Effects Processor Articles
How Digital Effects Processors Work |
| Posted by Administrator (admin) on Feb 12 2008 |
With the advent of the digital age, the ability to combine more than one type of effect within a single device was made possible by microchips and transistors. Instead of using physical hardware to process a signal, equipment makers were able to write computer code that simulated the analog effects they had previously achieved. They also opened the door to new effects and techniques that had never been previously imagined.
Audio effects processors have come a long way since people began to experiment with methods for altering an electronic sound signal. Initially, effects were analog in nature and employed a number of different methods to achieve the desired results. Reverb was achieved via metal plates or steel springs and distortion came from degrading the signal or adding noise or gain. Sometimes analog effects were quite elaborate, such as the Leslie rotating speaker which produced a phased effect by actually moving the speaker 360 degrees.
Modern effects processors are digital as well as analog based. A digital effects processor essentially transforms an audio signal by sending it into a processor where it is sampled and converted into binary code – ones and zeros. Sampling is a process by which thousands of sound images are taken of an analog signal per second. Sampling occurs at 44 MHz - or 44,000 sound images – or 96 MHz for most recording purposes. These images form the digital representation of the signal. What makes digital effects processors so powerful is their ability to manipulate this digital information with an incredible degree of accuracy and precision.
Once the signal has been digitized, it is then analyzed and processed by the programming of the effects processor. A good example of how this occurs can be seen with the application of a digital reverb effect on a signal. Typical parameters that a musician might want to be able to control when applying reverb are room size, saturation, and decay. Once these are entered into the device, the processor’s programming begins to construct a digital simulation of the conditions that would be required to generate the desired sound. Using a complex series of mathematical calculations based on painstaking research into actual sound behavior, the processor is able to recreate how the sound would be reflected by the walls in that specific room in combination with the desired amount of decay. It then applies this formula to the signal and mixes the correct amount of dry signal with the reverb output to create the dialed in sound. The signal can then be converted once again into analog if it is being sent out to an amp or mixing board or it can be digitally transmitted directly into a software workstation.
Obviously, this is a simplification of a complex process. What is most important to remember is that once it is digitized, the sky is the limit in terms of how a signal can be transformed or edited. Processors are essentially only limited by the power of their computer chips and the creativity of both the original programmers and those using them to shape their own sound.
Last changed: Feb 12 2008 at 8:05 AM
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