AUDIOREPAIR – enhance your listening experience
- Windows 7 – 10 / 32 & 64 Bit
- Mac OSX Lion 10.7.3 – High Sierra 10.13.x
Inter Sample Peak Repair
Frequency Response Repair
Compact Disk De-Emphasis
64Bit High Precision Audio Engine
Input Audio Formats (16Bit / 44.1kHz): WAV, AIFF, FLAC, ALAC and MP3
Output Audio Formats (16Bit / 44.1kHz or 88.2kHz): WAV, AIFF and FLAC
Batch processing with freely configurable output file names
32x Up-Sampling to 1.4112MHz to detect Inter Sample Peaks
Metadata (album, cover picture, etc.) transfer for AIFF to AIFF and FLAC to FLAC conversion
The AudioRepair can be fully tested before purchase. Only the playback and analysis is limited to 30 seconds.
The activation allows 3 parallel installations. In fact, it is possible to use the Windows and MacOS X Version in parallel by just using one activation key.
WHY SHOULD I USE AudioRepair?
INTER SAMPLE PEAKS
The 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 RANGE
The 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 (½ x Sampling Frequency => ½ 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.
COMPACT DISK DE-EMPHASIS
AudioRepair optionally applies a specified De-Emphasis curve to linearize older CD-Records (until the early 90s) that used Emphasis to achieve a better reduction of noise. Especially the suppression of audible quantization noise during playback was extremely important, because early analog to digital converters had only 14 significant bits.
Some of todays CD-Rippers already apply the De-Emphasis if the correct information bit is set.
SHOW ME AUDIOREPAIR IN ACTION!
REPAIRING INTER SAMPLE PEAKS
The following screenshot shows an audio track, analyzed with the MusicScope that contains numerous Inter Sample Peaks of up to around +3dB True Peak Level (TPL).
We use AudioRepair (pls. see screenshot below) to just repair the Inter Sample Peaks (ISPs).
The repair process is straight forward by loading the audio file or several audio files via the “Load Dialog” or “Drag & Drop”.
Selecting “Inter Sample Peak Repair” and de-selecting “Frequency Response Repair” makes sure that the AudioRepair tool only removes the ISPs.
It is possible to configure the location and the naming of the output file.
The tool either generates WAV or AIFF output files but it is able to read various input formats WAV, AIFF, FLAC, ALAC and MP3.
Start the repair process by clicking on “Start”. The State of each audio track is presented within the batch-processing list, whereas the overall progress is shown in a progress-bar at the bottom of the window.
During the Pre-Processing the audio signal is up-sampled to 1.4112MHz (32x 44.1kHz) to emulate an analog signal to detect the maximum amplitude of the Inter Sample Peaks. During the Processing a high precision 64Bit audio engine reduces the amplitude to make sure that the level within the analog domain stays below 0dB full scale avoiding any distortions which could impact the listening experience. During the whole processing the integrity of the audio signal is perfectly preserved.
The following MusicScope Screenshot shows the repaired audio track now reaching a maximum of 0dB True Peak Level.
REPAIRING THE FREQUENCY RESPONSE
The following MusicScope screenshot shows a typical audio track using the full frequency range up to 22.05kHz, making it difficult for a Digital to Analog Converter to do its job without introducing aliasing.
Similar to the Inter Sample Repair process the audio files to be worked on are loaded via “Load Dialog” or “Drag & Drop”.
Select “Frequency Response Repair” to activate the function.
Attention: “Inter Sample Repair” is a precondition for “Frequency Response Repair” and gets automatically activated.
1.) FIR – Steep Response
The FIR Steep Response Low Pass Filter starts to reduce the amplitude at 20 kHz, reaching -96dB at 22.05kHz. To achieve this steep filter curve it needs a longer impulse response with a bit more pre-ringing. Nevertheless, for most cases we recommend to use this filter type.
2.) FIR – Smooth Response
The FIR Smooth Response Low Pass Filter already starts to reduce the amplitude at 18 kHz with a smooth filter curve reaching -96dB at 22.05kHz. The advantage of this filter is its short impulse response producing a minimum of pre-ringing.
Finally it is possible to set the output sampling frequency by selecting either 44.1kHz or 88.2kHz.
A higher sampling rate has its advantages during the digital to analog conversion where a DAC needs to apply a steep low pass filter, probably impacting the music signal. With a two times higher sampling rate the reconstruction filter acts far away from any music signal. Dedicated DACs even allow a setting of smooth filter responses or IIR instead of FIR filters. Those configurations in combination with a higher sampling rate are able to achieve great results.
The following MusicScope screenshot represents the repaired audio track reaching -96dB at 22.05kHz as specified in the CD-Standard.