processing – Engineering Radio

The Voltair

I have put off writing anything about this for several reasons. First of all, there is a lot of secrecy surrounding the use of the Voltair magic machine. No one will admit to it, however, I have had several off-the-record conversations with various engineers. All of this is hush-hush, unofficially off the record and on the QT, so no names, call letters, or cities of license can be disclosed.

The general gist of these conversations is this; the Voltair seems to be increasing ratings in some cases but not others. It is sometimes too early to tell whether the increased ratings are a one-time anomaly or something more permanent. In one case, an AC station saw 30% increase in numbers, while a certain talk station saw next to nothing. Results are mixed.

In the credit where credit is due department; the Telos Marketing campaign has been effective. Again, from a variety of different sources; Program Directors, Market Managers, and Sales Managers are “beside themselves,” or “giddy” when the UPS truck delivers the Voltair to the front door. In one case, requiring that “I (the market engineer) drop everything” to get it installed as quickly as possible and “acting like it is God’s gift to radio.” It looks like all those trade publication ads are paying off, $15,000 at a time.

Voltair PPM encoder enhancing device, in the wild

One interesting thing about the Voltair, you can program simulated listening environments such as sporting events, restaurants, kitchens, vehicles, etc. This allows the user to see how their program material is being decoded by a PPM survey device in those types of environments. For example, if you are a sports station, having your program material decode well at sporting events or restaurants and bars might be important.

Of course, we have all seen the confidence display:

Voltair PPM encoder enhancer "confidence display" — Voltair PPM encoder enhancer “confidence display”

So, what does this mean? Perhaps there is an inherent flaw in the Nielsen PPM encoding technology. In the past, PPM has been blamed for the demise of the Smooth Jazz format. I always had the notion that Smooth Jazz was responsible for the demise of the Smooth Jazz format. However, if PPM is indeed causing certain program material to disappear from the airwaves, then it would be a case of the tail wagging the dog. If PPM requires that station owners purchase a $15,000 in order to get credit for their TSL and cume, then there is a pretty big problem with the technical aspects of the system.

Of course, there are others that say there is no “Voltair effect.” The Voltair machine is simply a fancy and expensive gizmo that looks good but does not really do anything.

Nielson Audio is having a Webinar on July 21 to address some of the questions regarding the Voltair and PPM encoding for subscribers only. It will be interesting to see what the outcome is.

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.

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.

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.

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.

They do it a little differently in Europe

Old world and all that. I am of the impression that European broadcast engineers are a more studied lot. Their process involves much more deliberation, thoughtful analysis, and planning than ours does. For example, when it comes to station loudness, most programmers and many engineers (myself not included) to do more is better. It is thus that we get the Omina 11 and other audio squashers.

The EBU technical group takes a different approach:

EBU R128 (ed: Loudness Recommendation) is the result of two years of intense work by the audio experts in the EBU PLOUD Group

Aside from the above-mentioned EBU R128, there are four technical papers dealing with implementation, meters, distribution, and so on. The body of work is a recommendation, not a requirement. I can’t imagine the voluntary implementation of something like this in the US. Even so, there are advantages to having a single acceptable level of programming audio. It is interesting reading.

Sound Cards for Broadcast Use

Computer audio sound cards are the norm at nearly all radio stations. I often wonder, am I using the best audio quality sound card? There are some trade-offs on the quality vs. cost curve. At the expensive end of the curve, one can spend a lot of money on an excellent sound card. The question is, is it worth it? The laws of diminishing returns state: No. High-quality reproduction audio can be obtained for a reasonable price. The one possible exception to that rule would be production studios, especially where music mix-downs occur.

I would establish the basic requirement for a professional sound card is balanced audio in and out, either analog, digital, or preferably, both. Almost all sound cards work on PCI bus architecture, some are available with PCMCIA (laptop) or USB. For permanent installations, an internal PCI bus card is preferred.

