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I my experience, if you use single coils and play in venus that are located on old buildings, the xlr can certainly make a difference. Also, if you run long cables, you'll loose less treble frequencies.
Originally Posted by rpjazzguitar
Apart from that, I've found three reasons to use XLR instead of Jack (ts unbalanced) outputs:
1) you need to send less volume, as the xlr inputs expect mic level - depending on your gear, that can be a good thing.
2) even if we're talking about very cheap mixers, I've found the mic preamps present in xlr inputs to help with the sound
3) where I live at least, when the mixer is far from the stage, you have what here i called a "stage box", where you connect rour rig and that box connects everything to the maix mixer - and all stage boxes I found until today were xlr exclusive.
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11-28-2023 06:40 AM
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Is that still the case with (modern?) XLR inputs that have a jack input down the centre? Sorry I'm not tech savvy. Originally Posted by jorgemg1984
Last edited by garybaldy; 11-28-2023 at 10:54 AM.
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That's an excellent question, no idea, maybe it depends on the implementation? Actually my Mambo PA cabs have a dual xlr/jack input, not sure how it adjusts to gain input, since I never had neither distortion neither lack of volume. Originally Posted by garybaldy
Last edited by jorgemg1984; 11-28-2023 at 11:08 AM.
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If the output is truly balanced, the 1/4" jack in the center of those combination jacks is usually a TRS (3 conductor) jack that will send a balanced signal over cables with 3 conductors. Some devices do send an unbalanced signal through a 2 conductor center 1/4" jack, but this is usually a sign that the XLR is also an unbalanced line. You can feel whether it's a TRS or a simple TS 2-conductor jack when you insert a plug. If you feel a detent before it seats, it's TRS. If it goes in smoothly until it's fully seated, it's a TS. Originally Posted by garybaldy
When in doubt, check the specs.
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I’ve plugged my jazz box straight into Originally Posted by rpjazzguitar;[URL="tel:1300710"
a couple of different PA mixer-amp
units just using a normal longish
guitar cable
and It all worked fine ….
but for plugging straight into a Mackie Powered speaker you would probably need use a pre amp such as one of these
Phantom Block
– Quilter Laboratories
to match the levels and impedance correctly
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One thing that occurred to me this morning ---
The problem with using the mic input (from the ME80) was bass frequencies blooming, apparently because of an impedance mismatch.
ME80 output impedance is 2k ohms.
SRM350 input impedance is listed at 94k ohms (and they don't give different numbers for mic and line).
But, if I inserted a simple passive DI box, I could pad the signal and match the impedances, or so I think. And, then I could set the SRM350 to mic. Unless I've misunderstood something it seems like it should work.
If it does, it solves the problem simply, cheaply and with one less wire (no power supply to the DI).
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Not as fancy but my $30 (less when it goes on sale) Beringer DI has worked well so far..
Amazon.com
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The two situations where impedance matching becomes import are: Originally Posted by rpjazzguitar
- transfer of power, like between a final amp and the speaker. Those are typically low impedance in the 2-16 ohm range where the conductors are un-shielded from interference because low impedance is virtually immune to interference.
- in balanced lines; what is balanced (matched) in the various implementations to cancel interference noise is the impedance of the signal conductors.
Mic, instrument, and line levels, and connections between gain stages do not match impedance.
- the output is relatively low impedance going into an input with a high impedance
- minimal power is transferred
- the output with a low impedance resists interference
- the input high impedance accepts the signal without "loading" the source
- loading causes frequency response anomalies
This is why 2000 ohm headphones may be connected to the speaker outputs of an amplifier (power transfer is only a couple of milliwatts). Input impedance for audio is typically in the 50Kohms up to 150Kohms or more to ensure there is a big favorable "mismatch" so as not to load the signal.
This is why most guitar amps have a 1 megohm input impedance so as not to load the pickups which may have an output impedance of 5Kohm to 9Kohms or more.
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Paul, are you saying that the reason the SRM350 doesn't sound good on the Mic setting is something other than an impedance mismatch? If so, what might it be?
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No. A 2 kOhm source into a 94 kOhm input is an excellent match. Originally Posted by rpjazzguitar
The "ideal" signal source output impedance is 0, because the signal voltage is applied across both the source impedance (in this case, the output impedance of the ME80 electronics) and the input impedance of the SRM350. In the real world (since 0 Ohms output impedance is not possible), the input impedance of the driven stage should be at least 10 times higher than the output impedance of the signal source driving it. A 20-fold ratio is even better and the ideal is greater than 50 times higher. As 94/2 = 47, the ME80 / SRM350 impedance ratio is fine and should not cause a problem by itself.
