existing FOH console and to the automatic mixer. A separate scene was created in the system processor a Shure P4800, the aux fed sub input was moved over to an additional Shure DFR22 to free up a physical input on the P4800 for the automixers.

I installed the IRP first. Given IRP has disappeared I had to use the internet wayback machine to look at some of their old web pages to ascertain the method their automatic mixer uses. It is not gain sharing ala the original Dugan patent.

IRP's description: The Voice-Matic® Microphone Mixer is unique in operating principle using Dynamic Threshold Sensing (DTS) to differentiate between active and inactive microphones. Dynamic Threshold Sensing utilizes a combination of the amplitude and history of the signal to determine channel access. DTS generates a threshold reference which decays over an 80dB range from a high level to a low level in a 10ms or less interval. All microphone inputs are referenced to the threshold simultaneously. The first channel whose instantaneous amplitude equals or exceeds the sweeping threshold is given temporary ON status for 200ms. Simultaneously, the threshold is reset high for 7ms and another sweep is initiated. Since any active input will equal or exceed the decaying threshold earlier in its sweep, the average sweep time is only 3 to 4ms and the frequency of the intervals will be increased. On a repeated sweep the same or a different channel may have the highest instantaneous amplitude and receive temporary ON status. In this way, the active microphone channels are detected and their ON status is renewed. Several microphones can have simultaneous ON status and be effectively updated without dropouts. Rather than turning an inactive microphone OFF it is attenuated by an internally pre-set 30dB. The microphone is rapidly turned ON when it receives a signal, preventing upcutting of leading syllables. A continuous count of the number of ON microphones is used to adjust the master amplifier gain according to the rule of reducing gain by 3dB for each doubling in the number of simultaneously active microphones. The automixer may be set to LAST MIC HOLD mode for the continuity of room ambience in the sound system and to other output devices. The LAST MIC HOLD feature keeps the most recently activated input channel held to the ON status until another active input is detected.

I found its automatic circuitry to work quite well with only a single weakness. If a person wearing a wireless stood in front of the pulpit or Lectern and forgot to turn off their wireless the IRP would activate both channels and then reduce the master gain. I appreciated its 100Hz onboard high pass filters, excellent sounding three band EQ and built-in ability to split the mic lines itself, IE it would not have needed the Whirlwind. I also found its onboard AGC IRP terms it Level-matic II to be quite effective with voices ranging from small children to loud adults it did a good job of managing overall levels with a minimum of audible pumping.

IRP's Description: The Level-Matic II AGC incorporated in the DE-4080/4040 mixer performs an important function typically performed by a manual operator in a sound system. The Level-Matic II AGC circuitry automatically adjusts the master output gain to maintain a uniform output signal level for input level
variations of as much as 10dB. The Level-Matic II gain control is based on the loudness versus frequency and loudness versus time response of the human ear. A loud talker causes the master gain to decrease. If the talker stops talking the gain holds as established by the talkers average speaking level. If a quiet talker then begins to use the system, the gain increases to a new value as established by his average speaking level. Two control voltages are created in the circuitry, one circuit establishes a semi-stationary control voltage and another circuit creates a voltage which follows the signal peaks. At any instant, an attenuation circuit is governed by one, but not both, of these control voltages. Gain corrections are made at a constant slew rate to minimize gain “hunting”.


I tried a few different dynamic and condenser mics in the floor jacks all worked quite well for speaking applications. No combination of system settings yielded resonable performance for singing save switching to standard mode, bypassing the automatic function.

I then pulled the IRP and installed the Shure SCM810.

Shures automixer function termed Intellimix is again not gain sharing.

Shure's Description: IntelliMix delivers seamless automatic mixing by combining three separate functions:
1) Noise Adaptive Threshold. Distinguishes between constant background noise (such as air conditioning) and changing sound (such as speech) for each input channel. It continuously adjusts the activation threshold so that only speech levels louder than the background noise activate a channel.
2) MaxBus. Controls the number of channels that may be activated for a single sound source. One talker activates only one channel, even if multiple microphones “hear” that talker.
3) Last Mic Lock-On. Keeps the most recently activated microphone open until another microphone is activated. Without Last Mic Lock-On, a long pause in conversation would cause all microphones to turn off, which would sound as if the audio signal had been lost. Last Mic Lock-On ensures that background ambience
is always present.