Keeping an apples: apples comparison, this comparison it limited to PCI bus, stereo input/output, and analog and digital balanced audio units for general use. Manufacturers of these cards often have other units with a higher number of input/output combinations if that is desired. There are several cards to choose from:

The first and preferred general all-around sound card that I use is the Digigram VX222HR series. This is a mid-price range PCI card, running about $525.00 per copy.

Digigram VX222HR professional sound card

These are the cards preferred by BE Audiovault, ENCO, and others. I have found them to be easy to install with copious documentation and driver downloads available online. The VX series cards are available in 2, 4, 8, or 12 input/output configurations. The HR suffix stands for “High Resolution,” which indicates a 192 KHz sample rate. This card is capable of generating baseband composite audio, including RDS and subcarriers, with a program like Breakaway Broadcast.

Quick Specs:

2/2 balanced analog and digital AES/EBU I/Os
A comprehensive set of drivers: driver for the Digigram SDK, as well as low-latency WDM DirectSound, ASIO, and Wave drivers
32-bit/66 MHz PCI Master mode, PCI and PCI-X compatible interface
24-bit/192 kHz converters
LTC input and inter-board Sync
Windows 2003 server, 2008 server, Seven, Eight, Vista, XP (32 and 64 bit), ALSA (Linux)
Hardware SRC on AES input and separate AES sync input (available on special request)

Next is the Lynx L22-PCI. This card comes with a rudimentary 16-channel mixer program. I have found them to be durable and slightly more flexible than the Digigram cards. They run about $670.00 each. Again, capable of a 192 KHz sample rate on the analog input/outputs. Like Digigram, Lynx has several other sound cards with multiple inputs/outputs which are appropriate for broadcast applications.

Specifications:

200kHz sample rate / 100kHz analog bandwidth (Supported with all drivers)
Two 24-bit balanced analog inputs and outputs
+4dBu or -10dBV line levels selectable per channel pair
24-bit AES3 or S/PDIF I/O with full status and subcode support
Sample rate conversion on digital input
Non-audio digital I/O support for Dolby Digital® and HDCD
32-channel / 32-bit digital mixer with 16 sub outputs
Multiple dither algorithms per channel
Word, 256 Word, 13.5MHz or 27MHz clock sync
The extremely low-jitter tunable sample clock generator
Dedicated clock frequency diagnostic hardware
Multiple-board audio data routing and sync
Two LStream™ ports support 8 additional I/O channels each
Compatible with LStream modules for ADAT and AES/EBU standards
Zero-wait state, 16-channel, scatter-gather DMA engine
Windows 2000/XP/XPx64/Seven/Eight/Vista/Vistax64: MME, ASIO 2.0, WDM, DirectSound, Direct Kernel Streaming and GSIF
Macintosh OSX: CoreAudio (10.4)
Linux, FreeBSD: OSS
RoHS Compliant
Optional LStream Expansion Module LS-ADAT: provides sixteen-channel 24-bit ADAT optical I/O (Internal)
Optional LStream Expansion Module LS-AES: provides eight-channel 24-bit/96kHz AES/EBU or S/PDIF digital I/O (Internal)

Audio Science makes several different sound cards, which are used in BSI and others in automation systems. These cards run about $675 each.

Audio Science ASI 5020 professional sound card

Specifications:

6 stereo streams of playback into 2 stereo outputs
4 stereo streams of record from 2 stereo inputs
PCM format with sample rates to 192kHz
Balanced stereo analog I/O with levels to +24dBu
24bit ADC and DAC with 110dB DNR and 0.0015% THD+N
SoundGuard™ transient voltage suppression on all I/O
Short length PCI format (6.6 inches/168mm)
Up to 4 cards in one system
Windows 2000, XP and Linux software drivers available.

There are several other cards and card manufactures which do not use balanced audio. These cards can be used with caution, but it is not recommended in high RF environments like transmitter sites or studios located at transmitter sites. Appropriate measures for converting audio from balanced to unbalanced must be observed.

Further, there are many ethersound systems coming into the product pipeline which convert audio directly to TCP/IP for routing over an ethernet 802.x based network. These systems are coming down in price and are being looked at more favorably by broadcast groups. This is the future of broadcast audio.