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MPX over IP

In the progression from Circuit Switched Data to Packet Switched Data, I can think of many different applications for something like this:

FMC01 MPX to IP CODEC

FMC01 MPX to IP CODEC

The FMC01 MPX to IP encoder can be used for multi-point distribution (multi frequency or same frequency network) of FM Composite audio, or as a backup solution over a LAN bridge, LAN extension, or public network.  I can think of several advantages of using this for a backup when composite analog STL’s are in use.  There are many compelling reasons to extend the LAN to the transmitter site these days; Transmitter control and monitoring, security cameras, office phone system extensions, internet access, backup audio, etc.  I would think, any type of critical infrastructure (e.g. STL) over a wireless IP LAN extension should be over a licensed system.  In the United States, the 3.6 GHz WLAN (802.11y) requires coordination and licensing, however, the way the rules are set up, the license process is greatly simplified over FCC Part 74 or 101 applications.

Another similar CODEC is the Sigmacom Broadcast EtherMPX.

Sigmacom Broadcast EtherMPX CODEC

Sigmacom Broadcast EtherMPX CODEC

Features include:
• Transparent Analog or Digital MPX (MPX over AES), or two discrete L/R channels (analog or AES).
• Built-in MPX SFN support with PTP sync (up to 6.000km in basic version). No GPS receivers!
• Unicast or Multicast operation to feed unlimited number of FM transmitters with MPX from one encoder.
• Linear uncompressed PCM 24-bit audio.
• Very low audio latency: 2,5mS in MPX mode.
• Perfect match with Sigmacom DDS-30 Exciter with Digital MPX input.
• Can be used with high quality 802.11a/n Ethernet links.
• DC coupled, balanced Analog inputs & outputs with -130dBc noise floor.
• No modulation overshoots due compression or AC capacitor coupling.
• Decoder provides simultaneously Analog & Digital output for transmitter redundancy.
• Aux RS232 serial transparent link, Studio to Transmitter.
• Auto switchover to Analog input when Digital signal is lost.
• Centralized remote control & management software

One last thought; separating the CODEC from the radio seems to be a good idea. It allows for greater flexibility and redundancy. Using an MPX type STL allows sensitive air chain processing equipment to be installed at the studio instead of the transmitter site.

Analog Sauce

A little blast from the past. This was found in a transmitter manual at one of the sites we take care of:

CCA Optomod 8000

CCA Optomod 8000

I thought I would scan it and make it available here.  As luck would have it, there is also a corresponding piece of equipment to go along with it.  I had never seen a “CCA Optomod” (.pdf) before I was working at one of the radio stations in Trenton, Florida.  This unit was rescued from under a pile of garbage out in the lawn shed.  It was full of mud was nests and mouse droppings.  Needless to say, it required a bit of TLC to return it to operation.  I replaced the electrolytics, cleaned it up and ran some audio through it.  It is probably as good as the day it left the factory.  Bob Orban made some really good stuff in his day.

CCA Optomod 8000

CCA Optomod 8000

The original Optomod 8000 was an evolutionary design that made FM radio processing what it is today.  The idea of combining broadband limiter, AGC and stereo generator in one box was a radical departure from the norm.  The audio limiter functioned as a 15 KHz low pass filter and broadband AGC.

Orban Optomod 8000 audio limiter block diagram

Orban Optomod 8000 audio limiter block diagram

The stereo generator used very modest amounts of composite clipping to reduce overshoot and transients. Many people disparage composite clippers. If done correctly, it is transparent to the listener and increases perceived loudness by stripping off modulation product that is non-productive.

Orban Optomod 8000 Stereo Generator block diagram

Orban Optomod 8000 Stereo Generator block diagram

Some thirty five or so years later, there are still many of these units in service in various stations around the world.

Stereo Tool

FM and AM broadcast radio processing has gone through many iterations.  At first, the main processing function was to limit the input audio to a transmitter and prevent over modulation.  This was a particular problem with early tube type AM transmitters, where over modulation could create power supply overloads and kill the carrier while engineers scrambled around resetting things and hopefully pressing various buttons to get the transmitter back on the air.