I found the Shure to be somewhat disappointing after the IRP and I am a Shure guy. It has direct ouputs again meaning it could have functioned without the mic splitter. It has a 6dB/octave high pass filter with corner adjustable from 25-320Hz and a 5KHz shelving EQ on each channel but it lacks any other low or mid EQ controls. Its automatic section did not do nearly as well with the range of voices that I tested. I set its channels according to the manual and it sometimes missed leading consonants or entire words especially with children or female voices. If the channel level was increased beyond this point it did much better with soft voices but distorted audibly with a loud talker. I engaged the onboard limiter to just to see if it had any effect but it is an output bus peak type limiter only so it could do nothing about the input preamp overloading. The Shure was also operating with only a -15dB channel attenuation it would have been even more uneven sounding attenuating to -30dB like the IRP, based on the Shures's performance with channel attenuation set to infinity. (-15dB and Infinity are the only stock switch selectable options, -10, -20, or -30dB require breaking out the soldering iron). The Shure also has no dynamics processing save its unhappy sounding peak limiter which meant loud talkers were loud and soft talkers were soft. I ended up having to engage the AGC built in to the system processor to even keep the SCM810 in the game.

I pulled the Shure and installed the Lectrosonics DM84.

Originally I had been looking for an AM8TC to represent Lectrosonics but I happened upon a pair of DM84 units and ended up buying them.  Lectrosonics automixer is gain sharing.

Lectrosonics says: The automatic mixing algorithm applies a patented gain proportional algorithm (US Patents 5,414,776 and 5,402,500) allowing each input assigned to a particular output to behave differently relative to the other inputs assigned to the output. In Automatic mode the input applied to the crosspoint is mixed into the output channel using the the Adaptive Proportional Gain automixing algorithm in the normal manner.

The Lectrosonics was substantially more time consuming to set up as it has a digital interface I had to download and wander through. This unit supports on each input:
1) six fully adjustable filters selectable from parametric, high pass, low pass, band pass, low shelving, high shelving.
2) six feedback filters user selectable for static or dynamic deployment
3) Compressor
4) Delay
The outputs offer a similar but even larger array of processing options. The Lectrosonics automixer works very well as good as or better than the IRP. Its powerful input processing allowed me to set individualized compressors for each channel and its totally flexible EQ arrangement supported everything I needed to do. Surprisingly though it could do no better with the piano or singers in auto mode than the others and its "Power of the Mix" level control system does not work nearly as well as IRPs Level-Matic II at maintaining a consistent output level especially with soft input sources.

I chose to stay with the Lectrosonics over the IRP based on its substantially more flexible EQ and dynamics processing capabilities. I ended up using the AGC in the system processor to make for its shortcomings in that area.

Ultimately this was not really a fair comparison in terms of technology as the IRP I used was manufactured in 1998, the Shure in 2001, and the Lectrosonics in 2010. But due to market conditions I paid $50 for the IRP, $200 for a pair! of DM84, and $175 for the Shure. If I had wanted to avoid using a stand alone splitter my choice would have been the IRP. For portable use I would have picked the IRP for its easy to use analog controls. In fact I intend to keep the IRP for children's drama and other events where I am juggling a bunch of extra wireless mic inputs. I was able to resell the Shure for slightly more than I had paid for it.
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John Roberts {JR}

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Re: Automatic Mixers part two the results.
« Reply #1 on: June 06, 2012, 02:31:09 pm »

It seems your time should be worth more than you paid for some of those mixers.  :-[

I'm sure I'm repeating myself, but many of the alternate AM strategies were developed during the time when Dugan's original patent was in force, so they were working around the "good' way, to avoid infringement.

Quote
I found its automatic circuitry to work quite well with only a single weakness. If a person wearing a wireless stood in front of the pulpit or Lectern and forgot to turn off their wireless the IRP would activate both channels and then reduce the master gain.
This doesn't quite add up. With two mics picking up the same signal (roughly coherent ), the AM algorithm would have to turn down the master gain approximately 6 dB to get back to unity, so to perceive a drop in level seems unexpected. The more common complaint with wireless and lectern mics is comb filtering from slight arrival time differences. Is it possible that one of those two mics was inverted polarity? That could cause a more significant drop in level, when both mics are active.

====

I looked at both of those Lectrosonic patents and couldn't tell much from the claims. They seem overly focussed on detecting ambient room noise.
 
I didn't realize AM were so cheap these days. While it seems they could be an inexpensive software plug-on for digital consoles.

