Posts Tagged ‘Sonorisation’

Get the right cable to connect your iPhone to a mixer

In Accessories, Gears, Sonorisation on June 28, 2016 at 7:04 pm

iphone-7If you don’t connect your iPod to your mixing desk with the right cable, you’ll lose half your sound!

You may think that when you buy an iPod, all you need to do is load it with a bunch of backing tracks and simply plug it in to your PA system and you’re all set and ready to go…

Well, actually that’s quite correct – it really is that simple to connect an iPod to a PA system, but you would be surprised how many people write to me every week who have done this but find that they get either a very poor sound quality and/or low volume levels, and they can’t figure out what they’ve done wrong!

The wrong way…

Nine times out of ten, the mistake that I find singers make when connecting an iPod to their PA/mixing desk, is that they use the wrong type of cable and/or only use ONE channel on their mixing desk (usually they’ve tried to take a jack out of the iPod in to one spare channel of their mixer). Remember, the iPod is stereo so needs TWO inputs on your mixing desk.

I often hear of singers (wrongly) using a single jack to jack cable (usually with a 3.5mm jack at one end and a quarter inch/6.3mm at the other). This is NOT the correct cabling for connecting your iPod to your PA/Mixing desk!

Just think about it for a moment…when you buy an iPod it comes shipped with a set of earphones which consist of ONE 3.5mm jack going to TWO earbuds. So, this means that the iPod is designed so that ONE jack takes the audio signal out of the iPod, but this signal goes in to TWO places (ie your left and right ears). So, if you want to connect the iPod to your PA/mixing desk, you need to apply the same cabling configuration – ie use two input channels on your mixer. The mini-jack/earphone output of the iPod is stereo which means it sends out both Left and Right channels (which is why you have two earphones of course)! So, as you can imagine, if you connect a cable which has only one jack at the mixer end and is plugged in to one input channel of your mixing desk, you will only get a partial sound (ie only one side of the iPods output).

Even if the jack you use at the mixer end is a stereo jack, the input channel on your desk is mono, so you will still only get half the sound. It’s worth noting that even if your desk is mono and/or you run your amplifiers or PA in mono, you still need to use TWO inputs of your desk for the iPod because the issue is not one of whether you work in stereo or mono – the issue is that stereo actually simply means two channels (Left & Right), and in order for you to hear the FULL sound of any stereo device you connect, you need to feed a Left and Right signal to it, regardless of whether your PA system is set up in mono or stereo.

The correct way…
To feed the FULL sound from your iPod in to your PA/mixing desk, you need a stereo cable which has a stereo mini-jack (3.5mm) at one end, and TWO 1/4″ (6.3mm) jacks at the other end. You could also use a cable which has a stereo mini-jack (3.5mm) at one end and two XLR jacks at the other end if your mixing desk uses XLR connections. Both cables will do the same job – just pick whichever cable suits your mixing desk inputs.

By connecting your iPod in this way, the full signal (Left & Right) will come out of the 3.5mm jack in your iPod and the two jacks at the other end (XLR or 6.3mm) will feed both those Left & Right signals in to two channels of your mixing desk – resulting in a full sound with nothing missing.

Troubleshooting
If you are using the proper cable and have correctly connected your iPod to your mixing desk, you should now have a perfect sound and no more problems. However, if you are still experience sound problems (and don’t know what’s to blame), I’ve devised a fairly simple test which should help you troubleshoot your system.

Even if you don’t have any problems and you are quite happy that your cable, iPod and PA are all working just fine, I still recommend you run this simple test – it’ll only take you a couple of minutes (and you may just discover that not all is as well as you thought)!

The test uses a process of elimination to troubleshoot the problem, and goes through each possible problem area step by step, so make sure you complete the test in the proper sequence using the steps below:

Step 1: Download the mp3 audio test file  and then transfer it in to your iPod, just in the same way you would download and transfer any other mp3 backing track file to your iPod. The audio test file contains a voice speaking the words “left channel” and “right channel”. When the voice says “left channel”, you should hear it coming out of the left channel ONLY. When the voice says “right channel”, you should hear it coming out of the right channel ONLY.