Over the years, processors incorporated not just limiting, but compression, gating, equalization, clipping and so on all in an effort to keep ahead or at least abreast of the station across town.

Today, broadcast air chain processors come in all shapes and flavors.  In addition to that, internet streaming stations have their own unique set of issues to deal with.  The top of the line Telos Omina or Orban Optomod systems are great, however, they can set one back a pretty large sum of money.  Enter then, the Stereo Tool PC based software processing program.

Stereo Tool sofware screen shot

Stereo Tool sofware screen shot

The first difference between, say the Omina and Stereo Tool is the end user decides the hardware and basic operating system.  The second difference is Stereo Tool comes with a free trial.  Then there is the price difference, which ranges from about $48.00 US for the basic version, to $161.00 US for the basic FM version and finally $269.00 US for the full version (actual prices are in Euros, which will fluctuate day to day and the credit card company will likely charge an exchange fee).  Add to that a medium speed (2 Ghz) Intel Pentium4 or better computer, 1 Gb or more of RAM, good sound card and it all comes out to a reasonably priced audio processor.

Here are some of the specific features for broadcasting:

  • Declipper
  • Hiss Removal Filter
  • FM Hiss Removal Filter
  • Automatic Gain Control (AGC)
  • 10-band multiband compressor / limiter / clipper
  • Phasing error (AZIMUTH) correction filter
  • Stereo booster
  • Bass booster
  • Final limiter
  • Distortion masking Loudness filter
  • Lowpass filter
  • FM pre-emphasis filter
  • FM stereo encoder
  • FM RDS encoder
  • Composite limiter

Much more info at the Stereo Tool website.

The idea of PC based audio processing is new and interesting to most of us.  The designer and owner of Stereo Tool, Hans van Zutphen, was nice enough to answer a few questions I posed to him via email:

PT: What prompted you to write audio processing software?

HvZ: Since I was very little I’ve always wanted to have my own radio station. I remember playing with walkie-talkies and trying to receive their sound on a real radio when I was about 8 or 9. I never really did anything with it until I found out in 2001 that you could easily start a webradio station – I actually found out because I was listening to a pirate station in my car which turned out to have a stream; within a week my own station was online.

Of course I needed a bit of processing for it, and I wrote some command line tools – a singleband compressor, a stereo to mono convertor that didn’t cause any loss of audio (I was broadcasting hard trance on a mono 56 kbit/s stream, and this was the only way to get a decent sound out of it), and some time later a multiband compressor.

In 2004 I left the company I worked for (ASML, they make machines to make computer chips, customers are companies like Intel, AMD etc.) to start working for Philips Healthcare, where I was going to work on image processing for X-Ray systems. I had 2 months of ‘spare time’ between those jobs, and I wanted to learn to program in Visual C++, so I decided to a GUI around my command line tools, and make a Winamp plugin out of it. I called it ‘Radio Tool’. I never really planned to do anything with it, it was just an exercise project.

About a year later I came across the Winamp site again and I saw that you could upload plugins. So I uploaded my program, now renamed to ‘Stereo Tool’ because a Google search for “Radio Tool” gave far too many hits. Within a week there were over 1000 downloads and a while later it surpassed 90,000. At that point I decided to create a new version, Stereo Tool 2.0.

For quite a while this remained a hobby project, I occasionally worked on it for a few months and then I wouldn’t look at it for months. But at some point I was approached by someone people who worked at a “real” (FM) Dutch radio stations who asked for some extra features – he couldn’t get the audio loud enough, and that’s how I got into clipping. Things started to get better, I learned more and more about processing, the number of downloads increased and people became more and more enthusiastic about it. At some point, after reading something about how an FM stereo signal looks, I thought it might be possible to output a stereo signal with a 192 kHz sound card, so I bought one and did some tests and it worked that same night, and within a few weeks I added RDS.

PT: Do you know, approximately, how many stations (AM/FM/internet) Stereo Tool is being used on?

HvZ: FM: About 500, ranging from small community and pirate stations up to large nation-wide stations which run Stereo Tool at a dozen transmitter sites. Streaming: Not sure, but definitely over 1,000, probably a lot more.

