November 2018
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Summing to Mono

It is surprising to me how many times I have seen this done incorrectly in the field. Summing a stereo source, whether it is balanced or unbalanced is not simply twisting a couple of wires together.  This will effectively reduce the impedance of the outputs by one half. With newer, active balanced outputs, this may cause damage to the output amplifiers.

The parallel resistance formula is thus:

Therefore a 600 ohm stereo output tied together would look like:

Rt = 1/(1/600+1/600) or 300 ohms.

It also creates an impedance mismatch with the next piece of gear, which will effect the common mode noise rejection of the circuit.

The best way to sum is through a resistive network.  That way stereo separation is maintained, the impedance of the output circuits is maintained and the output amplifier will not current cycle.  That looks like this:

resistive summing network

resistive summing network

Pretty easy to fabricate in the field.  It is good to do things the right way, it sounds better on the air too.

The old humming console

We are starting to work at a new client’s studios.  It is a bit like stepping into a 1980’s 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.

A what?


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, the 12 volt supply, not so much:

12 volts DC, 2.7 volts AC

12 volts DC, 2.7 volts AC


12 Volt power supply

12 Volt power supply

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.

12 volt linear power supply

12 volt linear power supply

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.

The Ground Loop

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:

Ground Loop schematic

Ground Loop schematic

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).

Radio Station Common Point Grounding

Radio Station Common Point Grounding

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.

Differential Signaling

Differential Signaling

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.

They do it a little differently in Europe

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.

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 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 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.

Digigram VX222HR professional sound card

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 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.

Quick Specs:

  • 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.

Lynx L22-PCI professional sound card

Lynx L22-PCI professional sound card


  • 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
  • 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

Audio Science ASI 5020 professional sound card


  • 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.

Unbalanced to Balanced 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.

simple balanced to unbalanced audio connection

simple balanced to unbalanced audio connection

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.

Balanced to unbalanced audio using 1:1 transformer

Balanced to unbalanced audio using 1:1 transformer

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.

Balanced to unbalanced audio using a 4:1 transformer

Balanced to unbalanced audio using a 4:1 transformer

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:

Ratio Jensen Hammond UTC
1:1 (600:600) JT11E series 804, 560G A20, A21, A43
4:1 (10K:600) JT10K series 560N A35

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.

Everything we do is destined for one place.

I give you, The Human Ear:

Anatomy of the human ear

Anatomy of the Human Ear, courtesy of Wikipedia

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.

Audio frequency vs. wavelength chart

Audio frequency vs. wavelength chart

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.

Base 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 clasical 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.

Audio over IP, what is it, why should I care?

IP networks are the largest standardized data transfer networks worldwide.  These networks can be found in almost every business and home and are used for file transfer, storage, printing, etc.  The Internet Protocol over Ethernet (802.x) networks is widely understood and supported.  It is robust, inexpensive, well documented, readily deployed and nearly universal.  Many equipment manufactures such as Comrex, Telos, and Wheatstone have developed audio equipment that uses IP networks to transfer and route audio within and between facilities.

IP protocol stack

IP protocol stack

Audio enters the system via an analog to digital converter (A/D converter), often a sound card, at which point a computer program stores it as a file.  These files can be .wav, .mp3, .mp4, apt-X, or some other format.  Once the audio is converted to a digital data format, it is handled much the same way as any other digital data.

IP stands for “Internet Protocol,” which is a communications protocol for transmitting data between computers connected on area networks.  In conjuction with a transmission protocol, either TCP (Transmission Control Protocol) or UDP (User Datagram Protocol) IP forms what is known as the Internet Protocol Suit known as TCP/IP.  The Internet Protocol Suit contains four layers:

  1. Application layer – This is the protocol contains the end use data.  Examples of these would be HTTP, FTP, DCHP, SMTP, POP3, etc.  Telos Systems uses their own application called “Livewire” for their equipment.  Wheatstone uses “WHEATNET.”  Digigram uses “Ethersound.”   This is an important distinction.
  2. Transfer layer – This contains the TCP or UDP header information that contains such things as transmitting, receiving ports, checksum value for error checking, etc.  It is responsible for establishing a pathway through multiple IP networks, flow control, congestion routing, error checking and retransmission.  TCP allows for multiple IP packets to be strung together for transmission, increasing transfer rate and efficiency.
  3. Internet layer – This is responsible for transporting data packets across networks using unique addresses (IP addresses).
  4. Link Layer – Can also be called the physical layer, uses Ethernet (802.x), DSL, ISDN and other methods.  Physical layer also means things like network cards, sound cards, wiring, switches, and routers.