Step 2: Check that your iPod is putting out the correct audio signals by connecting earphones/headphones to your iPod and “playing” the audio test file. Do you hear the “left channel” and “right channel” spoken audio coming through the left and then right sides of your earphones? If the answer is “yes”, then you have now established that your iPod is working properly and is correctly giving out proper left and right audio signals. Now move on to step 3.

Step 3: Plug your microphone in to the first channel of your mixer/PA that you intend using for your iPod and speak in to it to check that the channel is accepting sound and producing audio. Now plug your microphone in to the second channel that you intend using for your iPod and speak in to it to check that this channel is also accepting sound and producing audio. If you can hear yourself talking through your microphone when it’s connected to each channel, then congratulations, you’ve now established that your mixer/PA is accepting audio signals on the two channels you’ve earmarked for the iPod and you can move on to step 4.

Step 4: Connect your iPod, via the cable, to the two channels of the mixer you just tested in step 3. Remember, you may need to adjust (ie reduce) the gain/trim control before connecting the iPod because the input level of the iPod will probably be much stronger than the input level of the microphone you used to test these channels – you don’t want to end up with a distorted sound! Now play the audio test file from the iPod. You should clearly hear the spoken words “left channel” and “right channel” coming out of your PA speakers. If you are running your PA in mono, the “left channel” and “right channel” spoken words will obviously come out both your PA speakers and if you are running your PA in stereo, you will hear the “left channel” and “right channel” spoken words come out of their respective, separate speakers.

If the above test fails at step 1, you haven’t downloaded the audio test file properly or transferred it to your iPod properly.

If the test fails at step 2, your iPod or earphones are faulty.

If the test fails at step 3, your microphone or one or both channels of your mixing desk are faulty.

If the test fails at step 4, then your cable is faulty.

Setting the iPod volume level
The other thing that singers often write to me about is “low signals” or “low volume” problem. There is a simple solution to this…turn up the volume! OK, OK, I know…if it was that easy, everyone would just do it and wouldn’t be writing to me and asking about it! And the reason many people don’t know how to set the iPod’s volume control is because it is kinda “hidden” (by that I mean that it is does not have a master volume control in its settings menu as you would expect). It does have a “volume limit” control in the “settings” menu (which I recommend you set to just a little below maximum) but to set the main volume, you need to have an actual song playing before you can do this (a bad bit of designing Apple – shame on you)!

So, to adjust the main volume, when a song is playing, turn the click wheel clockwise and this will increase the volume. I like to keep this volume level just a little below maximum too. The reason I suggest this is that this volume setting will give a fairly high output signal so you don’t need so much gain at your mixing desk (and it also maximizes the signal to noise ratio keeping background noise down and giving a much cleaner signal).

And, while I’m on the subject of setting the correct gain on your mixing desk, I must stress very strongly that you MUST make sure that you set the controls on your mixing desk correctly before you connect the iPod to it, especially the gain control (sometimes known as the “trim” control). If you don’t know what I’m talking about, then it is essential that you read my article on setting volume levels BEFORE you go plugging your iPod in to your PA system!

 

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Eventide’s Mixing Link™

In Effects, Gears, Sonorisation on June 28, 2016 at 6:47 pm

 

1600-MixingLink_detail04Published April 2014, by Paul White

Eventide’s Mixing Link is one of those products that seems so obvious that you wonder why nobody has done it before. At its simplest, it is a microphone preamp, complete with switchable phantom power, built into a stompbox along with a footswitch‑controlled effects loop for the connection of effects pedals or processors. This format makes it ideal for live performance, as vocalists can now create their own pedalboard of vocal effects in much the same way as guitarists do.

The Mixing Link accepts microphone, instrument or line‑level inputs, and includes some additional I/O and routing options that extend its flexibility. For example, it can also be used to switch a guitar between two different amplifiers, using the effects send as the second output; or, conversely, by using the effects return jack as a second input, it could switch between two sound sources connected to one amplifier. It also works well as a stand‑alone mic preamp, offering up to 65dB of gain for feeding into the line‑level input of a recording system. The headphone output may be used for vocal monitoring, silent rehearsal or simply as a studio headphone amp.