PT: I have read through the forums on your site, Stereo Tool looks like a very complete processing system. Any plans for new features, future upgrades, etc?

HvZ: Yes. I’m currently working on a new multiband compressor. The multiband compressor in Stereo Tool is still based on the code that I wrote in 2001 for my webradio station, which in turn was based on an even older version that I had used on 8-bit audio. It also has far too many bands. Because of this, the multiband compressor is currently the weak spot of Stereo Tool. In the last weeks I have made a new singleband compressor that sound a lot better, it actually outperforms other compressors I have tested, and I expect great results for the new multiband compressor, which will also have less bands. Something else that I’ve been planning for a long time is a composite clipper, which will add 1-2 dB of extra loudness and especially better highs. Stereo Tool can already be louder with good audio quality than nearly any hardware box on the market (see for example this video, Radio 538 uses an Orban 8600 http://www.youtube.com/watch?v=4VpfcqUPQys – unfortunately due to the mpeg compression it’s a bit difficult to compare but listen for distortion ) – but there’s always room for improvement.

PT: What are the advantages of a PC software based processor vs. a hardware based (e.g. Omni or Optomod)?

HvZ: Ah, good question. Not sure if it’s the right question… With processing, a lot of things come down to taste, and there are several stations that have replaced their hardware processing by Stereo Tool not because it’s software and PC based but because they preferred the audio that comes out of it. Stereo Tool is also one of only 2 processors that contain a declipper (the other one is the Omnia 9, I licensed my declipper to them). For a demo of the declipper see: http://www.youtube.com/watch?v=oqOljvx9KaM

Also, Stereo Tool contains a stereo and RDS coder, most other processors don’t, so instead of having a whole bunch of devices everything can be done in a single PC, which also results in a better quality. Recently I added a new feature that enables synchronizing multiple FM transmitter signals that all connect to a simple Shoutcast stream (video: http://www.youtube.com/watch?v=GYQ5CYs0ZX8 ), so you also don’t need any streaming hardware anymore.

Of course there’s the price. A hardware box that gives “similar” quality (of course every processor sounds different, and it’s a matter of taste, so it’s difficult to compare, but I’m assuming that things like low volume levels, gain riding, distortion and lack of clarity in the highs are bad) easily costs $10,000 or more. And you can always easily upgrade to new versions. If you already have a PC with enough spare processing power you don’t need to buy anything.

I know that some people at radio stations are ‘afraid’ of using a PC in their processing path, but based on feedback I get from the stations that run my software it’s completely stable – and of course if a PC does break, you can replace it with any fast enough PC you have lying around – you just need to put the proper sound card in.

But for development, the advantages are huge. If you use DSP’s, it’s usually a lot of work to even make a very small change. When I worked at Philips Healthcare, the image processing that had been done – without much changes – on DSP’s for many years was being converted to PC’s because of speed of development and price of hardware. Once the conversion was finished, the development speed increased dramatically and 2 years later the image quality had improved beyond anything that was imaginable with DSP’s. PC’s get faster every year, and you don’t have to do anything for that – for the same price the processing power that you can buy roughly doubles every 1.5 years, and if you pay more you can get even more. If you use DSP’s, you have to do a lot of work yourself, you cannot just ‘buy a faster DSP’. Testing things is very easy, I can write some code that does something new, post it on my forum and I’ll have feedback from users the next morning – with DSP’s that’s a LOT more difficult and it takes a lot more time. I’ve learned by now that everyone hears things in a different way, and occasionally there are groups of people who hear something they find very annoying while many other people (often including myself) don’t hear anything wrong with it at all. Especially in cases like this it’s really great to be able to quickly send new versions to several people all around the world for testing.

PT: Are there any particular sound cards that work better with Stereo Tool?

HvZ: Yes. For the best results, use the Marian Trace Alpha, with the ESI Juli@ as second-best choice (it needs calibration).

Thank you very much, Hans, for the interesting insight.

Checkout the videos, especially the declipper video, which is quite amazing.  That will cleanup all but the most ham handed DJ mistakes.