An IP network can be established to transmit data over almost any path length and across multiple link layer protocols.  Audio, converted to data can thus be transmitted around the world, reassembled and listened to with no degradation.  Broadband internet connections using cable, DSL, ISDN, or T-1 circuits can be pressed into service as STL’s, ICR’s, and TSL’s.  This translates to fast deployment; no STL coordination or licensing issues, no antennas to install if on a wired network.  Cost reductions are also realized when considering this technology over dedicated point-to-point TELCO T-1’s.  Additionally, license free spread spectrum radios that have either DS-1 or 10baseT Ethernet ports can be used, provided an interference free path is available.

IP audio within facilities can also be employed with some brands of consoles and soundcards, thus greatly reducing audio wiring and distribution systems and corresponding expenses.  As network speeds increase, file transfer speeds and capacity also increases.


Dissimilar protocols in application layer means a facility can’t plug a Barix box into a Telos Xtream IP and make it work.  There are likely hundreds of application layer protocols, most of which do not speak to each other.  At some point in the future, an IP audio standard, like the digital audio AES/EBU may appear, which will allow equipment cross connections.

Additionally, the quality of the physical layer can degrade performance over congested networks.  The installations must be carefully completed to realize the full bandwidth capacities of cables, patch panels, patch cords, etc.  Even something as little as stepping on a Category 6 cable during installation can degrade its high-end performance curve.  Cable should be adequately supported, not kinked, and not stretched (excessive pulling force) during installation.

TCP/IP reliability is another disadvantage over formats like ATM.  In a TCP/IP network, no central monitoring or performance check system is available.  TCP/IP is what could be called a “broadcast” protocol.  That is to say, it is sent out with a best effort delivery and no delivery confirmation.  Therefore, it is referred to as a connection-less protocol and in network architecture parlance, an unreliable network.  Lack of reliability allows any of these faults to occur; data corruption, lost data packets, duplicate arrival, out of order data packets.  That is not to say that is does not work, merely that there is no alarm generated if an IP network begins to loose data.  Of course the loss of data will effect the reconstruction of the audio.

Analog digital converter symbol

Analog digital converter symbol

Finally, latency can become an issue over longer paths.  Every A/D converter, Network Interface Card (NIC), cable, patch panel, router, etc has some latency in its circuitry.  These delays are additive and dependent on the length of the path and the number of devices in it.

Provided care is taken during design and installation, AOIP networks can work flawlessly.  Stocking adequate spare parts, things like ethernet switches, NICs, patch cables and a means to test wiring and network components is a requirement for AOIP facilities.

This is what you’ll get…

Back many, many years ago, in a city far away, I was driving down the road and I flipped one of “my” stations on the air.  The end of this song was playing:

The ending sounds an awful lot like a Moseley MRC-16 transmitter remote control’s return telemetry.  When I heard that on the air, my first response was “HOLY SH*T! The telemetry is on the main channel!”  A little voice in the back of my head said “That is not possible.  How is that possible?”  I grabbed the gigantic, then state of the art Motorola bag phone and dialed the studio hot line, just before I hit the  “send” button, the song faded out and the announcer came on back selling “Karma Police by Radiohead

Wow.  Radiohead?  Karma Police?  WTF?

I almost had a coronary.  When I got home, I tried explaining this all to my then girl friend, who didn’t get it.  Few do.  At the time, making such an error would be very bad form indeed and likely open the unfortunate party to all sorts of snickering and finger pointing at the next SBE meeting.

Crown D75 monitor amp goes terminal

Happened the other day, took out the monitor speakers too.  I am not sure how this happened, but the production director reported that the speakers began making very loud squeal.  Somebody finally thought to turn off the amp using the conveniently located on/off switch on the front panel.

Crown D75 audio board burned open resistor

Crown D75 audio board burned open resistor

The two watt resistor is burned open.  Also, this got so hot it burned a hole in the circuit board below it.  Truth be told, I think this amp was about 25 years old and due to be replaced when the new studios were built out.

I’ve seen these Crown amplifiers self destruct in the past.


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 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

Free counters!