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HQplayer - do I need it?
- Thread starter MisterDee
- Start date
Got my DAC60 to display 11.2 MHz !!! Yay! thanks for the pointers. Per Jussi's recommendation I upgraded the kernel version on my NAAs to 5.15 and got everything sorted out. I also have a RBPi4 working with the Holo Red image !!!
Yes. It's nice and good sounding free solution. Worth of using it if you don't want to pay for more.
Just note: Both Maxim Anisiutkin's foobar2000 component and HQPlayer are using oversampling and delta sigma modulation. It is like in delta sigma DAC, just implemented by means of computer software.
Do all delta sigma DACs have the same quality? Do all sound the same?
In the same sense also software solutions differ.
HQPlayer will perform the conversion for you to whatever is optimal for the DAC in question.
It has lots of features and configuration options that will enhance your DAC performance. For Example, I have Holo Spring 3 running optimal configuration for an RDR ladder system and a DAC-60 with ESS chips that will need different modulators as well and an ifi NEO iDSD with Burr Brown chips that also will have a different setting for best listening.
It has lots of features and configuration options that will enhance your DAC performance. For Example, I have Holo Spring 3 running optimal configuration for an RDR ladder system and a DAC-60 with ESS chips that will need different modulators as well and an ifi NEO iDSD with Burr Brown chips that also will have a different setting for best listening.
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Are you sure? I mean each of those dacs will apply their own internal filter no matter the source, and that is what you will hear
You did not understand the point of HQPlayer. You are totally mistaken. You cannot hear "their own internal filter" if it i skipped - not in use.
With Holo (discrete implementation) and iFi (TI Burr-Brown chips) DACs, when you send DSD data to it, no internal filter is in action and the DAC device acts as pure D/A converter. No resampling, no other filtering, no modulator. DSD signal enters D/A stage directly. That's usually called direct DSD mode.
With ESS chip based DACs, one cannot skip delta sigma modulator, but it is possible to skip complete DAC oversampling/filtering by providing DSD data on input.
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Ok, maybe Iam mistaken.
But does not dsd conversion to analogue still require an output filter at the analogue stage? I am not familiar with how you could bypass this in the dsd stream itself, i didnt think it worked like that?
Analog filter is always required and is always part of D/A conversion in audio DACs. No matter what's DAC input (PCM or DSD at different rates) and if DAC digital filters were used or skipped. D/A conversion without analog filtering would produce unwanted ultrasonic content. The need of analog filter comes from the sampling principle/theory. Sampled digital data represents not only audio band of interest, but also infinite number of spectral replicas (mirror images) of base audio band at integer multiples of sample rate. Delta sigma modulator used as part of delta sigma DACs transforms those mirror images (which could cause intermodulation distortion) to high frequency noise, which is no more correlated with audio signal (it does not change with input signal changes).
So that high frequency content, be it mirror images (PCM DACs case like R2R) or modulator noise (delta sigma DACs), originates from the sampling principle as such and hence the role of an analog filter. That unwanted content cannot be removed by digital filters since they operate in range <0, fs/2>. So every audio DAC must perform analog filtering as part of D/A stage or behind it to restrict output to audio band of interest.
Analog and digital filtering in DACs have different roles. Digital filtering is used as integral part of oversampling process to ensure that mirror image of audio band will not appear in increased band <0, fs/2>. Analog filtering is used to filter unwanted higher frequency content.
Both digital processing part (oversampling and delta sigma modulation, be it performed by hardware or software) and D/A stage with analog filtering affect sound since they both introduce some level of distortion. So they both are reasons why different DACs measure and may sound differently.
@AudioJester
Confusing for you and others may be that delta sigma DACs offer selection of
- digital filter for PCM input
- analog filter for DSD input
With PCM input, the D/A stage input operates with input signal which is always oversampled to the same rate. Therefore there is no need for analog filter adjustment. But since oversampling is performed with PCM input, user may become an option to select oversampling digital filter.
DSD data is already oversampled and delta sigma modulated. Oversampling was already performed, when DSD content was created (either by A/D converter or by conversion from PCM content). With DSD input to suitable DACs like the above mentioned, complete oversampling in DAC device is skipped/substituted by DSD data fed. Therefore there is no place to adjust anything in digital domain. It was adjusted when DSD content was created. But since you can feed different rate DSD data to DAC (for example DSD64 to DSD512, what means 2.8MHz to 22.5MHz), analog filter requirements for different DSD rates may differ. Hence the reason for adjusting analog filter parameters with DSD input.
Confusing for you and others may be that delta sigma DACs offer selection of
- digital filter for PCM input
- analog filter for DSD input
With PCM input, the D/A stage input operates with input signal which is always oversampled to the same rate. Therefore there is no need for analog filter adjustment. But since oversampling is performed with PCM input, user may become an option to select oversampling digital filter.
DSD data is already oversampled and delta sigma modulated. Oversampling was already performed, when DSD content was created (either by A/D converter or by conversion from PCM content). With DSD input to suitable DACs like the above mentioned, complete oversampling in DAC device is skipped/substituted by DSD data fed. Therefore there is no place to adjust anything in digital domain. It was adjusted when DSD content was created. But since you can feed different rate DSD data to DAC (for example DSD64 to DSD512, what means 2.8MHz to 22.5MHz), analog filter requirements for different DSD rates may differ. Hence the reason for adjusting analog filter parameters with DSD input.
I bought and received Holo Red Streamer last week after about 4 weeks waiting time. Unfortunately, with the default RedOS image (OS Release 2024.01.24) DSD out over USB doesn't play well with Accuphase DAC60. I am having all the issues which @Moises helped me resolve with RBPi4 setup and Accupahse DAC60. I guess I have to wait until RedOS updates. Meanwhile using Holo Red with HQPlayer NAA OS 4.5.0 image.
With the default RedOS image -
I have tried both with the HQPlayer NAA service and plain UPNP service for streaming of DSD file.
- With UPNP stream of DSD file (minim server) Accuphase DAC60 sound output is garbled and defaults to 354KHz PCM (displayed on my Accupahse AMP w/ DAC60 card)
- With HQPlayer NAA (HQPlayer Embedded up sampling everything to DSD256) the sound is clean, but can only see 384KHz PCM (no DSD256 or 11.2 MHz DSD displayed)
With the default RedOS image -
I have tried both with the HQPlayer NAA service and plain UPNP service for streaming of DSD file.
- With UPNP stream of DSD file (minim server) Accuphase DAC60 sound output is garbled and defaults to 354KHz PCM (displayed on my Accupahse AMP w/ DAC60 card)
- With HQPlayer NAA (HQPlayer Embedded up sampling everything to DSD256) the sound is clean, but can only see 384KHz PCM (no DSD256 or 11.2 MHz DSD displayed)
Share a screenshot of your embedded settings. Can also ask Jussi on Roon HQplayer embedded forums. Already we have version 4.6.0 of NAA, can download and etcher the new image: Here