The reason for this is that the goal going from source to preamp (or preamp to amp) is to match and transfer the signal voltage faithfully, remembering that it's only voltage that is being amplified in the preamp stages. The output voltage of the source (pickup, mic, preamp etc) will drop across any load to which it's applied. So it will drop across both the output impedance of the driving stage and the input impedance of the driven stage (which form a voltage divider together). Any voltage drop across the output impedance load will be lost from the signal before it gets to the input of the next stage of amplification. Essentially all of that loss will be as heat from the flow of current through the source impedance. If the signal drop across the output impedance is great enough, it will no longer be strong enough to drive the next stage to its specified output and it will not be a faithful image of the original input waveform (from mic, pickup etc) - i.e. there will be too little volume and it will be distorted.
If the input impedance is much higher than the output impedance, almost all of the voltage will be dropped across the input load (i.e. it will be transfered to the next stage). If a 10V signal is applied from a 1kOhm source to a 1 megOhm receiver, 9.9V will drop across the input load of the receiving stage. If that 10V signal is applied from 10 kOhm source, about 9.09V will be applied to the input load. But if a 10 volt signal from a 1 kOhm source is applied to a 1 kOhm load, it will only drop 5 volts across the output impedance - 5 volts will be lost across the input impedance. Half of the signal voltage will never make it to the input of the next stage and the frequency transfer of the signal will no longer be anywhere close to linear.
Impedances must be matched to achieve maximum power transfer, not voltage transfer. And impedance is frequency-dependent resistance. So the above also affects the linear (i.e. accurate) transfer of audio waveforms. If you plug a 10K guitar output signal into a 10k mic input, half of the signal will be lost in the output impedance of the pickups & circuitry, and the frequency transfer will not be linear - so it will be too quiet and sound as though someone applied the wrong EQ to it (because they did).
Given the specs you cite, I don't think the output impedance of the ME80 and the input impedance of the SRE350 are mismatched. Whatever the problem is, it's elsewhere.Last edited by nevershouldhavesoldit; 11-28-2023 at 06:39 PM.
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Thanks yet again. Suggests that the problem I was hearing was that there was too much voltage for the mic setting of the SRM350. I'll have to hook it all up again and try dropping the signal level coming out of the ME80 to microphone levels by lowering the output level, if that makes sense, and seeing how the SRM350 works set to "mic". Seems like the theory would predict more noise that way, but maybe it won't be noticeable. Originally Posted by nevershouldhavesoldit
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I'm still confused about your problem. The SRM350v3 has their wide-Z circuitry with a variable gain input that should accommodate anything you can feed it from mic to line level. Why you can't get decent volume with normal frequency response at any setting is a mystery to me. The impedances of the ME80 output and SRM350 input are well suited to each other. Something is not right, but I haven't yet heard what it might be. Originally Posted by rpjazzguitar
Is your SRM350 a v3 model or an earlier one? I don't know much about the first generation, so it may not have the input flexibility of the current one.
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It's the very first SRM350 afaik. One knob. Two buttons. The newer ones have some more features -- including WideZ and three channels instead of one. Originally Posted by nevershouldhavesoldit
I tried an Alto TX310. Similar issue. One knob, no mic/line switch. Had to turn it up really high to get any volume (running from the ME80) and the bass frequencies went crazy.
If I feed the SRM350 from the Little Jazz, and I put the LJ volume on maximum, the SRM350, set to "line", is fairly loud. If I lower the output of the LJ and put the SRM350 on "mic", it's even louder, but it doesn't sound as good.
When I turn it up loud, mic or line, the bass frequencies seem to bloom, but worse on mic. Or so I think. My hearing is not normal and that could be part of the problem.
If I put the mixer between them, I run the SRM350 on "line". Channel vol and output level are set to unity, but the gain control is at 10 o'clock or so (I don't know how much gain it's adding) LED output levels are in the green. It gets as loud as I want to play and sounds great. I do roll some bass off at the mixer, so it could be that I'm compensating for any bloom. If I lower the output of the mixer and run the SRM350 on mic, I don't think it sounds as good. Harsher.
It's all packed up for a gig tomorrow, but I'll spend some time and document the results on Thursday.
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That probably explains it. The line input sensitivity (which is unspecified in any document that I can find since you got me interested) is probably just too low and the mic input too high - and they’re fixed rather than adaptive. The v3 model should not have this problem and neither should most other current powered speakers. Originally Posted by rpjazzguitar
I had no problem driving even my JBL 305s with a Zoom MS60b or my old Samson 8x2 mixer.
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A useful trick is to take the full unbalanced level to the SRM350 and make a two resistor attenuator in the mic XLR shell, at the same time unbalancing the mic input. The suggested resistors give 25dB loss into the mic input 8k. Use values that put the gain control in a middle position. Most decent mic inputs don't mind being wired unbalanced.
