Krone is a standard system used for connecting audio wires together in a (semi-) permanent way. In the stoic complex it is primarily found on the back of the audio rack in the cool room and behind the audio patch under the sound desk. All of the audio patch bays terminate in krone blocks, which then have other wires to connect them on to another part of the studio complex. There are a number of advantages of krone which explain its widespread use in the audio set-up. Firstly, the connections are reliable once made, second there is no soldering required, thirdly, there is not a mass of connectors behind every piece of equipment and finally it offers another way of routing signals around faulty equipment. Connections are made by knife blades contained within the plastic housing of the blocks (which are mounted on krone frames bolted onto the rack) which cut through the insulation on the audio wire and make a secure contact with the conductor. This contact is achieved when the wires are "punched" onto the krone blocks with a special tool. The tool also cuts off any excess length of wire so that unwanted random connections are not made by loose wire-ends. Each krone block can connect six balanced audio lines using the 18 available slots. The blocks are marked "gp1 gp2 gp3 gp4 gp5 gp6" across the top row and "abs abs abs abs abs abs" across the bottom row. The convention used for krone is that "a" is the hot core of the balanced audio wire, "b" is the cold core of the audio cable and "s" is the shield. With almost all of the FST audio cable used in the stoic studio complex this corresponds to "a" = white, "b" = blue, "s" = green. Unlike patch there is no standard for whether signals in the top row are "in" or "out". The top row of any given krone block is connected to the patch bay and the wires are routed through the middle over the top of the block. Once the krone blocks are fitted to the frames the other connecting wires are routed through the plastic gates at the side of the block and below the bottom row before being brought up to be "puched" onto the krone. When working with krone in complicated audio setups such as the stoic complex IT IS VERY IMPORTANT TO BE NEAT. Groups 1,2 and 3 should be routed in from the left hand side of the krone block through the plastic gate while groups 4,5 and 6 should be routed in through the right hand side plastic gate. The wires should then be routed as neatly as possible through the metal gates on the krone frames to keep them out of the way of other krone blocks. Excess wire length should be routed somewhere out of the way and cable tied neatly away. This is important because it makes fault tracing and finding any given wire much easier. If these rules are not followed it is very easy to create a "rats nest" which is very difficult to work with in the future.
Krone blocks attached to patch bays are arranged on the krone frames in the following logical manner to make identifying which wire corresponds to which socket as easy as possible. Each patch bay has four krone blocks attached to the top row and four attached to the bottom row. The first block attached to the top row of the patch bay (i.e. sockets 1-6) is located on the first available slot on the krone frame. The first block attached to the bottom row of the patch (i.e. sockets 25-30, or 27-32 for 26-Way patch) is then located directly below the first block attached to the top row of the patch. This means that the krone connections for patch sockets which are normalised together are located directly above/below each other, making it easier to follow a signal. While there is only space for three blocks on any row of a krone frame, not the four which would be desirable, this is simply worked around by continuing on the next available slot and counting through the krone to find where the patch bay you are working with starts.
Krone connections can be undone using the fold out "unpicker" on the krone tool but this should only be done if it is absolutely necessary. Krone allows semi-permanent re-routes to be performed behind the patch bays by using a length of FST (properly routed, of course) to link two krone blocks directly, thus bypassing any faulty equipment in between. The FST is connected to the bottom half of each krone block, bypassing any faulty patch sockets. This should only be done if absolutely necessary because the bypass is impossible to see when looking at the front of the patch and still very difficult to see amongst the krone blocks themselves.
When searching for faults in the audio system the krone can be tested directly using a "krone-blade" connected to a set of headphones. The krone-blade should be inserted between the top and bottom row of a krone block so that the left terminal touches the "a" connector of a given group and the right terminal touches the "b" connector of the same group. If the krone-blade is not correctly positioned it will not be possible to push it far enough into the krone to make a connection, don't force it! This will then allow you to hear in the headphones the audio signal being passed by the krone and identify if it is coming from the desired source.
Audio Levels
DAs
Analogue Video
Usage of SPG and TBC - Yes, I know that TV abounds with TLAs (Three Letter Acronyms) but they're so much easier to say. An SPG is a Sync Pulse Generator; ours is a 1U cream-coloured box near the top of the vision rack in the cool room. This generates the black pulses described above and is best imagined as the station's heartbeat. All of the signals in the central vision system are syncronised to this.
The quality of audio recordings made depend largely on the understanding of a few basic principles of audio signals and equipment. These relate to audio levels, which are the voltage of the audio signal travelling from one piece of equipment to another.
Line level is the standard audio level from the outputs of equipment such as VT decks and the matrix. This is the standard range of voltages for audio signals to which all sound going into a destination, such as a VT deck, should be mixed. When operating a sound desk the main mix output should peak (i.e. have a maximum level at any time) at the 0dB level on the VU meters which show the mix output level. As with any other amplification system for electrical signals 0dB means that the signal is neither amplified, nor attenuated. If the audio signal for the mix on a sound desk (or any single channel) goes far above the 0dB level, then the sound will distort and crackle which should be avoided. If the signal is much below the 0dB level the sound on the recording will be too quiet and will need amplification later (i.e. on the television set).
Microphones always produce signals at a much lower level (usually 20dB less) and require additional amplification in the sound desk. This is why line level signals should not be routed through microphone channels on the sound desk (the eight mono channels on the K1) otherwise the desk will add 20dB of gain to the signal which will cause it to distort. Similarly, microphone level signals should not be routed through line level channels on a sound desk (the blue faders on the K1) as the desk may run out of gain before the signal is suitably loud.
