Why do we at Northern Audio love HQPlayer? It’s all about the filters! Any DAC uses filters to achieve it’s signature sound. HQPlayer allows you to sample some of the finest filters out there, IF you have the PC power for it.

Grigg Audio Solutions in Rostock, Germany has nicely summarized why the HQPlayer’s filter properties, user interface, and high quality algorithms are masterfully combined for an unrivaled audio experience. Their information is used below with permission to help get you started.

Want to learn more? Come see us for a free consultation on how to build your Ultimate Custom HQPlayer PC or Home Theater PC! Call today at 412.931.5850.

Audio PC HQPlayer Filter Properties

Table of contents

1. Filter qualities
2. Pre-ringing and post-ringing
3. Filters for finite impulse response (FIR – linear phase) and infinite impulse response (IIR – minimum phase)
4. Effects on space and transients and music genres
5. Apodizing and non-apodizing
6. Upsampling ratio
7. Compressed music
8. Table with filter properties

Although the HQPlayer can be used as an audio player in Windows, Linux and macOS, it cannot compare to Roon in terms of ease of use. However, when it comes to upsampling, the HQPlayer is worlds better than Roon. The ability to integrate the HQPlayer into Roon means that the intuitive user interface and the highest quality algorithms can be combined.

The HQPlayer offers filters which

– are apodizing and non-apodizing, – have
different effects on space , transients and timbre
, – prefer different sampling rates (CD or HiRes),
– require low or high computing power and
– even support different music genres .

The whole thing is quite confusing, which puts many people off. We’re trying to shed some light on the matter.

Filter qualities

For a CD (44.1 kHz) the audible range may only go up to a maximum of  22.05 kHz  . This is based on the  Nyquist-Shannon sampling theorem , which should only take half the sampling rate ( Nyquist frequency ) into account. If the sampling theorem is violated by a sampling rate that is too low, frequency components that were originally higher than half the sampling rate are interpreted as lower frequencies, since they are undersampled. This undesirable phenomenon is called the aliasing effect  . These so-called mirror frequencies should be blocked out by filters, which is why these filters are often also called anti-aliasing filters .

It therefore depends on the  filter quality  which frequencies are allowed through in practice ( passband ), at what point the blocking effect ( cutoff ) begins and how long it takes ( transition range ) until the low-pass filter develops its full effect ( stopband ). Independent measurements have shown that the HQPlayer filters are very well constructed.

If the  blocking effect is very low , this is referred to as  extremely  high attenuation . This is a good thing because image frequencies are effectively suppressed.  High attenuation leads to fewer noise artifacts and also to better reconstruction accuracy.

Equally important is when the cutoff begins. Some filters start in the audible frequency range, which is of course not desirable. A steep filter is also beneficial, as it achieves a very narrow transition band . If we take the example of an input rate of 44.1 kHz (CD), the passband of the “poly-sinc-gauss-xl” filter goes up to 20 kHz, the transition range is extremely narrow at 250 Hz and reaches maximum attenuation at the Nyquist frequency  fs/2  at 22.05 kHz. Filters of this high quality are often referred to as having a fast or sharp “rolloff” or “cutoff” . See also:

• Better understand the filters in HQPlayer .

Pre-ringing and post-ringing

Pre-ringing and post-ringing are two types of artifacts that can occur in digital signal processing. Pre-ringing is the phenomenon that a signal rises or falls before an abrupt transition or before it reaches its actual value. Post-ringing is the phenomenon that a signal oscillates after an abrupt transition before it stabilizes.

Both effects can affect the quality and accuracy of a signal and can cause audible effects. Pre-ringing in particular can be very disturbing because it does not occur in nature.

When designing a filter, it is important to keep pre- and post-echoes as low as possible. They cannot be completely avoided.

Filters for finite impulse response (FIR – linear phase) and infinite impulse response (IIR – minimum phase)

FIR and IIR filters are two types of digital filters used for signal processing. They differ mainly in their impulse response and phase characteristics.

FIR (Finite Impulse Response) filters have a finite impulse response that decays to zero after a certain time. They have a linear phase, which means that all frequencies have the same delay through the filter. They are always stable because they have no feedback loops. However, they require more coefficients and storage space than IIR filters to achieve a similar frequency response. They can also produce pre-ringing that precedes the actual impulse.

IIR (Infinite Impulse Response) filters have an infinite impulse response that decays infinitely. They have minimum phase , which means that different frequencies have different delays through the filter. They can be unstable if the feedback loops are too large, but they require fewer coefficients and storage space than FIR filters to achieve a similar frequency response. They do not have pre-ringing , but they can cause group delay distortion .

Although the lack of pre-ringing and the lower computing load speak in favor of IIR filters, the unwanted phase shifts in music playback are of course a serious problem. The linear phase of FIR filters is a strong argument for their use. Therefore, most filters in the HQPlayer are linear phase, unless otherwise stated.  The following articles are suitable for further in-depth study of filter design and for listening examples of pre-ringing:

• Which EQ Filter is Best For Mixing Bass and Low-End?
• Linear and minimum phase – Troll Audio (troll-audio.com)

Effects on space and transients and music genres

The group of  poly-sync-lp (linear phase)  filters improve the spatiality (space). They belong to the FIR filters and work in the time domain. Recommended for  classical  music and for music recorded “in the real world” (concert hall). A subtype of these are the AsymFIR filters, which are particularly suitable for  jazz/blues  .

The group of  poly-sync-mp (minimum phase)  filters improve the transients . They belong to the MinPhaseFIR filters and are particularly suitable for  rock/pop/electronics , as well as music recorded in the music studio. This also applies implicitly to  IIR filters .