JR


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Bob L. Wilson

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Re: Automatic Mixers part two the results.
« Reply #2 on: June 06, 2012, 05:01:07 pm »

It seems your time should be worth more than you paid for some of those mixers.  :-[
This was a project with some of the youth group kids, my time was well spent.

This doesn't quite add up. With two mics picking up the same signal (roughly coherent ), the AM algorithm would have to turn down the master gain approximately 6 dB to get back to unity, so to perceive a drop in level seems unexpected. The more common complaint with wireless and lectern mics is comb filtering from slight arrival time differences. Is it possible that one of those two mics was inverted polarity? That could cause a more significant drop in level, when both mics are active.
I didn't say I heard a drop in level it just sounded less intelligible with both mics on and the master level reduced, undoubtedly due to comb filtering. I am interested to know how Shure's unit which is otherwise pretty mediocre does this so well. it was able to gate on only a single unit for any single source every time.

I looked at both of those Lectrosonic patents and couldn't tell much from the claims. They seem overly focussed on detecting ambient room noise.
 
I didn't realize AM were so cheap these days.

Except the Shures. Why the market is putting a premium on those units over the totally superior Lectrosonics units makes no sense. I may repeat the experiment in the fall perhaps with an Ivie, an IED, and an EV. I know where there is a real Dugan perhaps I can borrow it for a test.
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John Roberts {JR}

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Re: Automatic Mixers part two the results.
« Reply #3 on: June 06, 2012, 05:31:13 pm »

This was a project with some of the youth group kids, my time was well spent. I didn't say I heard a drop in level it just sounded less intelligible with both mics on and the master level reduced, undoubtedly due to comb filtering.
Sorry I read "the IRP would activate both channels and then reduce the master gain" as dropping level.

Dugan gain sharing algorithms pretty much damps combing in that case since slight level differences between the two inputs get expanded to larger differences for the roughly coherent input sources, so depth of comb notches are likewise reduced with unequal inputs. 
Quote
I am interested to know how Shure's unit which is otherwise pretty mediocre does this so well. it was able to gate on only a single unit for any single source every time.
I suspect they describe it in their literature. it's been so long since I looked at them seriously that I don't remember.
Quote
Except the Shures. Why the market is putting a premium on those units over the totally superior Lectrosonics units makes no sense. I may repeat the experiment in the fall perhaps with an Ivie, an IED, and an EV. I know where there is a real Dugan perhaps I can borrow it for a test.

Shure has a well respected brand name and even though they are not a very high tech AM, they work adequately, so are a popular choice for people unwilling to invest the time researching the competition like you have. 

Dugan's original patent has expired to become public domain and free for all to use for more than a decade. So there are more "Dugan" algorithm mixers out there than just his namesake products. He is proud of his and they will cost more than your $200 budget, even used. 

JR
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brian maddox

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Re: Automatic Mixers part two the results.
« Reply #4 on: June 06, 2012, 11:50:10 pm »

Dugan's original patent has expired to become public domain and free for all to use for more than a decade. So there are more "Dugan" algorithm mixers out there than just his namesake products. He is proud of his and they will cost more than your $200 budget, even used. 
JR

ummm.  yes.  yes they will....  :)
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brian maddox
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Jordan Wolf

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Re: Automatic Mixers part two the results.
« Reply #5 on: June 07, 2012, 12:45:01 am »

I prefer to use automixers to augment my existing mixing console, not replace it.  I own and use a Shure SCM810 and find it much better to patch each mic normally into the console (leave them unassigned from any busses), then take the direct outputs into the automixer.  Feed the automixer's main output back into the console, and you're almost done.  I set my dip switches as follows:
1 - Auto (unless I'm ringing out the mics)
2 - Last Mic Lock ON
3 - 0.4 seconds Hold Time
4 - -15dB Off Attenuation Level
5, 6 - Limiter OFF/BYPASSED
7 - Local

For SR, I have no problem with the unit…when recording, however, the gating and opening is much more noticeable.  Fortunately, with my method of routing, I can choose to take the record feed from the automixed return or the individual mics as necessary.  Plus, all my inserts are still usable…

If you didn't try it that way, I think the SCM810 deserves another look, er…listen from you.  Try it and let us know what you think.
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Jordan Wolf
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brian maddox

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Re: Automatic Mixers part two the results.
« Reply #6 on: June 07, 2012, 01:04:33 am »