PC based audio processing software is a great solution stations on a limited budget that cannot afford high end air chain processors.  There are many LPFM’s, Part 15 stations and others that can get great sounding audio and RDS for a very reasonable price.  Currently, the AM settings do not allow asymmetrical modulation, which is more of a US thing.  There is some talk of adding it in a later update.

Eye candy

Well, it certainly looks good:

Wheatstone Vorsis FM processor

Wheatstone Vorsis FM processor

I will have to set up the video equipment and record a song while filming the front panel display. By the way, it also sounds good.

Orban Optimod 8000

A piece of vintage gear from the late 1970’s, the Optimod 8000 was and still is a good sounding box.  I have often thought that these processors would make an excellent internet audio processor using the test jacks on the back of the unit.  The audio on these jacks is unbalanced and has 75 µS pre emphasis.   It would be easy enough to make a de-emphasis network and create balanced audio with a 10K:600 ohm transformer.  Some experimentation may be required with the transformer primary impedance value.  Orban notes that not less then 1 MΩ impedance should be connected to the test jacks.  For the internet station looking to copy the “FM radio” sound, this unit would do the job nicely.

The 75 µS  de-emphasis network would look something like this:

75 microsecond de-emphasis network, unbalanced to balaneced audio conversion

75 microsecond de-emphasis network, unbalanced to balanced audio conversion

In this case, the values for the de-emphasis network are fairly critical, therefore 1% or better tolerances for the resistors and capacitors is required.

Even better, an LPFM or some other radio station on a budget could acquire one of these for relatively little on eBay or somewhere else.  With a little TLC, most of these units can be rebuilt and put back into service.  I would recommend that some type of limiter be used in front of it, such as a Texar Audio Prism or CRL SEP-800.

Some classical music stations prefer these units.  I have noticed that they have a nice,  mellow, open sound.  Not at all fatiguing and yet still offer a nice easy 10 dB gain reduction.  There is also a modification that can slow down the release time on the gain reduction.  More gain reduction, AKA compression, can be had with something else in front of the unit.

The best part about these units, there is no rebooting, no processor lock ups, software glitches or any of that non-sense.  Additionally, a quick look at the front of the unit shows very few user controls, making it almost impossible to screw up and sound bad. They are well built and so long as the electrolytic capacitors are changed out, fairly bullet proof.  Other processors, not so much.

Optimod 8000A under test

Optimod 8000A under test

This is an Optimod 8000A that I decided to put through its paces.

Really, how much more do you need?  I recorded this on the camera microphone using a replica table radio, seen near the end of the video on the right hand side of the frame.

I used the Technics SL-1200 MKII turntable through an ATI P100 turntable preamp into the Optimod.  The Optimod is feeding a BE FX-30 exciter running 15 watts into a dummy load.   The Optimod is running about 5-7 dB gain reduction, which is enough in my mind.  The BE FX-30 is still just about the best sounding analog exciter every made.

Rechipped Optimod 8000A, TL071 opamp

Rechipped Optimod 8000A, TL071 opamp

This unit has been re-capped and re-chipped at one point.  The re-chipping follows the Orban recommendation; the 4558 and 1556 opamps are replaced by TL071CP and TL072CP respectively, and the uA 709 and 301A opamps are left in the unit.  A good thing to remember, the uA709 and 301A opamps can be replaced by TLO71cP opamps in the event of failure.  The Texas Instruments TL0 series opamps are very good and readily available.

Optmod 8000A input and limiter board

Optmod 8000A input and limiter board

Overall, this unit is in good condition, however, like many such units, it is missing its brown “Optimod” cover, which goes over the input/output controls.

Manual is available at the Orban ftp site: ftp.orban.com.

Occupy Bandwidth

Or rather occupied bandwidth.  During a recent Alternative Inspection of an FM station, there was some question as to the accuracy of the modulation monitor.  Truth be told, a modulation monitor is no longer required at a radio station, so long as the station ensures that they comply with relevant FCC regulations for their service.  Many modulation monitors continue on, however, as air monitor receivers.

