Old world and all that. I am of the impression that the European broadcast engineers are a more studied lot. There process involves much more deliberation, thoughtful analysis and planing than ours does. For example, when it comes to station loudness, most programmers and many engineers (myself not included) to the more is better. It is thus that we get the Omina 11 and other audio squashers.
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 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.
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 for an excellent sound card. The question is, is it worth it? The laws of diminishing returns states: 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 buss architecture, some are available with PCMCIA (laptop) or USB. For permanent installations, an internal PCI buss card is preferred.
Keeping an apples:apples comparison, this comparison it limited to PCI buss, stereo input/output, analog and digital balanced audio units for general use. Manufactures 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.
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 on line. The VX series cards are available in 2, 4, 8, or 12 input/output configurations. The HR suffix stands for “High Resolution,” which indicates 192 KHz sample rate. This card is capable of generating baseband composite audio, including RDS and subcarriers, with a program like Breakaway Broadcast.
2/2 balanced analog and digital AES/EBU I/Os
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 192 KHz sample rate on the analog input/outputs. Like Digigram, Lynx has several other sound cards with multiple input/outputs which are appropriate for broadcast applications.
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
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.
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.
There is a large number of things that amazes me on an almost daily basis. To wit: a local mom and pop radio station called me because they couldn’t get their computer program to work right. I decided that I’d give them an hour or two, in exchange for my hourly labor rate, and see if I could fix their problem. The issue at hand was loud hum and other noise on the input source. I knew before I even looked at it that the likely culprit was a ground loop.
It was worse than I imagined, with several unbalanced and balanced feeds improperly interconnected, line level audio going to a microphone level input and so forth. I explained to the guy about putting line level into a mic level input, something akin to plugging a 120 volt appliance into a 240 volt outlet. Improperly terminated balanced audio nullifies all of the common mode noise rejection characteristics of the circuit.
In any case, there are several ways to go from balanced to unbalanced without too much difficulty. The first way is to wire the shield and Lo together on the unbalanced connector. This works well with older, transformer input/output gear, so long as the unbalanced cables are kept relatively short.
Most modern professional audio equipment has active balanced input/output interfaces, in which case the above circuit will unbalance the audio and decrease the CMRR (Common Mode Rejection Ratio), increasing the chance of noise, buzz and so on getting into the audio. In this case the CMRR is about 30 dB at 60 Hz. Also, newer equipment with active balanced input/output, particularly some brands of sound cards will not like to have the Lo side grounded. In a few instances, this can actually damage the equipment.
Of course, one can go out and buy an Henry Match Box or something similar and be done with it. I have found, however, the active components in such devices can sometimes fail, creating hum, distortion, buzz or no audio at all. Well designed and manufactured passive components (transformers and resistors) will provide excellent performance with little chance of failure. There several methods of using transformers to go from balanced to unbalanced or vice versa.
Using a 600:600 ohm transformer is the most common. Unbalanced audio impedance of consumer grade electronics can vary anywhere from 270 to 470 ohms or more. The 10,000 ohm resistor provides constant loading regardless of what the unbalanced impedance. In this configuration, CMMR (Common-Mode Rejection Ratio) will be 55 dB at 60 Hz, but gradually decreases to about 30 dB for frequencies above 1 KHz.
A 600:10,000 ohm transformer will give better performance, as the CMMR will be 120 dB at 60 Hz and 80 dB at 3 KHz, remaining high across the entire audio bandwidth. The line balancing will be far better into the high impedance load. This circuit will have about 12dB attenuation, so plan accordingly.
For best results, use high quality transformers like Jensen, UTC, or even WE 111C (although they are huge) can be used. I have found several places where these transformers can be “scrounged,” DATS cards on the old 7300 series Scientific Atlanta satellite receivers, old modules from PRE consoles, etc. A simple audio “balun” can be constructed for little cost or effort and sound a whole lot better than doing it the wrong way.