Chain Links
SeitenansichtBuilt into a die‑cast box with an attractive top panel and powered either from a 9V battery or from the included universal‑voltage external adaptor — which is required if you need the phantom power — the Mixing Link uses a ‘combi’ XLR/jack for the mic/line input, with a separate jack for the instrument input. A three‑way toggle switch on the rear panel selects phantom power on or off, or battery operation. A small push switch selects between high and low input‑gain modes for the mic input: high would be the normal setting for dynamic microphone use, though loud vocalists working close to the mic might get by on the low gain setting. The two level‑setting LEDs will let you know which to use. There’s also a ground‑lift switch, though this will probably only be needed if an input source that is already grounded (such as the preamp output of a guitar amp) is connected to the unit. Inside the case is another switch for setting whether the footswitch kills the connected effects dead or whether it allows any reverb/delay tails to continue to their natural conclusion. In other words, it kills either the effects return or the effects send.

The Mixing Link uses jacks for the effects send and return points, which may be used either balanced or unbalanced. The main output is on a balanced XLR, and there’s a recessed switch in the base of the pedal to select line or DI level. A further jack output is present, for sending the signal to an amplifier via the amp/phones level control (which also controls the level of the mini‑jack headphone out). Finally, there’s a bi‑directional Aux I/O connection. This is a four‑conductor TRRS mini‑jack socket that can operate as either a consumer‑style stereo input or a mono output.

eventide_link_tylBy connecting the Aux jack to a device such as an iPhone or iPad running effects apps, it is possible to use your choice of apps rather than conventional pedals. The same jack could alternatively be used as a recording feed to a suitable device, and it is also possible to play stereo backing tracks into the aux input (which remain in stereo in the headphone output), where they will be mixed with the main input. Any signal present at the aux input is also fed, in mono, to the amp, effects‑send and main outputs, though there’s no gain control for the aux input so levels must be controlled at source. All the circuitry has plenty of headroom, with the 500kΩ‑impedance instrument input able to accept signals up to +10dBu, and the line input up to +24dBu. The outputs can also manage up to +10dBu.

A pair of semi‑recessed miniature toggle switches on the top panel allow the footswitch that controls the effects loop to be set to either latching or momentary action, and for the central control knob to adjust the level of only the effects, effects plus dry signal, or the wet/dry mix. The headphone monitor output and amp signals are controlled by the knob on the left, while the knob on the right sets the input gain. Status LEDs show that the pedal is powered up and that the phantom power is switched on.

In The Mix
Clearly, a lot of thought has gone into making the Mixing Link as versatile as possible — the over‑used term ‘Swiss Army Knife’ is thoroughly deserved in this instance! It can be a DI box with ground lift, a signal source selector, an output switcher, a mic preamp, an interface for a smart‑phone recording system or the heart of a vocalist’s live effects setup. It could also be used as a headphone amp, to allow a guitar player to switch an entire chain of effects pedals on and off from one switch, or to switch a signal between two different destinations. It can even be used to mix backing tracks with a vocal or instrument input.

As a mic preamp, the Mixing Link has plenty of clean gain on tap for use with typical capacitor and dynamic mics. Even ribbon mics shouldn’t be a problem, as in DI mode you can apply more gain at the input of the mixer or other device into which the Mixing Link is plugged, if the existing 65dB isn’t enough. I couldn’t really hear any significant subjective difference between the preamps in my audio interface (which is a good one) and the Mixing Link, so I’d have no qualms about using it for recording. However, the most valuable aspect of the Mixing Link, at least for my own applications, is the one first mentioned, namely its ability to connect to and control live vocal effects via its send/return loop. The mode I tried first was with the switch set to give 100 percent dry signal pass‑through, and the middle knob adjusting the level of added effect — which, for vocals, often comprises a combination of delay and/or reverb. The other modes are equally useful, letting you pass the whole signal through an effect such as compression or even distortion (in which case external pedals need to be set to 100 percent wet, of course).

I experienced no noticeable added noise when connecting third‑party pedal‑style effects via the loop, and everything worked predictably and cleanly. Even using an iPhone to generate the effects worked fine, though you have to pick apps that can use the phone’s existing I/O and not the ones that rely on specialist audio adaptors if you want to use the direct mini‑jack connection. Some apps also add a bit of latency, though if you’re using them simply to add delay or reverb to your dry signal that shouldn’t be a problem. The dry signal always remains pristine in this mode, as it never passes through the connected effects.