A DA is a Distribution Amplifier (it has nothing specific to do with "Digital" or "Analogue"). All of our DA's are for analogue video or audio signals, and occasionally UHF. The purpose of a distibution amplifier is to provide many copies of the same signal without any quality loss or attenuation. DA's therefore almost always provide 0dB of gain but produce many copies of the signal using an op-amp, the circuit inside a DA is in fact very simple. In stoic video DA's are widely used to provide monitors with feeds of the same signals as those going into the vision mixer and to provide copies of the black signal generated by the Signal Pulse Generator (SPG) and other important sources such as the sustain service. For audio stoic only requires a Mix audio and sustain audio DA because all of the monitoring is done using the sound desk. The video DA's are a custom designed system by the Technical Manager (then called Chief Engineer) of 1993, Bryan Crotaz. In his system the DA's are mounted on interchangeable cards.
Each card can provide 6 copies of a signal and the connections on the back of the case are as follows: The top row is the input to each of the cards and the output of any given channel/card is the sockets following a zigzag pattern down from the input socket. Each card has a green LED on the front that under normal conditions should be fully illuminated. If one of the two fuses on the card blows the LED will be dimmer than usual and the card no longer provides output. If both fuses blow the LED will go out completely. The cards can easily be removed and fuses replaced to bring the channel back into operation or, if there is an urgent need for the channel, the cards can be swapped with spares for instant repair. The audio DA's are 1U units which are complete units with connectors on the back that are well marked for which channel they are, although they are usually phono connectors. The audio DA's do not have fuses to blow and generally do not fail.
Tech: ****
Genlock Theory
Note: some familiarity with analogue electronics is recommended if you want to make head or tail of this.
There are many different ways of sending video down electrical cables in analogue format. All of them rely on breaking the frame up into scan lines and the lines into pixels which are sent as analogue voltages in a stream down the cable.
This is simple enough in black-and-white; there is a defined scan speed, and the voltage at different times on the wire gives how white the given pixel is.
RGB video is an extension of this, in that there are three seperate lines, each monochrome (effectively black and white) giving the strength of red, green and blue on the pixel. The monitor then combines these three on its screen as different coloured phosphors. (In vision mixing there is often a fourth, known as 'alpha' or 'key', which gives the intended transparency of the pixel, but we don't need to consider this at present)
We use composite video, which is a system whereby all of the colours are sent down one wire. I don't know the exact details of it but this isn't important. Roughly speaking, the scanline starts with a 'colour burst' which relates different voltage levels in the scanline to different colours in the picture. Then the scanline itself appears.
So the signal would look something like this if you viewed it on an oscilloscope:
Each 'staircase' gives a single scan line of colour bars in composite video.
Notice the large negative swing at the start of the trace. This tells the monitor that it is the start of a scan line. Immediately after this is the colour burst. The burst varies depending on what colours are present in the picture. After that the tops of each 'stair' give each colour block in the picture. It is hard to make sense of the colour burst from the 'scope trace so a second piece of equipment called a vectorscope is used. This locks onto the burst itself and displays it in the form of a series of interconnected vectors:
Full colour burst from a series of colour bars
This is important later when we consider phase adjustment (in the Video section)
However, back to genlock theory.
When you think about it, there is no reason for two signals from separate sources (e.g. two cameras) to line up with each other. This presents a problem for mixing the two together. If you select a raw, non-genlocked source on a matrix monitor and switch to another (e.g. the Hi8 deck is not genlocked) you will see the monitor picture 'roll' as it tries to lock onto the new position of the video signal.
If you fed them through an analogue vision mixer such as the KM-2000 and tried to fade the result is even worse - the picture just breaks up, as the two signals become a complete jumble.
Genlock, put simply, is ensuring that these signals are lined up when it matters. This is done by feeding out a black video signal:
which as you can see consists of nothing but the sync pulse at the start (and, technically, a black burst and black frame). Equipment with a genlock or analogue sync input takes this signal in and outputs its scanlines in sync with the pulses fed in. This is why it is possible switch between two correctly set-up cameras in the studio on a matrix monitor without a roll, and indeed why you can use them on the vision mixer.
You may notice that above I mentioned the VT decks. Most of these are not genlockable (with the exception of the UMAT and SVHS decks). So how do we use them in the vision mixer? Answer: with a Time Base Corrector (TBC). This is a handy little gizmo which takes an unsyncronised signal in along with a black signal, and syncronises the video to the black. Put simply, it does the genlock for you. There are four of these, again mounted in the cool room. This system has implications for setting up the vision system through the matrix, as detailed in the video section.
OB Considerations - Again you may have noticed that throughout I have only referred to the KM-2000 vision mixer (the main studio one) as opposed to the MX-50 (the smaller 4-channel one, used for OB). This is because you do NOT need to genlock sources going into the MX-50. Why? Because the MX-50 has its own TBCs built in. However, this means that if you cut on the bus (i.e. simply pressing button 1 then button 2 on either bus) like you do on the KM-2000 the picture momentarily freezes as the TBC waits for the next signal to come through. Cutting across the bus (i.e. selecting channel1 on bus 2 and then cutting to bus 2) is fine.
Although the MX-50 has a black burst output it does not seem to be possible to genlock equipment onto it. This means that you must always be aware of the bus problem. However you don't need to worry about wiring genlock and adjusting phase in an OB set-up.
The other thing to remember about this is that if you're connecting an OB directly to the studio (as in Fresher's Fair 2002) you must run the output through a TBC.
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