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Melody makers

6th November 2007 [Computer Shopper]

Pop producers often put microphones right inside grand pianos, which is fine if you want a bright, edgy tone. However, if you want the sound of a classical grand piano, you're better off recording from about 10 feet away, where the sound has blended and smoothed out after reflecting off the inside of the piano lid. The same goes for ensembles of acoustic instruments - try to find a place that's not so far that the sound becomes awash with reverberation, but not so near that it over-emphasises the closest instruments. Putting microphones high up in front of an ensemble is an effective way to get a balance of the instruments at the front and the back. You might also want to put individual microphones, known as spot mics, on each instrument, but this often isn't necessary if you just want to capture the natural sound of a live performance.

In practice, the most effective way to find the best microphone position is to move your head around until you find a place where the instrument sounds best. However, this isn't possible if you're recording your own playing, in which case you'll need to jump to the next stage in the process. Put on a pair of headphones, set up the microphone and slowly move it around and hear how the sound changes. If your headphones don't cut out enough sound going directly from the instrument to your ears (which is bound to be true when recording very loud instruments such as drums), you'll need to do short test recordings to see the effect of microphone positions.

Use stereo pairs of microphones whenever possible. Except for point-source instruments such as brass and vocals, recording with two microphones and panning them to the left and right speakers will produce a sound that seems far more detailed and spacious. To re-create the sound of two ears in a room faithfully, position the microphones a head's width apart, facing at a 90 angle from each other and towards the performer. For pop recording, where such faithful accuracy isn't so
 
 
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important, feel free to experiment to get a pleasing sound.

Ok computer

The walkthrough tutorial later on will guide you through the various stages of recording and editing audio and MIDI performances on your PC. It may seem a complex process, but if you break it down into simpler steps it will be less daunting. That said, there are various extra details to consider that can make your recording sessions run more smoothly and yield better results.

A low-cost PC can do a very convincing impression of a top-flight recording studio. Modern music-production software emulates multi-track tape recorders, huge mixing desks and racks upon racks of effects processors, and can also turn its hand to grand pianos, analogue synths and symphony orchestras in the form of virtual instruments. The virtual instrument plug-ins have the advantage of being more affordable and better quality than hardware synths, and best of all, they integrate seamlessly with your recording software to give a fully self-contained recording environment.

However, there are compromises. One is that it's harder to visualise audio signal paths when they all exist within a single box with various pop-up windows for visual feedback. Another is that, unlike analogue audio devices, digital audio processing takes a quantifiable amount of time. Even if it's only a few milliseconds, this can cause problems.

Audio exists as a continuous stream, but music-production software needs to work with discrete blocks of audio. Once it has processed a block, it sends it to its next destination and gets working on the next. The size of each block limits how quickly the software can react to user input. A 50-millisecond block means there will be 50ms latency (delay) after pressing record, start or stop for the software to respond - not a huge problem. However, if you play or sing a note and have to wait 50ms to hear the result, this can be distracting. Low-latency drivers are essential. They allow the software to communicate with the audio interface using very small blocks; three to 10ms is considered acceptable when working with virtual instruments. Two types of driver - WDM and ASIO - can deliver this performance. We find ASIO is generally more reliable. Reducing the latency (sometimes known as the buffer size) too much can increase the load on the computer and cause glitches in recording, so experiment with settings before you have a critical recording job to do. Most audio software presents its latency setting in the same place as its audio driver settings.

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