Ultimately, the Mixing Link achieves everything it sets out to do with minimal fuss and with more than its share of style. Being picky, I’d say that some of the legending, especially that on the sides of the case, is difficult to read, but in all other respects the Mixing Link is a professionally designed piece of kit with a number of ‘save the day’ applications in addition to what I see as its primary function as a mic preamp with switchable effects loop.

Alternatives
The only practical alternative I can think of that might offer similar functionality is to buy a small mixer. There are several affordable small‑mixer options, but nothing so compact as the Mixing Link, and of course mixers don’t usually have built‑in footswitches to control the effects loop.

more review here: Review – Eventide MixingLink

User manual here also:  MixingLinkUG

EQ settings for Jazz guitar

In réglages, Recording, Sonorisation on May 28, 2016 at 8:20 am
from Guitar.com

A Few Tricks You Might Need to Try First

Before I get to the EQ settings for recording great jazz guitar, let’s clarify a few things: We are dealing with two basic musical styles known as “jazz,” straight-ahead jazz and smooth jazz. In general, the straight-ahead jazz guitarist uses an archtop, hollow-body, “F” hole guitar. But, so do many smooth jazz players. OK, as usual, there are no rules. On my CD Bebop, I used a solid-body guitar to eliminate frequency build ups and suckouts. More on that as things unfold.

In most cases the guitar will be a wide archtop, hollow-body, “F” hole guitar. This style of guitar typically creates frequency build-ups on certain notes. The amp can cause this as well but typically not as much as the guitar. When EQ’ing, keep this in mind. If you hear a low note “jump out” in volume, try to find this frequency by using the “sweep frequency technique” explained in part 3. If your mixer has parametric EQ or you have a parametric equalizer that’s great since the odds are good we will use the “notch out” concept using a very small “Q” (meaning frequency width) to eliminate the problem frequency or frequency range.

If the guitar player will be playing fairly loud, causing accidental feedback from the guitar to the amp, try putting a piece of scotch tape over both “F” holes. That is what George Benson has done on some of his guitars.

Here is another trick, best for a small body “F hole” guitars and not for big archtops: For years I played a Gibson 335 for many styles. When playing loud, I’d get a squealing feedback if I was too close to the amp. My guitar tech, John Carruthers, ended up stuffing the inside of the guitar “F” holes with a synthetic cotton type material and put a piece of slightly flexible plastic inside the “F” holes over the fake cotton to keep the cotton from coming out of the guitar. This fixed the problem. The tone of the guitar changed slightly, but not too much, and this internal “baffling” actually evened up the note balance of the guitar.

While on the subject of squealing guitar feedback (similar to microphone feedback), the Gibson 335, 345, and 355 “thin line” models have a block of wood that runs under the pickups through the length of the body. This helps note evenness and helps cut down on the feedback, compared to the Gibson 330 and most wide hollow-body guitars. If a guitar has no such block, the guitar is more susceptible to feedback and frequency build-ups

As with all styles of guitar playing, each jazz guitarist has a distinct sound. A player’s tone usually falls into one of three basic categories: dark, mid-rangy, or bright. Jazz guitarists are no exception to this rule. Probably 90 percent of the time, a jazz guitar player will use their guitar’s neck pickup (if more than one pickup is available on the guitar) to get a thick tone. Also, many jazz guitarists roll back the tone control on the guitar to get rid of high frequency information. If the guitarist is rolling off the highs on his guitar, adding EQ past 3 kHz is basically useless.

The EQ Settings that Work for Jazz Guitar

EQ-tutorial-fig4Otherwise, without further delay, here are the EQ settings that will help you record great jazz guitar:

Low frequency filter: Typically a very steep filter that eliminates low frequency information. If the guitar amp is being recorded in a room with other instruments (such as bass and drums) and there is low frequency leakage from other instruments into the guitar mic, it’s best to use the low frequency filter. Another use: If the guitar amp has a ground hum problem that can’t be gotten rid of, this filter should help. The filter may be frequency adjustable or a fixed frequency. If it’s adjustable, experiment with the frequency settings to find which works best in this application, but watch out that the roll-off does not hurt the guitar tone. Most likely the low freq filter will adjust sounds in the 100 to 150 cycle area. It’s best to use the 100 cycle or lower area if available. If the filter is a fixed frequency, the odds are good it is around 100 cycles.

