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Apparently there are at least three different kinds of XLR connections in gear on the market now days.
Apparently there are at least three different kinds of XLR connections in gear on the market now days.
If you recall from a previous post I recently sent my Nady equalizer back due to the fact it was described as having balanced inputs and outputs.. but I came to realize the XLR connections were actually unbalanced.. and this was confirmed by the a Nady representative as such:
“..The output is unbalanced. Pin 3 is driven and in opposite polarity to pin 2, but it doesn’t float and is not by definition quasi-balanced (also known as cross-coupled)….” he then went on to advise that in a studio situation it would be best to treat these connections as unbalanced.Again, I emphasize that the advertised specs call the above “balanced”
So, today I emailed them back and inquired about one of their other models which specs showed Balanced I/O, and was about $15 more.. The response was: “… Not true quasi-balanced outputs.”
So, back to ebay looking for a bargain equalizer to replace the one I sent back, and in researching some other brands ( I think it was JBL or Ashly brands) I came to realize there is yet another form of XLR connections called “servo-balanced”
— “Servo-balanced outputs provide automatic output level adjustment to accommodate either balanced or unbalanced feeds”The thing is the specs at a glance all of them say “Balance I/O”, regardless if it really is true balanced or not.. it’s not untill you take a closer look at the literature or diagrams that it becomes evident what the connections really are.
I suppose this type of thing is a new practice?
I wonder why.Anyway, I thought this bit of information might be enlightening to some.
Total posts : 45366With the advent of integrated circuits (IC’s) the use of magnetic transformers as impedance matching devices became too expensive to manufacture and market. Transformers were also limited in their bandwidth and efficiency. Digital circuits operate well above audio frequencies. With “servo” circuits, this meant working a 600 ohm source into a 2-10K load was now an acceptable practice. The 10K to 600 ohm source to load might be required to have an attenuation network to match levels and impedance. Servo Balanced Outputs is the practice of substituting a passive or active component derived circuit for the magnetic transformer.
By using a resistive network, IC’s (essentially current devices) could efficiently transfer audio signals from source to load. Transformers are voltage devices. The resistive load acting as a passive buffer. Additionally, this allows the user to (by using the proper connections from the source to load) connect unbalanced and balanced sources and loads in a variety of configurations efficiently.
That is why learning the load input and source output configurations and impedance can be very helpful in producing very high quality low noise audio with balanced and unbalanced sources and loads. The configurations can be very diverse. the following link may be helpful in figuring this challenge out:
http://www.jensen-transformers.com/an/an003.pdf
http://scopeboy.com/balance.htmlFinally, the advantage of balanced over unbalanced is that balanced lines between devices can be much longer due to unbalanced line susceptibility to induced outside noise and hum. Short unbalanced lines between devices can perform with the same degree of quality of balanced lines. Unbalanced lines tend to operate with lower signal levels under the same load impedance as balanced.
My recording/broadcast studio uses both balanced and unbalanced sources and loads and is considered to be on a par with some of the largest pro studios for quality and noise figures.
Total posts : 45366A bit of confusion about pin lineup for 3-pin XLR plugs has existed for years.
One constant is that Pin 1 is always shield ground, but Pins 2 and 3 have lacked a universal standard for assigning the + and – lines from a balanced signal.
A very good comparison of this situation can be found by looking at schematics from a number of microphone manufacturers. Some put the + signal on pin 2, others on pin 3.
Having a standard way of assigning the pins is important to maintain phase consistency throughout the audio chain.
Several years ago I chose the standard for my installation from that used on the EV 635A microphone, a very popular broadcast mic:
Pin 1 = Shield
Pin 2 = + Positive
Pin 3 = – Negative
Total posts : 45366Carl,
If the connectors at both ends of the balanced line are the same type (XLR, TRS, etc.), it simply doesn’t matter; does it?
When you begin to mix connector types, then wiring configuration does matter…a lot.
And balanced to unbalanced is even more critical because of distortion and noise.
You might find microphones made in the U.S. are, for the most part, standardized. Mic’s made oversees, not so much. On condenser mic’s, (with phantom power) it is critical the conductors go to the right pins.