A brief list, there are other types/manufactures that will work also:
A20, A21, A43
Keep all unbalanced cable runs as short as possible. In stereo circuits, phasing is critically important, so pay attention to how the transformer windings are connected.
All of the programming elements, all of the engineering equipment and practices, all of the creative process, the music, the talk, the commercials, everything that goes out over the air should reach as many ears as possible. That is the business of radio. The quality of the sound and the listening experience is often lost in the process.
Unfortunately, a large segment of the population has been conditioned to accept the relatively low quality of .mp3 and other digital files delivered via computers and smart phones. There is some hope however; when exposed to good sounding audio, most people respond favorably, or are in fact, amazed that music can sound that good.
There are few fundamentals as important as sounding good. Buying the latest Frank Foti creation and hitting preset #10 is all well and good, but what is it that you are really doing?
Time was when the FCC required a full audio proof every years. That meant dragging the audio test equipment out and running a full sweep of tones through the entire transmission system, usually late at night. It was a great pain, however, it was also a good exercise in basic physics. Understanding pre-emphasis and de-emphasis curves, how an STL system can add distortion and overshoot, how clean (distortion wise) the output of the console is, how clean the transmitter modulator is, how to correct for base frequency tilt and high frequency ringing, all of those are basic tenants of broadcast engineering. Mostly today, those things are taken for granted or ignored.
Every ear is different and responds to sound slightly differently. The frequencies and SPL’s given here are averages, some people have hearing that can go far above or below average, however, they are an anomaly.
An understanding audio is a good start. Audio is also known as sound pressure waves. A speaker system generates areas or waves of lower and high pressure in the atmosphere. The size of these waves depends on the frequency of vibration and the energy behind the vibrations. Like radio, audio travels in a wave outward from it’s source, decreasing in density as a function of area covered. It is a logarithmic decay.
The human ear is optimized for hearing in the mid range band around 3 KHz, slightly higher for women and lower for men. This is because the ear canal is a 1/4 wave length resonant at those frequencies. Mid range is most associated with the human voice and the perceived loudness of program material.
Bass frequencies contain a lot of energy due to the longer wave lengths. This energy is often transmitted into structural members without adding too much to the listening experience due to a sharp roll off starting around 100 Hz. Too much base energy in radio programming can sap loudness by reducing the midrange and high frequency energy from the modulated product.
High frequencies offer directivity, as in left right stereo separation. Too much high frequency sounds shrill and can adversely effect female listeners, as they are more sensitive to high end audio because of smaller ear canals and tympanic membranes.
Processing programming material is a highly subjective matter. I am a minimalist, I think that too much processing is self defeating. I have listened to a few radio stations that have given me a headache after 10 minutes or so. Overly processed audio sounds splashy, contrived and fake with unnatural sounds and separation. A good idea is to understand each station’s processing goals. A hip-hop or CHR stations obviously is looking for something different than a classical music station.
For the non-engineer, there are three main effects of processing; equalization, compression (AKA gain reduction), expansion. Then there are other things like phase rotation, pre-emphasis or de-emphasis, limiting, clipping and harmonics.
EQ is a matter of taste, although it can be used to overcome some non-uniformity in STL paths. Compression is a way to bring up quite passages and increase the sound density or loudness. Multi band compression is all the rage, it allows each of the four bands to react differently to program material, which can really make things sound differently then they were recorded. Miss adjusting a multi band compressor can make audio really sound bad. Compression is dictated not only by the amount of gain reduction, but also by the ratio, attack and release times. Limiting is a relative to compression, but acts only on the highest peaks. A certain amount of limiting is good as it acts to keep programming levels constant. Clipping is a last resort method for keeping errant peaks from effecting modulations levels. Expansion is often used on microphones and is a poor substitute for a well built quite studio. Expansion often adds swishing effects to microphones.
I may break down the effects of compression and EQ in a separate post. The effects of odd and even order audio harmonics could easily fill a book.