30 to 80 cycles: Basically useless for this application. Note if you have no low frequency filter, you could use this area for roll off help.

From here on, I’m going to break up the information separately for archtop guitars and solid-body guitars.

80 to 200 cycles

80 to 200 cycles (archtop guitar): The odds are very good the jazz guitar tone is thick, so unless you think adding in the low end is needed, it’s best to pass on this frequency range. But, I mentioned that jazz guitars may have frequency build ups and suck outs so, to find a note build up area, ask the guitarist to play chromatically from the open low string on up, and look (listen) for frequency build-ups and or suck outs. If you find a loud low note, you will need a parametric or graphic equalizer to pull back that frequency. If you’re using a parametric equalizer set the “Q” to as small a range as possible. Use the EQ sweep technique to find the offending frequency and roll out to taste to make the guitar sound as “even” as possible. Find the suck outs in the same fashion you would look for a build up and then add in the frequency, but be sure that the addition is very tight, using a parametric or graphic equalizer.

80 to 200 cycles (solid body guitar): Again, the odds are very good the jazz guitar tone is thick so unless you think adding in the low end is needed, best to pass. The odds are good the solid body guitar will not have frequency build-ups or suck outs of a hollow-body guitar, but the amp may have some. For other playing styles so far, I’ve mentioned that the compressor will help smooth this stuff out. For straight-ahead jazz guitar, compression hurts dynamics, and dynamics are key to this style of playing – so I rarely use compression on a jazz guitar. With that in mind, you may want to use the concept in the paragraph above to smooth out frequency bumps and suck outs. Note that when I recorded my Bebop jazz CD, I recorded “direct” using a direct box (no amp or microphones) to avoid the problem. More on this in the “Direct Box” articles to come.

200 to 300 cycles

200 to 300 cycles (archtop guitar): Same as the 80 to 200 cycles: Look for build-ups and suck outs.

200 to 300 cycles (solid body guitar): The odds are good you will not need any help here since the tone will be thick. However, you may have some amp frequency bumps and suck outs. Check and deal with those as described above if need be.

300 to 600 cycles

300 to 600 cycles (archtop guitar): As you know, jazz guitar players use medium to heavy gauge strings, which produce a thick upper bottom/low midrange tone, with the help of the guitar tone control roll-off many jazz players prefer. At these lower frequencies, then, we’re primarily still on the frequency bump and suck out issue, trying to “even up” the notes. One thing to consider: If you are having trouble with some of the lower frequencies (particularly in the 80 to 200 cycle range) building up in an undesirable fashion and you need to roll out in a wide fashion, you might try to add here to make up for some of the “meat.” Start around 300 to hear what happens.

300 to 600 cycles (solid body guitar): Again, the odds are good this area is thick on its own, but take a look for amp build ups and suck outs. Again, if you are having trouble with some of the lower frequencies (particularly in the 80 to 200 cycle range) building up in an undesirable fashion and needed to roll out in a wide fashion, you might try to add here as to make up for some of the “meat.” Start around 300 to hear what happens.

600 to 800 cycles

600 to 800 cycles (archtop guitar): Mid-range city as usual and the guitar will most likely be thick here! We are now out of the woods with frequency build ups and suck outs that can be controlled without hurting the overall sound. The odds are huge no help is needed in this area. Hey, again, no rules – so if you need to add or pull back a few dB, do so. If the sound is lacking in this area, try adding. In most cases, this area will not need help.

600 to 800 cycles (solid body guitar): Same as the archtop settings. Probably no help needed here.

800 to 1 kHz

800 to 1 kHz (archtop guitar): If the tone is too dark and needs a slight mid-range bump for note definition, try adding a dB or so.

800 to 1 kHz (solid body guitar): Again, if the tone is too dark and needs a slight mid-range bump for note definition, try adding a dB or so.