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I think that makes sense. Mic level is around -50dbu or something like that, SRM350 sensitivity is +4dbu and my ME80 is coming in at -10dbu. The mixer's gain control straightens it out. Originally Posted by nevershouldhavesoldit
The Alto speaker is designed similarly (being a much less expensive item) and worked similarly. I'm now thinking about the Bose S1. The manual doesn't give the specs I want, but does have some language about magically matching the input level. It does not seem to go so far as having a gain control on the input which is separate from the power amp output. But, I can try it out in the store.
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Quote Originally Posted by nevershouldhavesoldit
That probably explains it. The line input sensitivity (which is unspecified in any document that I can find since you got me interested) is probably just too low and the mic input too high...
In natural language we say that something is less sensitive when it takes more to get a result or response, but for the term used here, lower "input sensitivity" means less signal level is required for full rated output. Originally Posted by rpjazzguitar
-10dBV ("V") is 0.316V
+4dBu is 1.228 V
log(0.316/1.228)*20 is -11.8dB just about the -12dB target for clean headroom
What is the loudest average sound level you want to produce?
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I don't know how to quantify it. Loud enough for an outdoor venue with some wind, projecting maybe 80 feet for 100 people? Originally Posted by pauln
Or, as loud as it goes with the Yamaha mixer in the signal chain? Actually, probably not that loud.
Would you explain the -12db "target for clean headroom"? That's a new concept to me. Also why -10dbV? The spec for the ME80 was -10dbu.
Experiment today. On the SRM350 line setting, guitar>ME80>LJ>SRM350, I could get pretty good volume with the LJ turned up further than I would ordinarily think would work. It's an interesting preamp option because I like the sound, it's small and light and it can be used as a monitor or backup. Last night I played a venue with an elaborate PA system. I just brought the LJ and went XLR into the mixer. Turned off the LJ speaker and heard myself through the side-fill. Worked great. Tomorrow I'm going to try to get to GC to try a Bose S1 with the ME80.
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Higher input sensitivity means that less voltage is required to drive the ouput to its rated level (or that the same input will produce higher ouput levels). But rated input level means how much input signal is needed to reach rated output, while rated output level means the highest output level that a device can achieve within its specified performance parameters (primarily distortion). Rated output is a specific value - units of electric force (volts) or power (watts), depending on the stage of amplification. Input sensitivity is a ratio, not a single value. For preamps, it's measured as the voltage required to get a rated output in volts or volt-equivalent units (which is what dBu, dBV etc are). Originally Posted by pauln
So maximum output levels from line / preamp stages are stated in volts or one of the many dB based equivalents (dBu, dBV, dBv etc). Maximum usable input levels are stated the same way. But input sensitivity is stated as a ratio of output to input, e.g. watts per volt. Higher input sensitivity means higher output for the same input. Lower input sensitivity means that more signal is required for full rated output. And there's no relationship at all between input sensitivity and rated output. The designer / engineer decides on these independently. He or she could make a 25 watt amplifier with an input sensitivity of 1 watt per volt or a 250 watt amplifier with an input sensitivity of 10 watts per 100 mV. A preamp could just as easily be made to have a maximum spec'ed line out level of 1/4V or 5V despite having the exact same mic and line inputs.
If an input is rated to accept +4 dB (on any scale), that's the input level at which it reaches its rated output at its rated distortion levels. So a -10 dB signal won’t drive a +4 input anywhere close to its rated output. This is why a consumer level line out won’t drive a pro amplifier to full output, resulting in low output volume at maximum input settings.
It’s also why a pro line out will drive an amplifier with a consumer level line input into severe distortion. The same thing happens when you run a line out into a mic input. The applied input signal level is far too high for the design parameters of the next stage, so it clips and distorts. If the signal source has a gain control or an output trim pot, you can dial its output far enough down to get acceptable sound. But impedance issues may cause excessive distortion even at lower signal levels. The purpose of those "wide Z" input circuits is to sense and adjust input sensitivity to actual input levels to avoid under- or overdriving the stage. It's a modern form of AGC (automatic gain control).Last edited by nevershouldhavesoldit; 12-01-2023 at 01:50 PM. Reason: clarification
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Is this going to be on the test? Originally Posted by nevershouldhavesoldit
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I'm sorry if I seem a bit pedantic about this. But incorrect statements that are left without comment are often accepted as correct by other readers, who then have problems because they acted on wrong information. I was responding to the statement "lower 'input sensitivity' means less signal level is required for full rated output" in post 67. The more sensitive something is, the more it responds to stimuli (in any language). The more sensitive your skin is, the more dramatically you respond to tiny amounts of substances that cause no reaction at all in people with low skin sensitivity. More sensitivity = more reaction to less stimulation. Originally Posted by rpjazzguitar
So the more sensitive your input is, the more output you'll get from less signal.
