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Technical Question

B

brobin1595

Guest
Hey Guys,

I've got a techincal question for you. In order to take a conversation straight from the phone line and plug it into an audio board to broadcast over the air, what would i need, adapter wise?

Thanks,
Brandon
 
> I've got a techincal question for you. In order to take a
> conversation straight from the phone line and plug it into
> an audio board to broadcast over the air, what would i need,
> adapter wise?

A hybrid. It is a device that takes the phone line and splits it into a send path (which you feed from the board), and a receive path (which you feed into the board). It does the neato task of not feeding what you send into the phone line back to you.

As for adapters, it depends on the situation and the exact equipment involved.
 
Re: What is "mix/minus"?

> What is "mix/minus"?

Consider doing a live talk show.

Until you have a caller you are producing a "mix" from various
audio sources. Your live microphone. Music off CD's (or some
other source). Spots/PSA's off a hard drive or maybe carts
if you're living in the past. It's everything a listener hears
on the radio; it's "the mix". Now, add in a caller. The
caller becomes part of "the mix" as heard by the listener. BUT
if you feed that entire "mix" back into the phone for the caller
to hear you get screeching feedback. So what you need to send
back to the caller is "Mix Minus". Which is to say, the mix
your audience is hearing MINUS the caller.

Many boards offer special modules and/or selective mix-minus
capability and then you just set it up properly and feed it to
the phone.

Odd this should come up just now. I'm working with two boards
that are used for talk or recording off the phone. In the
one case there is a single phone module which is fed caller audio
from the hybrid and which feeds mix-minus (minus the incoming
on that module) back to the hybrid to be heard by the caller.
The producer wanted the ability to have a second caller present
at the same time as the first one. That could mean buying another
$600 module which would also eliminate not just one, but TWO
inputs to the board, these module being single input/single purpose.
Since this particular phone line is intended for calls being made
TO someone...like a politician who's to be a "guest", a little
inconvenience was acceptable. The same solution applied to another
room where recording off a phone will be a rarity:

Connect the phone hybrid output (caller) to a standard line input
on the console. Take the L and R outputs of the AUDITION channel
of the board to a summing amplifier. I'm using some stuff a company
called "Henry" made and which were cast off from a previous purpose.
Feed the output of the summing amplifier back into the hybrid for
audio to go back to the caller. The trick is to assign to PROGRAM
AND AUDITION all of the modules EXCEPT the one being used for the
phone. That will create, on the audition channel, what is effectively
"mix-minus". Of coure this assumes you're using the program channel
on the air or to feed your recording device. If you wanna be lazy,
you could connect just L or R to the hybrid whose input is mono anyway.
The risk is that the guest caller might miss something important.

Everything will be fine until somebody forgets and assigns that module
with the phone to audition as well as program...but they'll find out
real fast!<P ID="signature">______________
God save us from those who would save us from ourselves! P-l-e-a-s-e!!!!!</P>
 
There are inexpensive couplers that will allow you just to pick up the receive audio. These are fine where you don't need to carry on a conversation with the incoming phone caller. Or want to receive audio from a distant location, such as a feed off a PA system.

For that matter, they can be used to send audio down a phone line. Yep, you guessed it, two are handy to send audio from remote location to your studio, one at each end. Essentially they are nothing more than a transformer, capacitor, and diode (to control the maximum level sent into the line). Usually they have a modular to plug into a telephone jack and a 1/4 inch phone jack or terminals to feed audio in or out.

Check out www.bswusa.com; as well as some of the other broadcast equipment sellers. Last time I looked they ranged in price from around $50 to $200.
 
Re: What is "mix/minus"?

> Everything will be fine until somebody forgets and assigns
> that module
> with the phone to audition as well as program...but they'll
> find out
> real fast!

Before I built my mix-minus box (one op-amp and a few resistors is all you need), like you I used one of the program buses. Don't know if it has anything to do with the station being student ran, but they kept assigning the phone module to that bus as well, despite all my explanations (and obvious feedback on the air). I ended up disabling the bus switch. Got to make it fool-proof...


Regards,
Goran Tomas
 
Re: What is "mix/minus"?

> Before I built my mix-minus box (one op-amp and a few
> resistors is all you need), like you I used one of the
> program buses. Don't know if it has anything to do with the
> station being student ran, but they kept assigning the phone
> module to that bus as well, despite all my explanations (and
> obvious feedback on the air). I ended up disabling the bus
> switch. Got to make it fool-proof...

