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- July 5, 2010 at 1:22 pm #7481
Hi,
I’m just starting and I need your help setting up my audio equipments. I have the followings:
1) Behringer 1002B
2) Behringer Composer-Pro MDX2600
3) Behringer Ultragraph-Pro FBQ1502Hi,
I’m just starting and I need your help setting up my audio equipments. I have the followings:
1) Behringer 1002B
2) Behringer Composer-Pro MDX2600
3) Behringer Ultragraph-Pro FBQ1502
4) Laptop where all my music is stored
5) Mike ? don’t which one to buy (I would to start with good cheap one for the purpose of learning)I play mostly jazz and classical music. Please educate me, how do I hook up the equipments to get the most soothing stereo sound (good base). What is your recommended settings for the EQ and the MDX2600? If these equipments are not suitable what are your recommendations?
It is worth it to have the Behringer DSP2024P VIRTUALIZER PRO?
Thanks a bunch,
–JudeJuly 5, 2010 at 2:19 pm #19114Carl Blare
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Total posts : 45366Hello Jude:
Interested in offering a few ideas.
Will you be AM, FM or both?
What audio format are your files recorded in, i.e., .wav, .mp3?
What media player are you using, i.e., Windows Media Player, Winamp?
July 5, 2010 at 3:19 pm #19115Carl Blare
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Total posts : 45366Hi Jude:
I just did some quick research to give you a few more answers.
Microphone: Shure PG48 cardioid pattern voice mic $39.97 balanced XLR
A fine starter mic that might be all you ever need.Because classical and jazz recordings are already carefully balanced by their producers, I suggest that transferring that quality without much change is the goal.
The mixer board has EQ adjustments for each input channel, but I recommend only using the microphone EQ settings and leaving the music inputs “flat.” At the start, keep the microphone EQ flat unless you want more “voice presence,” in which case boost 3.5kHz by 3 to 5dB.
The limiter should be set so that the very loudest sounds cannot over-modulate the transmitter.
The compressor will be a complicated problem, because recordings vary so much. The ones I hate are the classical recordings where the loud sections can be heard in the next county but the soft passages are way off in the breeze. I’ve never found a good answer for that. Also, voice compression and music compression require different settings, so I’m addressing only music settings at this time. Perhaps start with a very small compression of 1:1.5
AM transmitters benefit from an EQ boost in the upper mid-range, to make the audio sound “bright,” but this is not a good idea for FM, which already sounds good. The SStran AMT-3000 has a jumper to add +8dB boost that rolls up starting at 2kHz. If your transmitter does not have this feature, it can be done with your Behringer EQ.
You’ve got some high-end equipment!
July 5, 2010 at 3:22 pm #19116jude
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Total posts : 453661) Will you be AM, FM or both?
FM2) Will you be AM, FM or both?
For now FM only.3) What audio format are your files recorded in, i.e., .wav, .mp3?
Mostly MP3 (128 and 190kbps) and OGG (160 kbps), and some FLAC4) What media player are you using, i.e., Windows Media Player, Winamp?
foobar2000 (winxp) and Rhythmbox on linux (ubuntu 10.04).Between the mixer and Laptop I’m using Berhinger UCA222 for playback only. In other word the UCA222 serves as an external usb soundcard and link my laptop with the mixer.
Thanks,
–JudeJuly 5, 2010 at 3:57 pm #19117jude
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Total posts : 45366Blare,
Thanks much for the advice. Right I’m not too worry about the setting the microphone. I would to ficus on the music been played over the air. I’m not too sure the way I interconnect the mdx2600 and the eq are correct. I have the EQ side chain to the MDX2600. Having this setup I do hear the EQ’s punch. So I’m sure if the equipments hook up is correct. For now I would to focus on that.
Thanks,
–JudeJuly 5, 2010 at 5:06 pm #19118Carl Blare
Guest
Total posts : 45366OK Jude:
I think there might be a standard for setting up what radio stations call “the audio chain.”
The microphone and music source each get their own mixer input (1002) and the output of the 1002 would go to the EQ would go to the compressor would go to the transmitter. The EQ and compressor probably have a bypass switch so you can easilly take them out of the circuit.
I don’t think the return paths need to be incorporated except in certain types of recording studios.
Your speaker monitor would be well served by an A-B-C selector switch, so you could listen to the (A) mixer output before it’s processed; (B) the sound after the EQ & compression; (C) a radio tuner. Then you could instantly compare every stage of the operation.
Another engineering task is to avoid loudspeaker feedback while you are speaking on the microphone. The cheap way is to manually turn down the speakers while on the mic.
July 6, 2010 at 2:06 am #19120Ken Norris
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Total posts : 45366… can become overtly complex and easily abused. Here’s an article I’d trust from MIX magazine:
http://mixonline.com/mag/audio_understanding_compressors_compression/FWIW, although I have them, I generally don’t use hardware processors much anymore. Being a Mac guy, I have a plethora of AUs I can use, and my broadcast software offers a way to control placement in the chain. But the theory is basically the same. In this case, I get pretty good results from a Talking House TX with MultiBand compression and a declipper. I can recover quite a bit of volume for modulation while avoiding distortion.