1K to 2 kHz

1K to 2 kHz (archtop guitar): 1K is the center of the mid-range. The bandwidth of a telephone comes to mind. The odds are good nothing needs to happen here.

1K to 2 kHz (solid body guitar): Same as above. The odds are good nothing needs to happen here.

2 kHz to 3.5 kHz

2 kHz to 3.5 kHz (archtop guitar): If the sound is dark and you want to make it brighter, try adding here. A pro jazz guitar player that uses a dark tone may dislike the frequency addition as well as any other upper frequencies. You will find out how the guitar player feels about this on the first playback. Always remember to make the player happy.

2 kHz to 3.5 kHz (solid body guitar): If the sound is dark and you want it brighter, try adding here. For smooth jazz playing, the addition might be needed, but a pro jazz guitar player that uses a dark tone may dislike the frequency addition as well as any other upper frequencies. You will find out how the guitar player feels about this on the first playback. Always remember to make the player happy.

As I’ve mentioned in other EQ set-ups for other guitar styles, a dB or so goes a long way with this EQ area in the audio spectrum! If the guitarist is using a bright sound, of all the EQ areas, this can be your best friend or your worst enemy. When adding, listen loud to hear if you are adding too much. This area can get painful! No rules friends, just guidelines.

3.5 kHz to 5K

3.5 kHz to 5K (archtop guitar): This area starts bringing up the “sparkle.” In most cases, the jazz guitar sound will not like this area or any of the following upper-end EQ areas. OK, after saying that, since there are no rules, try adding as long as it does not thin out the sound. If the amp sound is not overly bright and the tone is not thin, this area may sound good to get a slight bit of “air” in the sound. Most jazz guitar players like a big sweet sound, but if you make it too bright, the guitarist will tell you for sure!

3.5 kHz to 5K (solid body guitar): Same as for archtops.

5 kHz to 8 kHz

5 kHz to 8 kHz (archtop guitar): More sparkle. If adding, watch out not to make to thin. If the guitar sound is dark, adding here will only bring up noise.

5 kHz to 8 kHz (solid body guitar): Same as for archtops.

8 kHz to 12 kHz

8 kHz to 12 kHz (archtop guitar): The pristine, sheen area. If a dark tone is preferred, this will not add anything but noise in most cases. If a bright tone is being used, this may make the sound thin. This may be a spot to roll out if the amp is noisy. If you’re recording to analog tape, you may want to do the roll out later when mixing to also take down the tape hiss at the same time.

8 kHz to 12 kHz (solid body guitar): Same as for archtops.

For smooth jazz playing, if the guitar player is going back and forth from playing rhythm to lead, you most likely will not have an EQ problem. If this is a problem, when in doubt, my school of thought is to always EQ on the bright side, especially if using analog tape. It is better to roll off a little top end when doing the final mix since you will also be taking down noise recorded (especially if analog tape) and recorder return/module path noise as well.

For straight-ahead jazz playing, if the guitar player is going back and forth from playing rhythm to lead, don’t worry about finding an average EQ. The key it so get the single line – the soloing – to sound as good as possible.

Many others excellent articles written by Jay Graydon, can be found at Guitar.com, Jay Graydon

Who says you can’t play jazz on a Strat?

In Gears, Sonorisation on May 27, 2016 at 3:24 pm

Who says you can’t play jazz on a Strat?! This partscaster is not “set up for jazz” in any special way, and yet it sounds great. No humbuckers, flatwounds, or severely rolled off tone here. See full specs at the end of the video.

How to get your Jazz tone on a Telecaster?

In Gears, Sonorisation on May 27, 2016 at 2:46 pm

Uploaded on May 4, 2011
http://www.soundpure.com/m/nash-guita…
In this demo we use the Henriksen Jazz Amp 112 er with a Nash T-57, T-63, and T-72 DLX. The Fender Telecaster is an icon in the guitar world. Its bright, gritty tone is well known throughout many genres such as rock, blues, country, and more. Not many people realize that it can sound fantastic in a jazz setting as well. The key to getting a jazz sound on a telecaster is cutting down the high’s a little and boosting the lows on the amp, while having the tone knob on the guitar halfway and the volume all the way up. This will give you a warm, creamy tone that works well with jazz. If you have any questions on about this video or the gear that was used, please give the experts at Sound Pure a call. 919.682.5552. or toll free at 888.528.9703. To check out the guitars used in this demo click the following links: http://www.soundpure.com/p/nash-t-57-… http://www.soundpure.com/p/nash-t-63-… http://www.soundpure.com/p/nash-t-72-…