My studio condenser mics all use an outboard dbx 286a channel strip to keep things isolated and straightened out. It’s a great tool; love it.
This link is a great resource: (you might want to bookmark this)
Total posts : 45366Before I get to business on the subject, this page just went “sproing” and is scattered all over the screen and much off the screen. I’ll go over to the troubleshooting thread and post a repair notice.
Absolute phase for audio is described from this wikipedia excerpt:
“In the realm of high fidelity reproduction of music, absolute phase refers the phase of the reproduced sound waves relative to the original sound waves, or to the relative phase of the various channels of stereo or multi-channel reproduction. In most cases, it is actually a question of the polarity of the channels, i.e., an equal phase shift of 180° at all frequencies. Some audiophiles claim that reversing the polarities of all the channels simultaneously makes a perceptible difference in the sound quality, even though the relative phases of all the channels are preserved.”
That last sentence is key to my bringing this subject into the discussion. SOME audiophiles believe in absolute phase, others don’t think it matters, but a most common group are unaware of the entire question.
I believe I know that it matters.
Notice the contracting of “belief” and “knowledge,” which are usually two different alternatives?
But tonight, in this situation, it is my purpose to bring it up, but from this point it is up to others to dismiss the subject or continue discussion.
Total posts : 45366I’m glad you all posted this.
I must admit that the variations of XLR connectors throw me off, and thought that it would always be as simple as plugging an XLR jack into the XLR plug and the connection would be correct.
But apparently that is not always so.When I bought a Rode NT1 mic, the seller provided a Ross PB23 phanthom power box with it. My original thought was to not use it, and just use the mic inputs of my mixer to power it. But due to what I just read here, I looked over the specs of my microphones again.
The Yamaha mixer mic inputs are 1:ground 2:hot 3:cold
And the NT1 describes itself as: Pin 1 screen, Pin 2+ Pin 3-.
So I believe this says the same thing and it will be fine to connect the NT1 to the mixer.The point is I’m glad you all brought this point up, otherwise I would never have thought twice about inserting any powered mic into it,
Total posts : 45366Audiophiles sometimes deal in the practice of voodoo. By flipping the phase (polarity) of all the channels and having the “belief” that there is some perceptible difference in audio quality is not support by laboratory test data.
In fact, when setting the levels in the stereo generator of an FM exciter, one channel is fed 180 degrees out of phase to test the linearity of the audio chain and exciter audio section and to set the phase and levels for maximum channel separation when transmitting in stereo. The FM decoder uses left plus right (L+R) and left minus right (L-R) to produce left and right channel stereo audio in the receiver. In other words, R-R=L and R+R=R. If you have followed so far, then you will notice there is no reason to believe that a perfectly setup system will exhibit more than an extremely small phase error. Most FM stereo generators are set for maximum separation near voice audio frequencies. This is because phase errors occur more often at high frequencies due to the short wavelength (more difficult to set with simple audio metering). Phase differences could then introduce distortion in the audio and keep the exciter from fully deviating the carrier without over modulating. Usually one channel shows more high frequency distortion (phase error) than the other because of the phase dependent stereo decoder.
I have designed and built dozens of working recording studios, over a half dozen AM radio stations and twice that many FM radio stations before all of this was accomplished in a chip automatically. Flipping both audio pairs has never made a measurable difference in the quality of the audio, unless one of the audio channels was out of phase relative to the other. If the audio levels are properly set, the mono audio fed into the (out of phase) system should essentially disappear at the output.
When using stereo “imaging” processors, there are adaptive circuits that shift the audio phase of left and right channels to give the perception of a wider separation, or image, of channels using what is known as “delay”. The actual stereo separation doesn’t change. And if you were to send this “out of phase” audio signal through an FM transmitter, it could produce audio distortion dependent on the amount of audio phase error.
