Instrument level signals represent the audio signal strength generated from musical instruments. Generally, these are stronger than microphone levels but reside in a middle-ground between line level and mic level signals. They also have a high-impedance characteristic, meaning they have high output impedance and very low current relative to the output voltage.
Pro line level and consumer line level are designed for different applications and requirements. Pro line level is suited for high-impedance and long-distance transmission, using balanced signal transmission. Consumer line level is optimized for short-distance unbalanced signal transmission and is relatively less expensive to manufacture.
In pro line levels, +4dBu is commonly used as the reference level.
+4dBu corresponds to a signal level based on 1.228V, and it is the standard signal strength commonly used among professional audio equipment. Transmitting at levels higher than this would be inefficient due to excessive power consumption of the preamplifier, and transmitting at lower levels may result in signal degradation due to the typical impedance of XLR or TRS cables.
It’s important to understand what a bit is at this point. A bit, short for “Binary Digit,” is the smallest unit for representing digital data. A bit can have one of two values, typically represented as 0 or 1.
Bits serve as the basic unit for representing information in computer systems. Using binary code, various types of data such as numbers, characters, images, and sounds can be represented. For example, an 8-bit binary code can express 256 different values through an 8-digit number composed of 0s and 1s.
Bits play a vital role in the basic data processing and communication in computer systems. Combining multiple bits allows for the representation of more complex data, and the number of bits affects the range and precision of the data. The more bits used, the higher the number of value combinations and the greater the precision.
For example, 8-bit can represent 2^8 (256) different values, ranging from numbers 0 to 255, alphabets, and symbols. Similarly, 16-bit can represent 2^16 (65536) values, and 32-bit can represent 2^32 (about 4.3 billion) values.
Harry Nyquist, after whom the Nyquist Theory is named, was an electrical engineer at Harvard University. He made significant contributions to signal processing and information theory in the early 20th century.
Known as a pioneer in the field of sampling theory and information theory, Nyquist presented the concept of sampling frequency in his 1928 paper “Certain Topics in Telegraph Transmission Theory.” This concept later became known as the “Nyquist Theory.”
The sampling rate refers to the frequency of the sampling process in hertz (Hz).
For instance, a 44.1kHz sampling rate means that 44,100 samples are taken per second. Higher sampling rates can capture a wider frequency range, thus more accurately reproducing higher frequency components. CD-quality audio typically has a sampling rate of 44.1kHz.
Digital and Analog are two primary forms of representing data or signals.
Analog signals possess continuous values, with smooth transitions in terms of time and magnitude. For instance, natural sounds, music, or human voices are examples of analog signals. Analog signals have an infinite number of possible values, capable of depicting continuous changes accurately.On the other hand, digital signals have discrete values and are represented in a binary format of 0s and 1s. Digital signals are derived from analog signals through a process of sampling and quantization. Digital signals, being represented in binary code, are easier to process in computers and digital systems. They’re less susceptible to noise, allowing for error detection and correction.
(While it’s typical for digital to be expressed as binaries of 0s and 1s, not every combination of 0 and 1 is digital. Digital refers to representing a continuous analog signal in discrete values. While these discrete values are typically represented as 0 and 1, other values can be used.)
dB represents a unit of a physical quantity derived by taking the common logarithm of a ratio compared to a reference. The name of this unit originates from the inventor of the telephone, Alexander Graham Bell.
dBSPL represents a value of sound level expressed on a logarithmic scale, converting the pressure of the sound into dB. Through this, we can express the intensity and magnitude of the sound as perceived by humans.
Sound arises from the change in air pressure due to molecular motion, and this change in pressure is referred to as Sound Pressure. Sound propagates in the form of waves, where the magnitude of the sound is determined by the intensity of the pressure.
The logarithmic scale used in dBSPL is useful for representing the relative size of sounds. Using a logarithmic scale makes it easy to represent even when the range of sound levels is broad. This scale reflects the perceptual properties of sound and assists the human auditory system to distinguish between a wide range of sound levels.
dBFS (Decibels Full Scale) is a unit used in digital audio. Given that the maximum value varies depending on the bit depth, 0dBFS represents the maximum signal level for a specific bit depth, illustrating the peak level. To prevent clipping, common reference levels (Rms) are typically set at -18dBFS (for 24 bits or higher) or -12dBFS (for 16 bits).