I have reached that wonderful age where I don't cater to
fools. I tell 'em how it works and, if they elect to screw
it up and make themselves deaf with feedback...well they deserve
to be a little hard of hearing.

I think of it as offering them a wondr'ous learning experience.
Some are volunteer interns who someday are going out into
the cruel world of commercial broadcasting and a little harsh
experience now will save them from serious trouble later.

Besides, I'm just a mean old fart.

I'm what Sam wants to be when he grows (I almost said "up")
older.<P ID="signature">______________
God save us from those who would save us from ourselves! P-l-e-a-s-e!!!!!</P>
 
Re: (Long Message Here But Hopefully Informative)

> > What is "mix/minus"?
>
> > BUT if you feed that entire "mix" back into the phone for the
> > caller to hear you get screeching feedback.

There is another reason to use Mix-Minus that you might want to discuss. It is also used for remote broadcasts. This includes remotes done via RPU transmitter, telephone devices such as Comrex Hotline, and Telos ISDN tranceivers. It would also apply to methods such as sattelite, and anyone daring enough to send audio over the internet, but we do not use either of these last two methods.

The problem is one of audio delay - just like you get in a recording studio when someone puts an echo effect on a guitar for instance. The farther away the talent is from the studio, the more delay you get. Yes - the audio signal travels at close to the speed of light through the wires and analog equipment that it passes through, but believe it or not, the human ear can still perceive this as an echo, at what might seem microscopic delay times - on the order of microseconds. On the surface, this may seem ridiculous, but how do you think the human ear and brain can tell what direction a sound is coming from? They determine direction by the tiny delay time between the arrival times of a sound to both ears - a sound coming from your left will arrive to your left ear sooner than it arrives to your right ear.

With analog devices, such as a fully analog studio console, an RPU transmitter, and FM Subcarrier for IFB, the delay is minimal. HOWEVER, with the explosive growth of digital devices in our audio world, the delay times have become much greater - in some cases as long as MANY SECONDS. Digital devices such as Hotlines, ISDN, sattelite, and don't forget CELL PHONES - all have computer chips and analog-to-digital converters that convert the audio of our analog world into the ones & zeros of the digital world. The problem is that this conversion takes time. It can take anywhere from a few hundreths of a second, to many seconds depending on the devices and transmission methods. Our Telos ISDN boxes have about a one-second roundtrip delay when using Layer 3 mode. Our internet audio streams are done on a PC with Windows Media Encoder, and it takes about 8 seconds to be encoded, sent to the internet, then decoded by a listening PC. Yes - EIGHT seconds..!

Ok fine, "why's he saying all this?" you're thinking. This is why: the problem arises when the talent listens to the air signal in his/her headphones. If the talent listens to the air signal, they will hear themselves in delay. In other words, they don't hear themselves immediately - they hear an echo of themselves. And the length of the echo can be anywhere from milliseconds to many seconds depending on the equipment. And this gets CONFUSING. Your talent will start to hesitate, stutter, and sound like a dork on the air because they can't hear themselves correctly. In some cases, they might even stop and drop a nice "is this thing on?" on the air.

If there is a mixer at the remote end, and the remote producer/engineer creates a "local mix" of the program audio & air signal, the effect will vary. What happens is that when the talent is not "potted up" and on the air, they sound normal to themselves in their headphones. BUT the problem starts as soon as they go live on the air, and start hearing themselves from the local mix, as well as through the air feed. Why? Keep reading.

If the equipment at both the remote and studio is all analog (and thus a very short delay time), the talent will hear themselves in their headphones ok, but the audio will sound like it's kind of out of phase, or like they are talking into a milk jug or a bucket. As the distance between the studio and the remote site increases, so does the delay time. Thus the "bucket" gets bigger and the audio more weird sounding. Most of the time, talent can work with this, but you still generally want to avoid it if you can.