Absolute phase could, then, only be realized if the audio circuitry was designed with an inherent phase error (sub-standard). Then, when flipping the + and – conductors of both channels, the amp output could then cause the audio to possibly have a perceptible improvement. This could have been the case when magnetic step-down impedance (voltage) transformers were used in the output of tube type stereo audio amplifiers. Today’s quasi-servo output amps do not suffer from this deficiency due to the resistive network approach and differential power supplies used with IC output amplifiers. (-refer to my earlier post.)
This would be my take on what this article is talking about (absolute phase) in relationship to broadcasting and studio design. Some of these explanations are over simplified, and as such, do not contain a complete description of the individual circuits or there performance. Some of the circuits are demonstrated and explained on a previous post on this thread.
50 years ago, flipping the AC plug over could sometimes improve audio performance, prior to the use of polarized AC plugs. This worked, but was still a part of the voodoo.
Total posts : 45366I mostly use Audio Technica 8050 and 8020 condensor mics in addition to several different Shure condensor and dynamic mics.
When I purchased a Focusrite TwinTrak mic pre/compressor/limiter they threw in a Cascade condensor mic as a package deal.
I put the Cascade next to an Audio Technica 8050 and recorded both to two separate tracks. Play back as left/right stereo, the sound was very thin, no bass. Putting either track on separately as a mono playback was fine.
Flipping the phase of the Cascade mic and recording again, the subsequent play back in stereo sounded fine, very full.
The mic cables were checked for polarity and found to be the same so it would seem the Cascade was phase reversed compared to the Audio Technica.
As such when using the Cascade, I always push the input phase reverse for that channel.
Total posts : 45366A sound occurring in acoustic space, let’s say a trumpet, sets up a particular movement in the air. This sound cannot be “phase reversed” while it’s happening in real space/time; it has a “definite phase.”
When an electronic rendition of that same sound comes out of a loudspeaker, the only way there can be an (near) accurate reproduction is if the speaker moves the air in exactly the same direction. If the speaker is 180-degrees out of absolute phase, the thrust of the sound will be backwards, into the speaker enclosure.
There are scientific papers on absolute phase and engineering practices which recognize it as being precise engineering.
In nature the positive peaks of a sound often contain more energy than the negative, and indeed, some ears can notice an improvement, be it somewhat slight, in a correctly polarized system.
Phase is time related. Flipping the phase of a channel shifts the time by a minute amount. Detection is most noticeable on equipment with good transient response. Condenser microphones have better transient response than dynamic mics, and certain loudspeakers have better transient response than others, with cabinets and crossover networks being very apt to contribute transient distortion, along with phase distortion between tweeters and woofers.
What I have noticed is that believers on both sides of the discussion stick to their beliefs dogmatically. I know I do.
Total posts : 45366For all practical purposes, mic cables for the past 30 years or more, are pin2+, pin3-, pin1G. Phantom (DC) power rides on top of those with 6280Ω resistors, returns on the ground. Remember audio is AC so if the pins on a subsequent connector are reversed, you get phase cancellation, i.e., nada. I haven’t run into that problem since I was a kid.
But many pro or semi-pro boards have XLRM connectors for mains out. These are not the same thing. You won’t be able to match impedance without transformers.
Last week a mixing board operator at the local community theatre was asked by the performers to record the show. Since the mono main out was being used for the amp to a flown speaker cluster, he used a pair of short mic cables to connect the 2-channel (stereo) mains outs to a M-Audio interface to a Mac.
Guess what? With the position of the gains and faders, it was impossible to attenuate the signal without cutting FOH, even with the interface gains all the way down, so the poor thing’s clip lights were constantly flickering, sometimes on steady for extended portions. The interface expects a LowZ mic signal when a XLR cable is connected, and sends it to its preamps. Since the mains outs are are balanced line, you have a bad mismatch.
AFAIK, the interface survived, but the recording will be distorted.
In the case of your (former) graphic EQ, a line level signal is expected, so it would have had enough spread to handle it. You wouldn’t plug a mic directly into that unit, because it doesn’t have a preamp, i.e., nowhere near enough signal. Not that you would want to do that, but just so you get the picture about the differences.
Total posts : 45366Yes, KDX, positive and negative pressure peaks associated with phase, I would agree with that.