That is all well and good, however,  many modulation monitors are notoriously inaccurate and tend to the overly sensitive side of the equation.  If used when setting the modulation levels, this can lead to under modulation, which, as we all know leads to disaster, destruction and bad ratings….  Because the volume knob on every radio in the entire metro, Total Survey Area (TSA), or even the whole country has been broken off and listeners are unable to compensate for the low audio levels from an under modulated FM transmitter.

But anyway.

FCC 73.1560 gives the maximum FM deviation as +/- 75 KHz from the carrier, with some allowance for SCA injection levels, up to +/- 78 KHz.  This is the definition of 100 percent modulation of an FM carrier.  Thus the entire occupied bandwidth is 150 KHz, leaving a guard band of 50 KHz between signals. That is, unless IBOC is employed, then the guard band is -100 KHz which is good science no matter how one looks at it.  On a spectrum analyzer, it looks something like this:

Occupied bandwidth of analog FM broadcast transmitter

Occupied bandwidth of analog FM broadcast transmitter

This shows that the 5 second average occupied bandwidth of 90 percent of the transmitted energy is within 153 KHz, which is slightly high but within the margin of error of the measurement device.  The vertical lines represent the -10 dB signal level as referenced to the carrier.  Thus this station is in compliance with FCC rules regarding modulation in spite of the face that the analog modulation monitor shows it being 10-20 percent over.   Had it actually been 110 percent, the occupied bandwidth would have been 165 KHz and 120 percent would have read 180 KHz.

Thus, before buying the latest squash-o-matic FM processor and setting it for full tilt boogy, a good engineer may want to check the occupied bandwidth with something other than the old analog FM modulation monitor in the rack.

FM Stereo vs station coverage

There are a few FM stations around here that intentionally broadcast in mono.  One is an FM talker, which from a technical standpoint makes a certain amount of sense since any particular human voice is a single point sound generator.

The other FM station broadcasting in mono, WKZE,  has a music format with a very eclectic play list.  It is a full Class A located in north western Connecticut.  The idea with this station is to garner a larger and more reliable coverage area.

It comes down to a simple physics discussion about free space loss.  The basic equation for free space power loss is:

where:

That formula works for a single frequency, say the carrier frequency, for example. As the signal gets spread out by modulation, the power density on any given frequency is reduced as the energy is divided between many other frequencies.

First, free space loss takes into account the spreading out of electromagnetic energy in free space is determined by the inverse square law, i.e.

\ S = P_t \frac{1}{4 \pi d^2}

where:

  • \ S is the power per unit area or power spatial density (in watts per metre-squared) at distance \ d,
  • \ P_t is the total power transmitted (in watts).

Second, with Frequency Modulation (FM), the power spectral density is a function of the differences in the highest and lowest frequency:

 P=\int_{F_1}^{F_2}\,S(f)\,d f + \int_{-F_2}^{-F_1}\,S(f)\,df.

Therefore, the narrower the bandwidth of a signal, the higher the density of the received signal will be in relation to the transmitted power.  An unmodulated FM signal will have a better, more reliable coverage area than a modulated one.  Of course, we need to modulate the signal, otherwise there is no point in having the transmitter on.

A baseband or composite FM signal has several components:

FM baseband signal

FM baseband signal

An FM station transmitting a mono signal will have a much lower bandwidth. With wideband FM, the modulation index is generally 2 or two times the maximum audio input frequency. Thus, a mono FM broadcast station will have an approximate deviation of approximately 30 kHz (plus any ancillary services like RDS) vs a stereo FM station, which has a 75-80 kHz deviation using the same carrier power.

For higher power FM stations, FCC Class C and B, this is not much of an issue.  Those stations tend to have a great deal headroom when it comes to power density, building penetration, multipath (picket fencing and capture effect).  For Class A and LPFM stations, it is a different situation.  For those stations,  unless FM stereo broadcasting is truly needed, it should be turned off.  On low power stations, stereo can be a great detriment to reliable coverage.