I use settings which work well when the audio comes from an internet stream through a little ol’ G3 iBook (iTunes) and feeds the transmitter. Basically all of the processing is done before it gets sent to the internet The iBook picks up the stream and sends audio to the TX
There are some standard parameters for processing audio for AM radio, but I believe each situation will have to be tweaked to fit. IOW, there is general advice, but from there you’ll need to experiment to get the best results. I actually have settings that vary with different genres of music … problems with playback of classical music already mentioned is a good example.
Nice to know there are so many experienced people in this forum … 😉 HTH …
July 6, 2010 at 6:38 am #19121Ken Norris
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Total posts : 45366“Another engineering task is to avoid loudspeaker feedback while you are speaking on the microphone. The cheap way is to manually turn down the speakers while on the mic.”
Most people already own some sort of headset, as in iPod or whatever, or any other headset, any of which (doesn’t have to be total isolators in most cases) will be better than taking chances with feedback in speakers. Not my particular style (I own a studio, have great mics and headsets), and, mind you, I haven’t tried the particular item below (no idea about quality), but there’s a reason why you see DJ’s wearing something like this:
http://www.amazon.com/Headphone-microphone-Combo-Djs-Broadcasters/dp/B0002D0HT4… what I’m saying: If you already have a headset, then use it instead of loudspeakers when on-air. I found a new Shure SM58 (significantly better than a Shure PG-anything), which is the most popular stage mic in the world, certainly the toughest, on eBay for a good price:
http://cgi.ebay.com/Shure-SM58-58-Pro-Dynamic-Microphone-NIB-/170509226535?cmd=ViewItem&pt=LH_DefaultDomain_0&hash=item27b3245227#ht_1969wt_1139… heck at that price I think I’ll get a couple myself. These come without cables, though … no problem for me, I own a sound studio, there’s probably a couple dozen 30-footers around here., but BSW has good deals on them:
http://www.bswusa.com/proditem.asp?item=SMM10… not Wave or Monster Cable quality, but they should be fine for short runs.
HTH …
July 6, 2010 at 5:14 pm #19124kk7cw
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Total posts : 45366Even though it’s not the least expensive solution, If you are going to do a lot of live programming on-air consider the Broadcast Tools CC-IIA console controller. It has 3 inputs for side chain mic inputs, a “muting” circuit for use when you switch on a mic in the studio and a monitor level control. It allows the use of speakers and up to 3 microphones. The unit also can run an “on-air” light and can switch the monitor between the mixer output and a received signal/external audio. The hook up is very straight forward.
I have used mine for several years. I wouldn’t operate without it.
They are available at several broadcast and electronics resellers.
July 6, 2010 at 9:41 pm #19128Carl Blare
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Total posts : 45366The speaker muting method described by Marshall Johnson, Sr. is undoubtedly the most conventional method and it’s the method I always employed in previous studios (now I do not have live broadcasts, only pre-recorded).
But I came across a marvelous setup observed during a visit to a 50kw talk station where the loudspeakers never turned off and yet there was no mic feedback. The loudspeakers stayed on for the sake of live guests who did not wear headsets, so they could be aware of spots, music, network feeds and so on with the speakers always on. How did they do it?
For fun I’ll let someone else tell the trick, or if it’s a super secret that no one knows, I’ll return to let the cat out of the bag.
July 6, 2010 at 10:51 pm #19129Ken Norris
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Total posts : 45366Dunno what ‘trick’ you speak of, but facing monitors away from mics, as the mains in most theatre and church clusters are, and using directional mics with their rear side pointed at monitors (directional mics don’t pick up sound directly behind them) will work. Pointing speakers away in a direct line reverses the direction of the sound as well, i.e., reversed phasing cancels much sound getting back into the mic.
Another method is to reverse the polarity of the monitors in relation to the mic … also causes phase cancellation. Tricky … depends on distance and sound frequencies, room acoustics, etc.
Theatre speakers can be cranked up really loud. But I use nearfield speakers which are never more than 3′ away from a condenser mic. The headset is the ticket for me, as it clarifies sound detail, because the mic gets swung around. Of course, it needs to be matched to the externally heard signal as well, so you need to know the relationship.
Still another method is to figure out which frequencies are causing feedback, and simply pull them out with a graphic equalizer.
… or any combination of the above that works, which is how we do it in both our local community theatre and in the church I go to. We ring out the room and play with controls accordingly. Most rooms have a dominant frequency where it tends to ring, plus harmonics.
July 6, 2010 at 11:59 pm #19130Carl Blare
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Total posts : 45366Close, Ken Norris, but not “the trick” I have yet to describe. I still want to milk the fun of giving someone else the opportunity to make that presentation.
But what you bring up is a huge and fascinating field (sound re-enforcement). And with that, I have always wondered about those mysterious devices known as “feedback eliminators,” which I’ve never had a chance to test. All I know is they are circuits which detect the frequency at which a feedback is building up and does a “pitch shift” to keep the feedback from happening. Maybe that would work in a radio studio.
July 11, 2010 at 7:16 pm #19156mighty1650
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Total posts : 45366About feedback…
My Monitor is right behind me so it actually faces the mic. But I have never experienced feedback unless I have it really cranked loud.July 16, 2010 at 4:17 pm #19204Carl Blare
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Total posts : 45366When I encountered a series about equalization written by Dave Moulton in TV Technology Magazine I was reminded of this thread, because in his 3-part series Mr. Moulton tells us everything we will ever need to know. This is text book complete.
July 17, 2010 at 7:16 am #19209Ken Norris
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Total posts : 45366So … what about your ‘trick’? When do we get to learn about that?
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