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Audio signal flow is the path an audio signal takes from source to output. [1] The concept of audio signal flow is closely related to the concept of audio gain staging; each component in the signal flow can be thought of as a gain stage. In typical home stereo systems, the signal flow is usually short and simple, with only a few components.
Signal flow is the path an audio signal will take from source to the speaker or recording device. Signal flow may be short and simple as in a home audio system or long and convoluted in a recording studio and larger sound reinforcement system as the signal may pass through many sections of a large mixing console, external audio equipment, and even different rooms.
A typical sound reinforcement system consists of; input transducers (e.g., microphones), which convert sound energy such as a person singing into an electric signal, signal processors which alter the signal characteristics (e.g., equalizers that adjust the bass and treble, compressors that reduce signal peaks, etc.), amplifiers, which produce a ...
Audio signal processing is a subfield of signal processing that is concerned with the electronic manipulation of audio signals. Audio signals are electronic representations of sound waves — longitudinal waves which travel through air, consisting of compressions and rarefactions.
PFL sends the channel's signal path to the pre-fade bus. [31] The bus is picked up in the monitor module and made accessible as a substitute signal that is sent to the mixer output. [31] Automatic PFL has been made available, almost universally, and no longer needs to be selected beforehand. [31]
An A/B sound system is a type of sound reinforcement system or public address system. Unlike a more typical sound reinforcement system, an A/B sound system provides two electrically isolated signal paths from microphone to speaker, resulting in a system where signals from two microphones only interact acoustically and never interact electronically.
The system also convolves the head-related transfer function with the impulse response from the signal recorded by the microphones and the energy is adjusted per the original time frame of the sound signal, and an additional delay is added to the sound to match the time frame of the impulse response. The convolution and delays are applied to ...
This original phase vocoder did not take into account the vertical coherence between adjacent frequency bins, and therefore, time stretching with this system produced sound signals that were missing clarity. The optimal reconstruction of the sound signal from STFT after amplitude modifications has been proposed by Griffin and Lim in 1984. [3]