Vintage Texar Audio Prisms

I give to you, the original Texar Audio Prism:

Texar Audio Prism

Texar Audio Prism

I love the sound of these units when coupled with an Optimod 8100A.  Many people have (or rather, had) difficulty setting these things up.  I found them to be very easy to deal with, just follow the instruction manual.  If that doesn’t sound good, then there is something wrong with the unit.  Over the years, there are only a few consistent problems.  The first thing is with the voltage regulators.  They have heat sinks attached with nylon screws.  The screws get brittle and fall apart, making the regulator overheat and go bad.  I have taken to replacing the nylon screws, and if the heat sink has fallen off, the entire regulator.  There are also a few electrolytic capacitors in the power supply and on the audio board, it is always a good practice to replace those.  Otherwise, unless the unit has been blown up by lightning, it should work.

As for set up, follow the directions in the manual:

  1. Bypass the units using bypass switch
  2. Turn on on board pink noise generator
  3. Using the test ports on the front of the unit, plug a Simpson 260 VOM set on 2.5 VAC  important: use the ground port on the front of unit, not the case
  4. For use with an Optimod 8100A, using the dB scale on the Simpson 260, set all the bands for a 4.0 reading.  Set the density to 3/4.
  5. Turn off pink noise generator and switch out of bypass mode.
  6. Make sure the levels in the studio are where they should be.
  7. Adjust the input gain so the “Buffer Active” light does not come on during normal level programming.
  8. Adjust the output levels so that the input buffer on the Optimod reads between -7 and -3 vu.

The rest of the settings are on the Optimod:

  1. Clipping = 0
  2. HF limiting = 5
  3. Release time = 2
  4. Bass coupling = 2
  5. Gate = 0
  6. Set the input attenuators for about 10 dB total gain reduction, with peaks around 15 dB or so.

Then set the L-R null.  To do this, make sure the program material is in mono, then adjust the L or R input attenuator for minimum reading.  Also, if the Audio Prism has PR-1 (phase rotators) installed, bypass the phase rotator in the Optimod.  There is also a replacement card 5 made by Gentner called the RFC-1  for the Optimod 8100A.  I notice little difference between a stock Optimod and on RFC-1 Optimod.

That is a good starting point.  Most people are quite happy with this, but if needed, the high and low settings on the Prism can be adjusted slightly to suit the station equipment.  When properly adjusted, this equipment rides gain, adds a certain amount of loudness, while keeping the programming material natural sounding.  Further, unlike some “modern” air chain processors, it does not boot up and it does not occasionally loose its mind, requiring a reboot.

The best paragraph in the manual, or any broadcast equipment manual is this:

There is a wealth of information available in the LED display.  A few minutes of watching them in reduced light (emphasis added) while listening to a familiar program input will greatly help in understanding their action.

It will also greatly enhance your buzz, dude.  It was the 70’s.

Breakaway Broadcast

I am a strong proponent of non-computer based air chain processors.  Something about listening to dead air while the computer reboots is annoying and every computer needs to be rebooted every now and again.

All of that being said, I recently had a chance to play around with Breakaway Broadcast audio processing software.  I have to say, as a low cost, very versatile platform, it can not be beat.  I would put it up against any of the high end FM audio processing, provided one uses a high quality sound card with an adequate sample rate.

Claesson Edwards Audio has developed several software based audio processors for a variety of end uses.   They make several recommendations for hardware and operating systems, Pentium 4 3.2 GHz or better, dual core preferred.  If one is interested in used the sound card to generate composite audio, then any sound card capable of true 192 KHz sample rate will work.  They list several that have been successfully tested on their web site.

For approximately $1,200 dollars or so, one could buy a decent computer, the Breakaway Broadcast software and the Airomate RDS generator software.  For a Mom and Pop, LP or community radio station that is looking to do some high end audio processing and or RDS, that is a good deal.  I would add a UPS to the computer and keep back up copies of the software installed on an emergency computer just in case.  One can never be too safe when it comes to computers, viruses, hackers and other malicious persons.

Things that I like

  1. Inexpensive, the fully licensed version is $200.00.  The demo version is free but there is a 30 second promo every thirty minutes.
  2. There are several factory presets, but everything is fully configurable, changes can be named and saved allowing some experimentation.
  3. Audio cards with 192 KHz sample rate or greater can be used to generate composite audio, eliminating the need for a separate stereo generator
  4. RDS capable with additional software (Airomate2, approximate cost $35.00)
  5. The same processing computer can be used for streaming audio and or AM audio processing simultaneously.
  6. Full set of audio calibration tools for AM and FM transmitters, allows correction for tilt, overshoot and linerity.  Can add pre-emphasis at any user selectable rate.
  7. Fully adjustable phase rotators.