[The test is open book anyway, so don't sweat it
]
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Open book? You mean we have to think? The prof always said that like it was "this test is going be easy". I always heard, "the book won't help, sue me". Originally Posted by nevershouldhavesoldit
I'm sorry if I seem a bit pedantic about this. But incorrect statements that are left without comment are often accepted as correct by other readers, who then have problems because they acted on wrong information. I was responding to the statement "lower 'input sensitivity' means less signal level is required for full rated output" in post 67. The more sensitive something is, the more it responds to stimuli (in any language). The more sensitive your skin is, the more dramatically you respond to tiny amounts of substances that cause no reaction at all in people with low skin sensitivity. More sensitivity = more reaction to less stimulation.
So the more sensitive your input is, the more output you'll get from less signal.
[The test is open book anyway, so don't sweat it ]
Sorry for the joke about the test. Please don't take it as negative in the least. I deeply appreciate the expertise and patience you bring to this. I went to engineering school back, iirc, just before Marconi, or, Julius Caesar, or somebody, invented electricity. It reminded me of that a little bit (the experience of being baffled by something others seemed to find at about the difficulty level of eating a doughnut.)
But, your efforts have not gone unnoticed or unrewarded. I took my ME70 (same -10dbu) to Guitar Center today and plugged into a Bose S1 Pro Plus. Other than the absence of slots for bread, it's pretty much a four slice toaster. Maybe six. Plugged in an unbalanced 1/4" and it was unfazed and louder than I'd ever want to play. Apparently, it's got the smarts to figure out what you want it to do, sensitivity-wise.
Sounded great too, although as it got really loud, bass frequencies seemed to bloom but I controlled them with the S1's two band EQ. I've noticed the bass frequency thing with every unit I've tried - I have an auditory nerve problem in one ear which might mean what I'm hearing is not what others might hear.
The Pro Plus, $649, has a juicier battery, if that's the technical term for more ampere-hours or whatever the unit would be. It also has wireless send and receive units, neither of which appeal to me. The Pro is the older model, now at $499, and seems to be a good choice.
Unless another epiphany goes for my throat, I'll probably get one. Rig will be guitar>ME80>Bose S1. Small increase in size and weight from the LJ, but big improvement in sound, I think.Last edited by rpjazzguitar; 12-02-2023 at 09:13 PM.
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RP, this has been a fascinating journey to watch you undertake. I remember talking about this problem back in the rec.music.makers.guitar.jazz days (or as I think of it, wreck.music after a certain self-appointed amp guru got involved). It seems like this is a problem which at its root is generated by a lack of true standardization which could just make everything work together seamlessly.
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I guessed it would make more sense if I knew the history of the engineering involved in getting to this point, so I asked ChatGPT. It suggested that consumer audio came first with standardization becoming important probably around the time that component systems came to market. Originally Posted by Cunamara
Pro line level was developed because it was more resistant to noise and interference with long cable runs. It didn't mention the issue of cost, but I suspect that it was cheaper to build consumer line level devices.
The basic problem I've had is that there are two line levels, consumer and pro. They differ in voltage by about a factor of 5. You can't get some powered speakers to go to full volume by feeding it 20% of the voltage it wants. Some device manuals don't seem to have the relevant specs, which makes the whole thing harder.
Then, there was the issue of impedance mismatching, which seems to not be the problem. I confess to lingering doubts.
I'm left with the bass frequency bloom at high volumes, which I can't explain.
And, what to buy? If I get the Bose S1, it turns out that it may be as much as $800. That's the new version (pro plus) and a wireless instrument attachment. Will it really be the equal (or superior) to the Bud/Blu and therefore be a thrifty choice?
The Bose has 2 band EQ whereas the Bud/Blu have 5 band. That could be a big advantage.
The wireless instrument attachment is interesting. It plugs into your guitar and works for about 30'. That enough to go out in the audience and check out your sound. It has another feature -- which is that you can then plug a TRS 1/4" into the jack on that channel -- the other end has to be two mono plugs in a Y configuration -- which you can use as input and output to your pedalboard. Like running the pedalboard in an FX loop. I think that's a well thought out feature for gigs.Last edited by rpjazzguitar; 12-02-2023 at 10:25 PM.
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That’s when I bailed out of there—I was really PO’d at the guy who attracted his attention to the newsgroup. Originally Posted by Cunamara
Danny W.
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