![]() To handle this bandwidth, this multiplication always runs at a minimum of double the audio sample rate (i.e. This operation is a non-linear process and potentially doubles the bandwidth. The gain cell is a central part of the compressor where the audio signal’s level is adjusted by the control signal delivered by the sidechain. To avoid aliasing in the control signal (which would severely limit the accuracy of the compression), the sidechain uses a bandwidth up to 20x wider than the audible bandwidth. It consists of several non-linear elements such as the threshold, RMS detection and timing filters. The sidechain is responsible for generating a control signal for the gain cell. To achieve this in an efficient manner, the algorithm is split in two parts, both typically running at higher rates than the original signal (given a standard rate such as 44.1kHz or 48kHz). Kotelnikov’s algorithm was carefully designed to avoid these issues. If the sample-rate is too low to handle the extended bandwidth, these harmonics will alias and lose their harmonic relation to the fundamental, resulting in inaccurate, “unwanted” behavior, which in turn directly produces an unpleasant sound. The more aggressive the non-linearity, the stronger and higher the newly generated partials (harmonics). All non-linear systems add harmonic (or non-harmonic) content to the processed signal and thus have the potential to extend the bandwidth significantly. The evil detail here is that contrary to analogue systems, signals exceeding the bandwidth aren’t gradually “faded out”, instead, they mirror at the Nyquist frequency and overlap into the audible range at full energy, and in this process, lose any harmonic relation to the original signal.Ĭompressors are non-linear systems. “digital”) systems as soon the frequency of a signal exceeds the Nyquist rate (half the sample-rate). it needs to know about the values between samples as well).Īnother huge problem in digital dynamics control is a phenomenon called aliasing or Moiré images. However, if one wants to control the amplitude of music accurately, it is absolutely essential to know the actual waveform (i.e. The true analogue signal is only reconstructed by the anti-alias filter inside the D/A converter. It is a very economic, intermediate format. Without delving too deep into the mathematics, it is essential to understand that the data stored in audio-files is not the actual analogue audio signal. “stepped”) nature of digital signal storage. The most significant restrictions are due to the discrete (i.e. A whole array of problems makes it very difficult to build a truly effective and musically attractive compressor in the digital domain. processing not interrupted)Ĭontrolling the dynamic range of an audio signal in the digital domain is not as easy as it looks.
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