Things that I don’t generally like:

  1. Computer based system using Windoze operating system

WXPK in White Plains, NY has been using this software to process their streaming audio for about 2 years now.  The software itself is extremely stable running on a stand alone Windows box with XP service pack 2.

The loudness wars are over, Apple has won!

Excuse me while I gag…

Okay, that’s a little better.  I was just reading up on the newest, greatest, holy cow, gee whiz, gotta have that expensive box processor, also known as the Omnia 11.  I have to hand it to Mr. Frank Foti and his marketing team.  They have created one heck of a buzz about this thing, and it seems like folks are jumping on board to shell out $10 – $12 K for the box.  But let us review a few things.

I will admit most freely that I tend to be an audio purest.  I do believe that a limited amount of processing has its merits, especially for those listeners in high noise environments like automobiles, work sites, etc.   With sloppy DJ’s working the consoles, there is a minor need for some limiting, gain reduction and so on, just to the air product levels aren’t all over the place.  Those are the real world considerations.

Does and Ipod have an air chain processor? No, if the Ipod user want more loudness, they turn up the volume.  Since most Ipod users are normal people and not some burned out DJ with bad hearing, the volume control on an Ipod has plenty of head room to satisfiy.  Does a Droid or a Blackberry or whatever else people are listening to these days have an air chain processor? No.  And most users/listeners of those devices are perfectly happy with the quality and quantity of audio.

Back in the day when loudness meant a bigger transmitter, more carrier power, bigger signal, was easier to tune manually with the non-digital dial readout, etc., perhaps a loudness war with the cross town rival was part of the game.  Nowadays, nobody cares except the program directors.  I repeat, NOBODY CARES.  Ask anybody on the street what the loudest radio station is.  They very likely won’t even understand what you are trying to ask and you likely could not explain it in terms that would make them understand, much less care about.

The average person doesn’t give a rat’s ass about loudness.  Nor do they really care about how deep and full the DJ’s voice is, or how well the noise gate works, or the six band EQ or any of that crap.  In fact, if the music sounded just like it does on the Ipod, e.g. completely unprocessed, they probably wouldn’t even notice.  The competition has changed and radio is being left behind because many people are stuck with old ideas about how things used to be.  Times have changed, what should be the driving force in radio, the listeners, want to hear the music that they like.  That is what the program director should be worried about, finding and playing good music that the listeners want to hear.  Or having the best talk show, the most interesting news, or whatever other programming the station carries.

If the programming content is good, compelling radio, they will listen.  Never mind the air chain processor, the mic processor, the limiter, how loud the station is, what power the transmitter is running at, etc.  That is for the Engineers to take care of.

Axiom


A pessimist sees the glass as half empty. An optimist sees the glass as half full. The engineer sees the glass as twice the size it needs to be.

Congress shall make no law respecting an establishment of religion, or prohibiting the free exercise thereof; or abridging the freedom of speech, or of the press; or the right of the people peaceably to assemble, and to petition the Government for a redress of grievances.
~1st amendment to the United States Constitution

Any society that would give up a little liberty to gain a little security will deserve neither and lose both.
~Benjamin Franklin

The individual has always had to struggle to keep from being overwhelmed by the tribe. To be your own man is hard business. If you try it, you will be lonely often, and sometimes frightened. But no price is too high to pay for the privilege of owning yourself.
~Rudyard Kipling

Everyone has the right to freedom of opinion and expression; this right includes the freedom to hold opinions without interference and to seek, receive and impart information and ideas through any media and regardless of frontiers
~Universal Declaration Of Human Rights, Article 19

...radio was discovered, and not invented, and that these frequencies and principles were always in existence long before man was aware of them. Therefore, no one owns them. They are there as free as sunlight, which is a higher frequency form of the same energy.
~Alan Weiner

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