Modern Recording Techniques 7th Edition Ch 14 Signal Processing - Posted on 2013-06-26 by Malecia Bynum

• It's the function of a signal processor to change, augment or otherwise modify an audio signal in either the analog or digital domain. 
• Plug-ins offer virtually every processing function imaginable, They are designed to be integrated into an editor or DAW production environment in order to perform a particular real-time or non-real-time processing function. 
• Currently, several plug-in standards exist, each of which functions as a platform that serves as a bridge to connect the plug-in through the computer's operating system (OS) to the digital audio production software.
• A signal processing device can be inserted into an analog or digital chain in several ways. The most common of these are: 
  • Inline routing 
  • Parallel routing
• Inline routing is often used to alter a signal. It occurs whenever a processor is inserted directly into a signal path in a serial fashion. This method for inserting a device is generally used for the processing of a single instrument, voice or grouped signals that are present on a particular line. *Often but not always this device tends to be a level-based processor (such as an equalizer, compressor or limiter).*
• Devices that offer an external "key" input can be quite useful, allowing a signal source to be used as a control for varying another audio path. 
• Send routing is often used to augment a signal (generally by adding reverb, delay or other time based effects). It occurs whenever a portion of the original signal is allowed to pass through the chain while a side signal is simultaneously fed to an effects device...Once effected the signal is then proportionately mixed back in with the original signal to create an effects blend. 
• The spectral content of a sound: in the form of equalization or intelligent equalization and bandpass filtering 
• Amplitude level processing: in the form of dynamic range processing
• Time-base effects: Augmentation or re-creation of room ambiance, delay, time/pitch alterations and tons of other special effects that can range from being sublimey subtle to "in yo face".
• An audio equalizer is a circuit, device or plug-in that lets us exert control over the harmonic or timbral content of a recorded sound, 
• EQ refers to the alteration in frequency response of an amplifier so that the relative levels of certain frequencies are more or less pronounced than others. 
• Older EQ's and newer "retro" systems often base their design around filters that use passive components (i.e. inductors, capacitors and resistors) and employ amplifiers only to make up for internal losses in level, called insertion loss.
• Most EQ circuits today however, are of the active filter type that change their characteristics by altering the feedback loop of an operational amp. This is by far the most common analog EQ type and is generally favored over its passive counterpart due to its low cost, size and weight, as well as its wide gain range and line-driving capabilities. 
• The most common EQ curve is created by a peaking filter. As its name implies, a peak-shaped bell curve can either be boosted or cut around a selected center frequency. 
• The quality factor of (Q) of a peaking equalizer refers to the width of its bell-shaped curve. A curve with a high Q will have a narrow bandwidth with few frequencies outside the selected bandwidth being affected, whereas a curve having a low Q is very broadband and can affect many frequencies (or even octaves) around the center frequency.
• Bandwidth is a measure of the range of frequencies that lie between the upper and lower -3dB (half power) points on the curve.
• The Q of a filter is an inverse measure of the bandwidth (such that the higher Q values mean that fewer frequencies will be affected, and vice versa). 
• Another type of EQ is the shelving filter. Shelving refers to a rise or drop in frequency response at a selected frequency which tapers off to a preset level and continues at that level to the end of the audio spectrum. 
• EQ types also include high-pass and low-pass filters. As their names imply, this EQ type allows certain frequency bandwidths to be passed at full level while other sections of the audible spectrum are attenuated. Frequencies that are attenuated by less than 3 dB are said to be inside the passband; those attenuated by more than 3 dB are in the stopband. The frequency at which the signal is attenuated by exactly 3 dB is called the turnover or cutoff frequency and is used to name the filter frequency, 
• Commonly, attenuation is carried out at rates of 6, 12, and 18 dB per octave. This rate is called the slope of the filter. 
• A high-pass filter in combination with a low-pass filter can be used to create a bandpass filter, with the passband being controlled by their respective turnover frequnecies and the Q by the filter's slope. 
• The 4 most commonly used types that can incorporate one or more of the previously described filter types are the:
  • Parameter equalizer
  • Selectable frequency equalizer
  • Graphic equalizer
  • Notch filter
• The parametric EQ lets you adjust most or all of its frequency parameters in a continously variable fashion. Because of its fleibility and performance, the parametric equalizer has become the standard design for most input strips, digital equalizers and workstations.
• The selectable frequency EQ as its name implies has a set number of frequencies from which to choose. These EQs usually allow a boost or cut to be performed ar a number of selected frequencies with a predetermined Q. 
• A graphic equalizer provides boost and cut level control over a series of center frequencies that are equally spaced (ideally according to music intervals). *The various EQ band controls generally use vertical sliders that are arranged side by side so that the physical positions of these controls could provide a "graphi" readout of the overall frequency response curve at a glance. 
• Notch filters are often used to zero in on and remove 60- or 50-Hz hum or other undesirable discrete-frequency noises. (They use a very narrow bandwidth to fine-tune and attenuate a particular frequency in such a way as to have little effect on the rest of the audio program.
• The audio spectrum can be divided into 4 frequency bands: low (20-22 Hz), low-mid (200-1000 Hz), high-mid (1000-5000 Hz) and high (5000-20,000 Hz).
• The fundamental notes of most instruments lie within the 200-1000 Hz (low-mid) range.
• A compressor, in effect can be thought of as an automatic fader. It is used to proportionately reduce the dynamics of a signal that rises above a user-definable level (known as the threshold) to a lesser volume range. 
• The most common controls on a compressor (and most other dynamic range devices) include input gain, threshold, output gain, slope ratio, attack, release and meter display:
  • Input gain: This control is used to determine how much signal will be sent to the compressor's input stage
  • Threshold: This setting determines the level at which the compressor will begin to proportionately reduce the incoming signal. Most quality compressors offer hard and soft knee threshold options. A soft knee widens or broadens the threshold range, making the onset of compression less obtrusive, while the hard knee setting causes the effect to kick in quickly above the threshold point. 
  • Output gain: This control is used to determine how much signal will be sent to the device's output. It's used to boost the reduced dynamic signal into a range where it can best match the level of a medium or be better heard in a mix. 
  • Slope ratio: This control determines the slope of the input-to-output gain ratio. In simpler terms, it determines the amount of input signal (in decibels) that's needed to cause a 1-dB increase at the compressors output. 
  • Attack: This setting (which is callibrated in milliseconds; 1 msec = 1 thousandth of a second) determines how fast or how slowly the device will turn down signals that exceed the threshold. 
  • Release: Similar to the attack setting, release (which is calibrated in milliseconds) is used to determine how slowly or quickly the device will restore a signal to its original dynamic level once it has fallen below the threshold point (defined as the time required for the gain to return to 63% of its original value). 
  • Meter display: This control changes the compressor's meter display to read the device's output or gain reduction levels. 
  • To achieve hot levels without distortion, multiband compressors and limiters often are used during the mastering process to remove peaks and to raise the average level. 
  • (Continue on page 496)