If there is ANY digital equipment in the audio path, the delay becomes MUCH GREATER. When I do remotes with ISDN in Layer 3 mode, the delay time is almost ONE SECOND. I do have a mixer at the remote site to create a local mix so they can hear themselves in their headphones, but sometimes the producer back at the station forgets to keep our fader OUT of Audition on the board (yes we do it that way - big cluster, many stations, lots of shared inputs - can't use dedicated MixMinus modules)... This usually results in a fast and heated phone call from myself to the producer back at the station to yell at them about it (again) (Arrghh!) :). Most of the time the talent takes it in stride (with difficulty), but every once in a while it really throws them off.

Another thing to consider when this kind of delay is involved is the queueing of talent with music - or anything really, including commecrials. If the talent likes to hit those song intros and outros like a hammer, it't NOT going to work with a one second delay! They're gonna be stepping all over the songs and their little egos will get hurt, and then they'll probably get all ticked off and start yelling at someone - usually the remote engineer (aka - ME). The talent AND the producer back at the station MUST be aware of the delay and that they have to "run the board loose", not to mention keeping the fader out of audition on the console.

In regards to the simple act of putting a phone caller on the air for a radio talk show, if there was not the problem of screeching feedback, there would still be the problem of the delay. I hear it myself every time I set up an RPU remote and call the studio for a sound check. If I have my cell phone up to my ear, and talk into the mic at the remote, and the producer has the studio speakers turned way up, I can hear myself in the studio speakers back through my cell phone - but with noticeable delay. Now because the RPU is analog, it's signal gets back to the station faster than the audio from my cell phone, which is delayed because of its digital audio converters. This delay isn't excessively long, but if the two signals were put together in a headphone mix, it would be pretty distracting. SO - the point here is that feedback or no, we still need mix-minus for phone callers.

Hope this helps.

Matthew Shea, Radio Engineer
Entercom Communications
WBEN, WTSS, WKSE, WWKB, WGR, WWWS, & WLKK
500 Corporate Parkway, Suite 200
Buffalo, New York, USA
(716)843-0262
 
Re: (Long Message Here But Hopefully Informative)

> Our internet audio streams are done on a PC with Windows
> Media Encoder, and it takes about 8 seconds to be encoded,
> sent to the internet, then decoded by a listening PC. Yes -
> EIGHT seconds..!

That's pretty small for internet broadcasting. The delay isn't caused by digital conversion...it's caused by buffers on the encoder, on the audio distribution server and on the client. Buffers are used to prevent hiccups in case of a transmission error on the line.<P ID="signature">______________
</P>
 
And then...

> The problem is one of audio delay - just like you get in a
> recording studio when someone puts an echo effect on a
> guitar for instance. The farther away the talent is from the
> studio, the more delay you get. Yes - the audio signal
> travels at close to the speed of light through the wires and
> analog equipment that it passes through, but believe it or
> not, the human ear can still perceive this as an echo, at
> what might seem microscopic delay times - on the order of
> microseconds. On the surface, this may seem ridiculous, but
> how do you think the human ear and brain can tell what
> direction a sound is coming from? They determine direction
> by the tiny delay time between the arrival times of a sound
> to both ears - a sound coming from your left will arrive to
> your left ear sooner than it arrives to your right ear.


Given all the complications introduced by digital, credence is
given to an excellent video/audio design engineer with whom I
worked on television products years ago:

"If digital had been invented first, we'd be going through the
analog revolution today." It has nothing to do with quality;
it's all about obsoleting technology and forcing people to buy
new stuff."

For my part, I long since decided that all progress is evil!<P ID="signature">______________
God save us from those who would save us from ourselves! P-l-e-a-s-e!!!!!</P>
 
Re: And then...

> "If digital had been invented first, we'd be going through
> the
> analog revolution today." It has nothing to do with
> quality;
> it's all about obsoleting technology and forcing people to
> buy
> new stuff."

I completly agree. The "digital revolution" has in fact greatly degraded audio and video quality across the board. Psychoacoustic audio compression (aka - MP3, etc.) has ruined the audio, and DVD has ruined video. I am constantly hearing slushy-sounding audio on TV and radio. And anyone can see MPEG compression artifacts on TV or DVD - all you have to do is look for the colored checkerboard patterns in large fields of similar color; for instance when the camera is pointed at the sky, or an underwater shot like whales with nothing behind them but clear blue water.
 
Re: (Long Message Here But Hopefully Informative)

> That's pretty small for internet broadcasting.

Ahh... Yes. That's was just within our LAN when testing the connection to the server. I don't know what kind of delay our listeners actually get.
 
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