The Frequency Spectrum of all Instruments

In Sonorisation, Theorie on May 27, 2016 at 9:49 am

frequency instruments rangeInstrument-Sound-EQ-Chart

l’effet Larsen

In electrics, Sonorisation on March 22, 2014 at 10:39 am

L’effet Larsen est un phénomène physique de rétroaction acoustique découvert par le physicien danois Søren Absalon Larsen. L’effet Larsen est souvent désigné sous le terme anglais de feedback.

Cet effet se produit lorsque l’émetteur amplifié (exemple : haut-parleur) et le récepteur (exemple : microphone) d’un système audio sont placés à proximité l’un de l’autre. Le son émis par l’émetteur est capté par le récepteur qui le retransmet amplifié à l’émetteur. Cette boucle produit un signal auto ondulatoire qui augmente progressivement en intensité jusqu’à atteindre les limites du matériel utilisé, avec le risque de l’endommager ou même de le détruire.

Ce phénomène est particulièrement fréquent dans tout système de sonorisation (conférence, concert, téléphone avec haut-parleur, prothèse auditive) et produit un sifflement de fréquence quelconque, pas forcément aiguë contrairement à une idée reçue (un larsen basse fréquence existe aussi et peut être très pénible) mais en général très désagréable (signal à la puissance maximale du système). La fréquence du son résultant dépend des fréquences de résonance des composants électriques et électroniques du système audio, de la distance séparant émetteur et récepteur, des propriétés acoustiques du lieu d’écoute et du caractère directionnel du récepteur.

Dans les années 1960, les guitaristes électriques du rock (en particulier Jimi Hendrix) cherchèrent à exploiter ce JimiHendrixphénomène auparavant considéré comme nuisible pour élargir la palette de leurs effets. Ils rapprochent alors volontairement les micros de leur guitare de l’ampli afin de lui arracher des sons stridents qu’ils tentent de moduler.

exemple: On retrouve un spectaculaire effet Larsen, aussi appelé sustain infini, dans la chanson Parisienne Walkways, de Gary MooreOn peut voir sur la figure le signal généré par l’effet Larsen suite à une impulsion continue donnée. Un signal sinusoïdal dont l’amplitude change de façon assez fluide mais qui s’amplifie est alors généré. Cette amplification est exponentielle. Le volume du son reste au maximum des contraintes techniques pendant une durée d’environ 1 seconde, puis s’atténue de façon linéaire.

Introduction to Audio

In Sonorisation, Theorie on March 11, 2014 at 8:50 pm

SoundThis beginner-level tutorial covers the basics of audio production. It is suitable for anyone wanting to learn more about working with sound, in either amateur or professional situations. The tutorial is five pages and takes about 20 minutes to complete.

What is “Audio”?

Audio means “of sound” or “of the reproduction of sound”. Specifically, it refers to the range of frequencies detectable by the human ear — approximately 20Hz to 20kHz. It’s not a bad idea to memorise those numbers — 20Hz is the lowest-pitched (bassiest) sound we can hear, 20kHz is the highest pitch we can hear.

Audio work involves the production, recording, manipulation and reproduction of sound waves. To understand audio you must have a grasp of two things:

  1. Sound Waves: What they are, how they are produced and how we hear them.
  2. Sound Equipment: What the different components are, what they do, how to choose the correct equipment and use it properly.

Audio Tutorials

IntroIntroduction to Audio; The basics of sound theory, sound equipment and audio work.
Audio ConnectorsConnections; Audio cables and connectors, wiring instructions, etc.
MicrophonesHow Microphones Work; Basic microphone technology, examples of common types, characteristics, etc.
Using Microphones, How to choose the correct microphone and use it properly.
Audio Mixing DeskSound Mixers; An introduction to sound mixers, from small portable units to studio consoles.
Balanced Audio TechnologyBalanced Audio, How balanced audio works and how to use it in your systems.
Sound Quality, Controlling sound levels and quality.
Audio NoiseNoise Types & Colours, White noise, pink noise, etc.

