We are starting to work at a new client’s studios. It is a bit like stepping into a 1980s time machine, as the newest console seems to be the Broadcast Audio console in the FM studio. I feel I should wear a wide colorful tie and part my hair in the middle when working there. There is also an older UMC console in the second production room.
It seems the UMC console (UMC was a Connecticut-based console manufacturer that was later sold to Broadcast Audio) was having an intermittent hum problem on all the audio buses.
After poking around under the hood for a few minutes, I decided I should begin with the basics. Checking the power supply for ripple seemed like as good a place to start as any. This console has a 30-volt and a 12-volt power supply. The 30-volt supply checked out good, but the 12-volt supply, not so much:
2.7 volts AC on the 12-volt DC power supply. That will put some hum on the audio, all right. I tried to replace the power supply main filter capacitor, but it had no effect. The regulator must also be bad and it is a Motorola part number which is likely not made anymore.
This is a pretty standard off the shelf power supply, I should be able to get one from Mouser for about $60.00 or so for a linear unit, which will be cheaper than us trying to trouble shoot and repair the old one. In the meantime, I took the 10 amp 0-30 volt bench supply and pressed it into temporary service. The console is working again, for now.
At some point, all this old, um, stuff needs to be replaced.
Audio Engineers will know this subject well. Grounding has many purposes, including electrical safety, lightning protection, RF shielding, and audio noise mitigation. Although all types of grounds are related in that they are designed to conduct stray electrons to a safe place to be dissipated, the designs of each type are somewhat different. What might be an excellent audio ground may not be the best lightning ground and vice versa. Sometimes good audio grounds can lead to stray RF pickup.
The basic ground loop looks something like this:
Where RG should equal zero, in this representation it is some other resistance. This causes a different potential on the circuit (V1), which in turn causes current to flow (I1). It is that unexpected flow of current that creates the problems, causing voltage (V2) to be induced on another part of the circuit. In cabling applications, this will result in a loud, usually 60-cycle hum impressed on the audio or video being transmitted through the cable.
The resistance can come from something as mundane as the length of the conductor going to ground. This can often happen when using shielded audio wire in installations when the connected equipment is already grounded through the electrical plug.
There are two proven methods for eliminating ground loops, both of which are best implemented in the design phase of construction (aren’t most things).
The first is a single ground point topology, also known as a common point or star grounding system. A common ground system consists of one grounding point or buss bonded together so that it has the same potential. All grounded equipment is then connected to that point creating a single path to ground. All modern electrical equipment has a path to ground via the third prong of its electrical cord. Problems can or will occur when audio equipment is plugged into separate AC circuits, grounded via the electrical plug, and then tied together via an audio ground. The longer the separate grounding paths, the more severe 60 cycle (or some harmonic thereof) hum can result.
To eliminate this problem, the shields should be broken at one end of the audio cable. Never cut the third prong off of an electric cord, which can create another problem called electrocution. Given the choice between a ground loop and electrocution, I’d stay away from electrocution, mine or somebody else’s.
For installations in high RF fields, the open shield or ground drain can act like an antenna. In those situations, the open end can be bypassed to ground using a 0.01 uf ceramic disk capacitor. Electrically, this will look like an open at DC or 60 cycles, but allow stray RF a path to ground. This problem can be a common occurrence when studios are co-located with transmitters.
The second is by using balanced audio or differential signals as much as possible. This poses a problem for those stations that use consumer grade components, especially in high RF fields. For shorter cable lengths, two or three feet, it is usually not a problem. Anything beyond that, however, and trouble awaits.
It is relatively easy and inexpensive to convert audio from unbalanced to balanced. As much as possible, equipment and sound cards that have balanced audio inputs and outputs should be used. In the end, it will simply sound better to use higher quality equipment. Also, longer cable runs need to be properly terminated at both ends.
Installing equipment using good engineering principles and techniques will eliminate these problems before they start.
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.
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.
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.
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.
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.
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
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.