Some filters also focus on the timbre , i.e. the sound colors.

Apodizing and non-apodizing

The purpose of the apodization filter is to replace the impulse response of the original decimation filter with another one.  This allows changing the time and frequency domain behavior of the original filter. A possibly more important point is to clean up the aliasing band at the highest frequencies. Depending on the ADC/mastering tools, there may be some aliasing band at the top of the frequency band. See also:

• Audio PC between ADC and DAC

Non-apodizing filters let the (possibly faulty) production-side digital filter properties through. Non-apodizing filters are good for HiRes recordings (88.2/96kHz and above) because the Nyquist rate is very high. Apodizing filters are most important for Reedbook/CD formats (44.1/48kHz) because of the low Nyquist rate. Here is a detailed explanation from Jussi Laako:

When using oversampled PCM, it doesn’t matter if it’s an R2R or Delta Sigma converter, you’ll get exactly the same ringing behavior with both because it comes from the digital filters and not the converter technology.

Ringing is the result of band limiting, and with RedBook content or other content sampled at a very low 44.1k or 48k rate, you need strict band limiting because you need to remove all frequencies above half the sample rate, otherwise they would be down- converted to the audio band. As a result, all A/D converters producing such content oversampled using digital decimation filters to reduce the sample rate to such low rates. By the way, all current A/D converters for music recording are delta-sigma converters. This results in ringing already built into your music content from the start.

Using R2R without oversampling at these low source content rates has a number of problems:

1) You have no proper reconstruction of the original analog signal at all, so you hear a heavily distorted signal that only slightly resembles the original.

2) Your transient level and timing are wrong because the music transient never coincides exactly with the sampling moment, but happens somewhere between samples. In the worst case, it is completely lost.

3) As a result of incorrect transient timing, you suffer from jitter in your waveforms.

4) Your output will have a large spread of frame frequencies into the MHz range due to lack of reconstruction, these will have intermodulation products that reflect each other back to the audio band.

5) You will have a strong high frequency rolloff starting at 1 kHz and being strongest between 10 kHz and 20 kHz.

Anything to avoid something that is already integrated into your source code anyway.

On such a NOS R2R sine wave, the 19 kHz sine wave looks like this:

You can also see that the signal is very unstable as the waveform is filled with blue areas where the signal has occasionally touched.

Now the amount of ringing depends on the filter length (IOW, number of taps). The longer the filter (more taps), the longer it rings. So, for example, Chord’s 1 million tap linear phase filter at a rate of 705.6k means it has almost a second of pre-ringing and the same amount of post-ringing (about 500,000 samples on either side; 705.6 / 500 = 0.7086 seconds ).

Because I’m interested in both  time and frequency domain behavior  , I’ve put a lot of effort into designing filters that are as good as possible in both domains simultaneously. IOW that gives you a correct reconstruction from all aspects that are inversely proportional to each other, through  a 1/x relationship . This means making a filter that is as short as possible in the time domain (as short ringing as possible) while giving as good a frequency domain reconstruction as possible. I give several filter length options for most filters so that you can choose the optimal balance for your enclosure, from short to very long filters, depending on the requirements of the source content. If you want minimal ringing and optimal time domain response, use the “-short” filters.

There are also completely ring-free interpolators (polynomials) that you can use for both PCM and SDM outputs, but these still have many of the adverse effects of NOS listed above due to incomplete reconstruction.

Coming back to the ringing that is baked into the source content, this is an aspect that can be changed with apodizing filters.Source:Holo Cyan 2 – Page 6 – DAC – Digital-to-Analog Conversion – Audiophile Style (audiophilestyle.com)

Upsampling ratio

“Integer” means that it can only perform even ratio conversions up or down. For example, a 44.1 kHz source cannot be upsampled to 196 kHz by filters with the ” Integer ” property. The correct way to do this would be to upsample it to 176.4 kHz. “Integer up ” means that the filter can only perform conversions to higher rates, but not to a lower rate (eg from 384k to 192k). ” Any ” has no restrictions.

The even ratio conversion is recommended as it generally reduces the computational load.

Compressed music

The problem today is over-compressed music ( loudness war ). This compression of the audio signal leads to a more consistent “audibility” of the music, but also results in a high loss of dynamics. As a result, the upper bands are cut off. With such recordings it is better to convert them to a DSD output with a higher rate. At least DSD256 or even better DSD512, using the EC modulators. Even PCM upsampling to 1.5MHz cannot compete with this, as it sometimes lacks “depth and airiness”.

If it’s a 44.1/48kHz source, using an apodization filter is generally recommended, see above. It’s also good to stick to a shorter filter, such as poly-sinc-short. And possibly minimum phase.

Another approach is to use longer filters, but particularly transient-optimized, such as the poly-sinc Gaussian family:

1x (44.1/48kHz) = poly-sinc-gauss-long
Nx (88.2/96kHz and higher) = poly-sinc-gauss-hires-lp

The developer Jussi Laako is so enthusiastic about this solution that these filters will be pre-defined in the settings in the future.

Table with filter properties

The table lists the most important filters without guarantee. From HQPlayer version 5.20 (September 30, 2023) the new filters will beNEWThe descriptions were taken from the HQPlayer manual and other sources. Links lead to further explanations and measurement results.

HQPlayer Filter Selection – Descriptions without guarantee –

Filters with *-2s have two-stage oversampling. The first stage rate conversion is performed by at least a factor of 8 using the selected algorithm. And further converted to the final rate using an algorithm optimized for converting content that has already been processed to at least 8x rate. This lowers the overall CPU load while maintaining conversion quality. Especially useful for highest output rates.

to fis Audio PC

to the modulators