Signal Processors

Signal processors are devices or software used to alter a characteristic of sound. A plug-in is an add-on software tool that provides a DAW with more signal processing alternatives than what is built in.

Spectrum processors affect the frequency aspect of the signal.
Equalizers (or EQ) alters the frequency response by increasing or decreasing the level of a signal at a specific portion of the spectrum. The increase or decrease is down around a center frequency, which is the most affected value while those around it have gradually lesser changes. The range of frequencies affected is called the bandwidth. Shelving affects all frequencies above or below the selected frequency equally. There are fixed frequency EQs with a certain number of fixed center frequency options, generally knobs for high, upper middle, lower middle and low frequencies. Graphic EQ uses sliders instead of knobs. Parametric EQ allows for continuously variable frequencies and bandwidth size, which gives more flexibility and precision.

Filters attenuate bands of frequencies, usually at a preset and with a steep drop. High pass (low cut) cuts frequencies below the preset point. Low pass (high cut) cuts frequencies above a preset point. Band pass filters have both a high and low cutoff point and allows the frequencies between them through. A Notch filter cuts out an extremely narrow band, such as the hum at 60 Hz.

Psychoacoustic processors dd clarity and definition with EQ and harmonics.

Time processors affect the time relationships in a signal.
Reveration is created by random multiple blended repetitions of a sound. Dry sound is without reverb, we sound is with added reverb.
Digital reverb is the most commonly used, where the original signal is delayed and attenuated multiple times and then added to the original signal. Predelay is the time between the direct sound and the early reflections, and tells you the room size.
Convolution reverb is a sample based process that multiplies the spectrums of 2 audio files. The first is the acoustic signature of a space, called the Impulse Response. It is acquired by recording the room response after a quick impulse is played in the room. The second file is the source signal.When the two files are multiplied, it applies the room characteristics to the original signal, overlaying any room or location.
Plate reverb is a from a mechanical-electronic device with a thin steel plate that vibrates and is miked and sent through a console.
Acoustic chambers are dedicated rooms that create realistic reverb, but are very expensive to build.

Choose a reverb that sounds natural, with bright highs and clear lows. Listen to vocals to check for clearless, and sharp transients like a drumbeat for density.

Delay is the time interval between a sound and its repetition. Digital delay routes audio through an electronic buffer and holds it for a specific amount of time. The delay time is how long the sound is held. Feedback is how much of the delayed signal is returned. Higher feedback increases the number of repetitions and longer decay. No feedback means just one repetition.
Uses of delay include doubling, chorus, slapback echo, and prereverb delay.
Flanging is the original signal combined with a 0-20ms time delayed replica. Phase cancellations occur and create comb-filter efects with peaks and dips in the frequency response that lead to a holly, swishy sound. In phase (positive flanging) accents the even harmonics for a metallic sound, while out of phase (negative flanging) accents the odd harmonics for a warm sound.
Phasing uses a phase shift instead of a time shift, giving more irregular peaks and dips for a wavering vibrato and less effect on the pitch.
Morphing is a continuous seamless transition from one signal to another. It is not crossfading - the signal takes on actual characteristics of the other sound. Examples are available here.

Amplitude processors affect the dynamic range of the signal.
Compressors output level increases at a slower rate than the input level increases, and therefore restricts the dynamic range for peak signal limitations. The compression ratio establishes the proportion of change between the input and output levels. This can range from 1.1:1 to 20:1, which means that every change in 20db for the input only gives 1db in output change. The compression threshold is the level where the ratio takes effect. When it is reached, compression begins, which reduces the gain according to the amount the signal exceeds the threshold level and the ratio set. Knee is the moment the compressor starts gain reduction; hard knee is abrupt while soft knee is smoother. Broadband compressors act on the dynamic range of the input signal across the entire frequency spectrum, while split band compressors affect the input signal independently by splitting the audio into multiple bands.

A Limiter is a compressor where the output level stays the same at a preset point, no matter what the input level is. Basically gives a ceiling on the loudness of a sound, and reduces the high frequency response.

De-essers are a fast acting compressor that attenuate high frequencies to remove hissy consonants in s, z, ch, and sh vocals.

Expanders increase the dynamic range, essentially the opposite of compressors. They are triggered when the signal falls below a set threshold at a set ratio. A noise gate is used to reduce or eliminate low-level noise from amplifiers, ambience, rumble, noisy tracks, etc.

Pitch Shifters use time compression and expansion to change a flat or sharp pitch to be in tune. With time compression, the signal runs faster and raises the pitch. Time expansion runs it slower and lowers the pitch.

Noise processors reduce noise using DSP (Digital Signal Processing), removing clicks, cracks, humming, etc. Multieffects processors combine a number of functions into one unit.

What is Sound

What is sound? Sound is the vibrations of molecules in the air as the energy is passed in waves of compression and rarefactions. It can be pictured similar to when a stone is dropped in a lake, and the radiation of the waves from that point.

The five main components of sound are: Frequency, Amplitude, Velocity, Wavelength, and Phase.

Frequency is the number of cycles a wave completes in one second, and is measured in Hertz (Hz). For example, if 500 cycles per second are completed, the frequency is 500Hz.

Pitch is how humans psychologically perceive frequency.

Amplitude is the wave's maximum height (crest) and minimum (trough). This is perceived as loudness. Loudness is measured in decibels, which is on a logarithmic scale in powers of 10. In order to double loudness, an increase of 6 dB-SPL is needed.

Velocity is the speed of a sound wave. Generally this is 1130 ft/sec at 70 degrees F at sea level through air. Thicker materials slow down the molecules - 4800 ft/sec in water, 11700 ft/sec in wood. Also, as air warms up, the speed of sound increases, and the inverse is also true.

The wavelength is exactly what is sounds like - the distance a sound wave travels to complete one cycle. Longer wavelengths correlate to lower frequencies, while higher frequencies have shorter wavelengths.

Sound movement examples: http://www.fearofphysics.com/Sound/dist.html
Decibel vs. loudness charts: http://www.gcaudio.com/resources/howtos/loudness.html
Reflection examples: http://www.kettering.edu/~drussell/Demos/reflect/reflect.html

Questions:
1. What is timbre? How does it define different instruments?
2. How does the inner ear convert sound waves to electrical pulses for the brain to understand?