Audio Engine Settings | Resampling, DSD & Dithering

Audio Engine

Audio samples are processed with accurate and ultra-precise (64-bit floating point) algorithms.

Sample Rate Converter

SoX resampler

Audio Engine uses high quality SoX resampler (licensed under LGPL).

SoX resampler is used for all sample rate conversions including the internal resampling requirements of all codecs used in EZ CD Audio Converter (Opus, MP3, AAC, DSD, etc).

Sample rate conversion is gapless which means there are no erroneous samples on track transitions of the resampled audio signal.

Sample Rate Converter quality tests

The SoX sample rate converter used in EZ CD Audio Converter has been tested in 3rd party sample rate converter quality tests at https//src.hydrogenaudio.org.

Quality Test Reports

Passband

Resampler's passband setting determines how much of the frequency content of the original signal (wrt the original sample rate when up-sampling, or the new sample rate when down-sampling) is preserved during conversion. The term 'passband' is used to refer to all frequencies up to the bandwidth point (e.g. for 44.1kHz sampling rate, and a resampling passband of 95%, the passband represents frequencies from 0Hz to ~21kHz). Increasing the resampler's passband results in a slower conversion and can increase transient echo artefacts.

Default | 95% passband

Default sweep 95% - Conversion from 96kHz to 44.1kHz

Default passband 95% - Conversion from 96kHz to 44.1kHz

Steep | 99% passband

Steep sweep 99% - Conversion from 96kHz to 44.1kHz

Steep passband 99% - Conversion from 96kHz to 44.1kHz

Max | 99.7% passband

Max sweep 99.7% - Conversion from 96kHz to 44.1kHz

Max passband 99.7% - Conversion from 96kHz to 44.1kHz

Allow aliasing

Aliasing above the passband is allowed. For example, with 44.1kHz sampling rate, and a resampling passband of 95%, this means that frequency content above ~21kHz can be distorted; however, since this is above the passband (i.e. above the highest frequency of interest/audibility), this may not be a problem. The benefits of allowing aliasing are reduced processing time, and reduced (by almost half) transient echo artefacts.

Aliasing is automatically disabled if the source or destination sample rate is below 44.1kHz

Default | 95% passband | Allow aliasing

Default sweep 95%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Default passband 95%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Steep | 99% passband | Allow aliasing

Steep sweep 99%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Steep passband 99%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Max | 99.7% passband | Allow aliasing

Max sweep 99.7%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Max passband 99.7%, aliasing allowed - Conversion from 96kHz to 44.1kHz

Dithering

Why dither ?

When the bit depth is reduced (quantized) there will be quantization errors. I.e. lower bit depth can't accurately represent the source signal. When each sample value is rounded to the nearest value, then the steppiness in the audio may be audible, especially with very quiet sounds. Dither (intentionally applied form of noise) can be applied to avoid it.

Dither can be applied (when enabled from the Settings) if converting from original source of
· 64-bit, 64-bit float, 32-bit, 32-bit float to 24-bit
· 64-bit, 64-bit float, 32-bit, 32-bit float, 24-bit to 16-bit
· 64-bit, 64-bit float, 32-bit, 32-bit float, 24-bit, 16-bit to 8-bit

Noise-shaped

Noise-shaped dither has higher peak amplitude than other forms of dither, but the noise is concentrated in the very high frequency range where it is less audible. Noise-shaped dither implements Lipshitz minimally audible noise shaping.

TPDF (Triangular)

Triangular dither produces a lower peak amplitude of noise than noise-shaped, but the noise is concentrated less in the higher frequencies.

RPDF (Rectangular)

Rectangular dither produces a similar noise amplitude as triangular, but with very small amounts of randomization. The frequency distribution is fairly even, so that it approximates white noise. Unlike the other two forms of dither, rectangular does not add noise to absolute silence.

RTN (Round to nearest, default, always on)

Round samples to nearest value (better audio quality than alternative of truncating samples)

EBU R 128 | ReplayGain Algorithm

These settings control how loudness and peak values are measured when writing loudness metadata (EBU R 128 | ReplayGain). The chosen method determines how playback volume is normalized and how consistently tracks match in perceived loudness.

Loudness Algorithm Options

Integrated
Calculates the average loudness over the entire duration of a track or album. This is the official EBU R 128 method and is widely used because it produces consistent and predictable loudness normalization across all types of audio content.
Top of Loudness Range
Focuses on the louder portions of the audio rather than the full average. This can better represent how loud a track feels during its most intense sections, making it useful for highly dynamic music where peaks dominate perception.

Peak Measurement Options

Sample Peak
Measures the highest amplitude found directly in the digital samples of the audio file. It is fast and commonly used, but it does not detect peaks that may occur between samples, which can sometimes lead to underestimating true signal peaks.
True Peak
Estimates the actual analog peak level by reconstructing the signal between samples (inter-sample peaks). This provides a more accurate indication of potential clipping and is especially important for lossy formats and modern playback systems.

DSD -> PCM

Quality setting

DSD to PCM conversion decimation low pass filter quality setting. Both filters provide flat response up to 30 kHz [DSD64], 50 kHz [DSD128], 100 kHz [DSD256].

Very High Quality

Faster, looser decimation low pass filter. Noise level below -176 dB.

Ultra High Quality

Slower, more accurate, steeper decimation low pass filter. Noise level below -191 dB.

DSD64 to PCM noise levels

Lossy to lossless format conversion Auto bit depth

Lossless formats "Auto" bit depth takes decoder bit depth as input. Because lossy formats such as MP3, Opus, AAC and Vorbis are decoded as 32-bit floating point the lossless output file can be unnecessarily accurate producing very large files. This option overrides the "Auto" bit depth and allows "Auto" bit depth to be set to more reasonable accuracy when converting Lossy to Lossless. Default is 16-bit (CD quality).