XLR to 1/4″ TRS Connector (wired for balanced mono)

In Sonorisation on March 11, 2014 at 8:42 pm

The usual way to connect a 3-pin XLR to a 1/4″ TRS (AKA stereo jack plug) is to use the following pin allocation:

  • XLR pin 1 to 1/4″ plug sleeve
  • XLR pin 2 to 1/4″ plug tip
  • XLR pin 3 to 1/4″ plug ring

XLR to 1.4-inch stereo jack plug

This wiring configuration gives you a balanced mono audio cable.

How to Wire an Unbalanced Microphone To A Balanced XLR Input

In Gears, Public Address, Theorie on March 11, 2014 at 8:23 pm
Sometimes it is necessary to wire unbalanced consumer microphones to a balanced professional input. This article describes how to connect a TS jack to XLR.

Using consumer microphones into professional audio equipment doesn’t often come up, but sometimes it has to happen. The problem is that pro audio gear usually has balanced inputs presented on XLR sockets, while consumer signal sources come presented as an unbalanced signal, usually a 1/4 in or 3.5mm TS jack for a microphone. Grounded consumer sources can cause hum by introducing ground loops into the system, but microphones usually float free of mains earth.First, if the consumer mic jack plug is not molded on, or will be cut off anyway, examine the microphone cable. If the cable has two cores and a shield, then it is a balanced cable wired to an unbalanced jack, and the cable can be wired to an XLR3 connector as a balanced connector as described in the article how to wire a XLR plug. The mic then becomes a regular balanced microphone. If the cable has a shield and just one core, or the plug is molded on and has to be kept, then proceed as follows.

Beware P48 Phantom Power on Professional Balanced Audio Inputs

The first thing to be aware of is P48 power. This is a 48V supply in series with about 6.8k ohms fed to XLR pins 2 and 3 relative to pin 1 (which is usually at ground potential). It is a common way to power professional microphones. Many mixers can be switched in sensitivity between microphone level and line level, but the phantom power supply may be present in both modes unless explicitly switched off.P48 power has the capacity to seriously harm the consumer microphone and must be switched off before a consumer source is connected to a professional input. On mixers there is usually a switch and associated LED to indicate phantom power on, but some items may have the phantom power enabling set in a menu somewhere. Before proceeding be absolutely positive that phantom powering is switched off.

Set The Mixer to Microphone Level Not Line Level

Consumer microphones output at low levels, anything from 25mV for 96dB SPL (loud!) for a sensitive electret down to a tenth of that for a dynamic microphone. The mixer should be set to microphone level sensitivity, and double checked for phantom power set to off. Some mixers automatically disable P48 powering on line level; it can be re-enabled on switching to mic level.

How To Wire a Balanced Consumer Microphone To a Professional Microphone Input

xlrplugxlr3

Wire the consumer microphone signal ground to XLR pins 1 and 3 (mixer ground and signal -ve) and wire the consumer signal core to XLR pin 2 (signal +ve).This can either be done as shown in the diagram, with a tip-sleeve line jack socket going via a short section of unbalanced audio single-core coaxial cable to a 3-pin XLR plug to go into the mixer.Alternatively the consumer mic jack plug can be cut off. If the microphone cable has a single center core and shield then it can be wired as follows:

  • XLR pin 1 and XLR pin 3 to microphone shield
  • XLR pin 2 to microphone center

it is a good idea to MARK the XLR connector as unbalanced, and a warning about the danger of phantom power!

Alternative Methods of Wiring Unbalanced Consumer Signals to a Balanced Professional Audio Input

An audio transformer can be used to interface the unbalanced microphone to the balanced input, however the cost of a good transformer can be prohibitive, and cheap audio transformers can impair the sound quality.Unbalanced equipment at line level can be interfaced to balanced inputs using a direct-inject (DI) box. If one is handy, it can be worth trying with a microphone but some types of DI box will be noisy with the low impedance weak microphone level signal.A different technique is needed if the requirement is to wire a balanced microphone to an unbalanced consumer microphone input.

from Richard Mudhar