I prefer to use automixers to augment my existing mixing console, not replace it.  I own and use a Shure SCM810 and find it much better to patch each mic normally into the console (leave them unassigned from any busses), then take the direct outputs into the automixer.  Feed the automixer's main output back into the console, and you're almost done.  I set my dip switches as follows:
1 - Auto (unless I'm ringing out the mics)
2 - Last Mic Lock ON
3 - 0.4 seconds Hold Time
4 - -15dB Off Attenuation Level
5, 6 - Limiter OFF/BYPASSED
7 - Local

For SR, I have no problem with the unit…when recording, however, the gating and opening is much more noticeable.  Fortunately, with my method of routing, I can choose to take the record feed from the automixed return or the individual mics as necessary.  Plus, all my inserts are still usable…

If you didn't try it that way, I think the SCM810 deserves another look, er…listen from you.  Try it and let us know what you think.

i used to use the SCM810 exactly the same way.  i even had one gig where i did this exact trick with 6 SCM810 linked together.  i fed the 48 channels of auto mix from a 48 channel Yamaha PM3500.  took the output from the auto mix rig into a second console along with various playback and lectern mics.  the ability to have 48 open tabletop mics on stage in a convention hall for an audience of several thousand was pretty cool.  and i still used the faders to 'mix' with...
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brian maddox
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Jordan Wolf

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Re: Automatic Mixers part two the results.
« Reply #7 on: June 07, 2012, 11:52:50 pm »

...i fed the 48 channels of auto mix from a 48 channel Yamaha PM3500…
I've got a similar event coming up, but somehow the local company has to rustled up a PM5000-48 for me to use.  At least I get and EZ-Tilt and parametrics on each channel…
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Jordan Wolf
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John Roberts {JR}

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Re: Automatic Mixers part two the results.
« Reply #8 on: June 08, 2012, 10:17:05 am »

When dealing with that many open mics, a linkable AM is really useful.

JR
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Bob L. Wilson

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Re: Automatic Mixers part two the results.
« Reply #9 on: June 08, 2012, 12:48:37 pm »

I prefer to use automixers to augment my existing mixing console, not replace it.  I own and use a Shure SCM810 and find it much better to patch each mic normally into the console (leave them unassigned from any busses), then take the direct outputs into the automixer.  Feed the automixer's main output back into the console, and you're almost done.  I set my dip switches as follows:
1 - Auto (unless I'm ringing out the mics)
2 - Last Mic Lock ON
3 - 0.4 seconds Hold Time
4 - -15dB Off Attenuation Level
5, 6 - Limiter OFF/BYPASSED
7 - Local

For SR, I have no problem with the unit…when recording, however, the gating and opening is much more noticeable.  Fortunately, with my method of routing, I can choose to take the record feed from the automixed return or the individual mics as necessary.  Plus, all my inserts are still usable…

If you didn't try it that way, I think the SCM810 deserves another look, er…listen from you.  Try it and let us know what you think.

I didn't restate it on this thread but the application in question is a fully automatic system where a user can walk in to the sanctuary flip one security switch (power things up) and push a single button (switch scenes on the system processor) to have reasonable quality sound for a small event. Even taking the Lectrosonics out of the picture the IRP is a totally superior unit to the Shure for this application.

The system has proven to work fine, to deal with piano accompaniment and/or a soloist I left the inputs for the PZM mics in the piano and one jack on the chancel floor in bypass mode. We added a lid lock many years ago to keep the piano microphones from walking off so at an automatic sound event a player can't even open the lid. I sourced an EV N/D mic with a switch for use by a soloist. We installed a recycled school locker to store the mics, cords, and stands needed for the automatic system. This way we can grant access any time via email, text or phone call but can also reset the combination, to prevent continuing access. Most importantly a user of the automatic system does not have access to the bulk of the church's microphones and equipment.
I needed a way to make the connections to the automatic system fool proof so I pulled eight Switchcraft connectors apart and had the mounting plates powder coated green. I then swapped them into our floor boxes for every input that is part of the automatic system. To further differentiate the inputs that are part of the automatic system, but not in automatic mode, (Piano L, Piano R, Soloist) I switched out the standard switchcraft inserts with the light green phenolic insulators for some with the black insulators. I laminated an instruction card to the inside of every floor box lid that either states "No automatic sound system jacks in this box!" or shows a picture of a standard jack, an automatic system jack, and an automatic system jack that is in always on bypass mode and needs a mic with a switch.
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