Production Tips & Techniques
Most sequencer users know that you can insert a Program Change command into the MIDI data to change sounds mid-song, and similarly, that Program Changes can be positioned at the start of a song to set up the required sounds before the music starts. As some synths take longer than others to change patches, it is advisable to leave a short gap between a Program Change command and the next note following it, but this can often be overlooked. Sometimes you'll get away with it, but if you work using a lot of sequenced loops and you have a Program Change command in the data stream right at the start of the loop, you might find that your sequence glitches every time the loop comes around. This can be very frustrating until you figure out what's going on, and the problem can be compounded if you work by copying tracks or cutting and pasting data, as it's easy to inadvertently copy an unwanted Program Change along with the note information.
The answer is to be more disciplined in your approach and put all the initial setup patch changes in the count-in bar of the piece, preferably as a separate pattern. If Program Changes are needed later in the piece, make sure they're not hidden away inside loops; instead, put all the Program Changes for each pattern in a separate track which isn't looped. Not only will this prevent glitching, it will also avoid the problems that sometimes occur when you program a track mute and end up muting a track where some vital Program Change should have occurred. Paul White
One of the great things about MIDI recording is the way in which it lets you continue changing and enhancing the sound of a part long after the performance has taken place. But haven't you ever wanted to do the same thing with guitar parts? Haven't you ever given a great performance, only to find that the sound really doesn't do it justice, no matter what you try in the mix? Well, in fact, guitar players too can have this facility. All you have to do is split the output from your guitar at the recording stage. Using a DI box (preferably an active one with a very high input impedance to avoid any loading effects on the instrument), send one leg to your normal amplifier or recording processor and send the other leg straight to the tape machine (via the mixer), with no EQ and no effects — above all, you need to keep it clean (to keep the dynamics intact), so if anything, play safe with the level. What you will capture with this track is the unadulterated output of the guitar; effectively, just the 'performance'. If you then have to try a 'rescue job' because the sound was not right, but the playing was great, you can route the DI track through an amplifier, or processor, just as if it were the original signal, playing around with the sound until it is just right.
The signal coming back from the system is going to average around a third of a volt (or even three-quarters, if it is a pro machine), so you need to attenuate this significantly until it replicates the handful of millivolts generated by a guitar, in order to get the distortion sensitivity about right — routing the return channel to a group output, or using a prefade aux, is usually the most convenient way of accessing the signal without hearing it (you only want to hear the version that has been through the amp/processor). There will also be rather more noise, as your amp/processor's sensitive front-end is seeing not the high-impedance coil of wire it was designed to expect, but an active, low-impedance source. Gating or expansion will often be required, but you can usually get away with it on guitars. Although it is always tempting to gate the signal off tape, before the amp feed (while the dynamics are intact), it usually sounds better if you gate it on the return from the amp/processor. The beauty of it is, of course, that you can now experiment with it for as long as you need to — the performance is already captured within the system, just like a MIDI part! Dave Lockwood
I read Paul White's review of the Tandy PZM mic in your July issue and would like to offer a suggestion as to how the mic can be used balanced. The PZM comes with a mono jack fitted, but according to the circuit diagram, the output is transformer coupled via a balancing transformer and the lead between the battery case and the mono jack is already balanced; only the plug is unbalanced. To convert for balanced operation, simply remove the jack and wire the screen, red and black wires to pins 1, 2 and 3 respectively of a male XLR connector.
The benefits of balanced operation can be enormous, especially in environments with computers or other sources of interference. Most mixers, even low cost models, have balanced mic input stages, and once the mic is balanced, it won't be affected if phantom power is applied to it. The mic can't actually use phantom power as it stands, and though some successful mods have been done, these are tricky; the difficulty is compounded by the fact that the circuitry inside the PZM has been known to change from time to time. Replacing the battery with a 9v PP3 is an easier option; it increases the sensitivity and headroom slightly. The new battery won't fit inside the battery box but can be taped to the outside.
The mod described here is of most benefit if the mic is to be used with a long lead, as the section of cable between the mic and the battery box will remain unbalanced. The more adventurous may want to shorten this section of cable to further reduce the risk of interference, but remember, any modification will invalidate the warranty. Andy Groves
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