AudioRepair for Mac

Name: AudioRepair
Rating: 4 (1 reviews)
Author: XiVero GmbH

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Key Details of AudioRepair

The AudioRepair Tool works on the most prominent issues of today's digital standard recordings.
Last updated on April 21, 2015
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Developer’s Description

By XiVero GmbH

The AudioRepair Tool works on the most prominent issues of today's digital standard recordings.

INTER SAMPLE PEAKSThe so-called Inter Sample Peaks are caused by faulty signal limiting during recording or intentional loud mastering. Some statistics state that around 80% of today's CD recordings show Inter Sample Peaks.A digital signal cannot exceed 0dB Full Scale but the analog signal, re-constructed out of the digital domain, is able to reach levels of +3dB, overloading heavily the Digital to Analog Converters (DACs) and therefore creating audible distortions.The AudioRepair tool is able to fix the Inter Sample Peaks by doing a 32 times up-sampling with a target sampling frequency of 1.4112MHz to simulate an analog signal for an accurate audio processing.The repair process adjusts the audio track level in a way that the resulting analog signal does not exceed 0dB to stay in the specification limits of modern DACs.INSUFFICIENT LIMITATION OF THE FREQUENCY RANGEThe CD standard defines a frequency response of 20Hz - 20kHz to assure that the sampling frequency of 44.1kHz provides enough room to do the Digital to Analog Conversion by applying a brick wall Low Pass Filter.Recordings from the 80's stick to the standard by reaching levels of -96dB at 22.05kHz but modern recordings exceed amplitudes of -60dB at the Nyquist-Frequency (1/2 x Sampling Frequency => 1/2 x 44.1kHz = 22.05kHz), making it difficult for modern DACs to convert the audio signal into the analog domain without introducing aliasing.The AudioRepair tool applies a linear phase filter to make sure that the audio signal stays within the specification. The software provides the option to choose between two different FIR filters with their dedicated advantages.It is possible to save an up-sampled version of the repaired audio track.Even if there is no musical content beyond 20kHz it could make sense to convert a 44.1kHz track to 88.2kHz to avoid a negative influence on the frequency response during the digital to analog conversion.Even modern DACs need to apply steep FIR/IIR low pass filters to do their job. By having a two times higher sampling frequency the impact of those filters is far away from the music signal, preserving the listening experience.