In the case of my microphone comparison, when the two microphone recordings of the same source material were played back in stereo, the phase difference caused the positive and negative air pressure eminating from the speakers to cancel out and became noticable as a loss of bass. And, I suppose phase cancellation would have been noticable at other frequencies as well but the bass was most obvious.
Imagine a kick drum being recorded by two out of phase microphones. Seems easy to visualize the drum skin pushing out causing an increased (positive) wave front.
That wave front pushes on both microphone capsules. If pushing on one caused an instantaneous positive voltage when referencing the “+” wire to the “-” wire of the microphone connector, the other would produce a negative voltage if wired backward.
Assuming identical response (other than phase) the signals would cancel when equally mixed. Or, in the case of stereo playback, the left/right speaker cones would move in opposite directions causing positive and negative wave fronts that would somewhat cancel out.
As for phsyco accoustics, I’d swear I can hear a difference just listening to the ambient noise in the room when I reverse the phase of the microphone.
Total posts : 45366There have been several situations where I suspected that the AP (absolute phase) of a recording was backwards, and I noticed that open backed (cheap) speakers can be used for (subjective) listening.
The “direction” of the sound, which I’ll call the “main lobe,” is stronger back behind the speaker when wired in reverse, and stronger in front of the speaker when wired correctly.
In order to make this test one must be wide awake, not drinking, and the room must otherwise be quiet.
When fatigued or otherwise stressed, the ears are not so critical.
And what MRAM mentioned about the ambient noise of the recorded material sounding better in AP has also been witnessed here, where AP produces a more “open” sound.
If I can find it, I have an article about a circuit (from memory) from ABC Network that either detects or corrects the phase of a single channel.
I had the pleasure of knowing an FM engineer who participated in the “invention” of FM stereo in the 1950s and he KNEW that AP was important and explained to me why the ear is sensitive to it. I’ll try to find his references.
Total posts : 45366Just a note: If you’re interested, you might want to download Blue Cat’s Gain Suite, Sonalksis, Ltd.’s FreeG. Both are free, and work quite well if you use them judiciously.
http://www.bluecataudio.com/Products/Product_GainSuite/
http://www.sonalksis.com/freeg.htmWhen I record live, I regularly use Blue Cat’s Stereo Widening Gain and Sonalksis’ FreeG in post to make corrections to what the mics have picked up, but I confess I haven’t tried them in a live broadcast chain.
Total posts : 45366“The “direction” of the sound, which I’ll call the “main lobe,” is stronger back behind the speaker when wired in reverse, and stronger in front of the speaker when wired correctly.”
Of course. The drivers are designed to push from the amplitude. It’s a motor (the driver and coil has often been referred to as a motor), and just like any motor, it’s primarily set to turn in one direction. The difference is that it runs horizontally. The AC return voltage has spent energy, and is less. But if you run it backwards, it will not be as efficient, because the motor and especially the cone are for moving sound forward.
In a mic recording (or live reinforcement) situation, phase reversing has advantages in certain situations. The most obvious is one where you want to pick up the buzz from a snare, or the ring of a floor tom, by placing a mic under the drum as well as above it. You can easily see that as the drum is struck the sound below the drum is going the opposite direction of the sound above the drum. If you don’t reverse the phase of one or the other of the mics, you will get cancellation, albeit it not totally (they’re close together), so you’ll likely end up with muffled mud.
When placing mics on stage or in the studio, you want avoid phase cancellation, so you avoid placing mics that will pick up the same sound as other ones if you can. Guitar mic, banjo mic. If the players mics are placed such that they each pickup sound from the other instrument, then the sound of the banjo gets into guitar mic slightly later than the intended guitar because it’s further away, and the opposite true for the banjo mic. The resulting phase mismatch muddies the sound. This is one reason why we use cardioid (directional) mics.
Also the reason most of the new handheld stereo recorders have their built-in mics in an X-Y configuration, and also why M-S stereo gives you stereo image width control.
Total posts : 45366Here’s a good read for those who aren’t yet engineers. It even explains the anomaly concerning XLR audio connections which aren’t actually balanced:
http://www.dplay.com/dv/balance/balance.html#loop
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