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How do you down sample DSD256 to DSD128?

 

My DAC can play DSD128 max. With the earlier version of HQP I could limit the bit rate to 5644800.

Now with the Auto feature it follows the source file settings in the bit rate drop down menu.

So with DSD256 material HQP defaults to 11.2 MHz and higher in the Transport window.

 

Regards.

 

Hmm. Depending on the filter used, there's some correlation between this and the auto setting. If the selection of the filter is DSD7 256+, the Auto output cranks to DSD256. If a lower order filter is used, like ADM7 (plain) the max auto setting should match that of the settings in the main settings panel. As to which one to use?? I would have thought Auto would be the limit on the DAC too.

 

That said if you set the playback to 564480 in the drop downs, it generally stays like this until HQ Player or the computer is restarted or the settings are "OK'd".


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Bingo! Thanks for the the help One and a half!

 

I was using the Closed Form filter ... which makes sense.

After changing to poly-sinc-shrt-mp the bit rate options opened up.

 

BTW the options you were refering to are Dither/Noise Shaping modulators.

Changing these does not help. Just clarifying for those who encounter the same problem.


Win10 Transport + Fidelizer 8 + JRMC 26 & HQPlayer | Job INT | Green Mountain Audio Eos HX

 

 

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Bingo! Thanks for the the help One and a half!

 

I was using the Closed Form filter ... which makes sense.

After changing to poly-sinc-shrt-mp the bit rate options opened up.

 

BTW the options you were refering to are Dither/Noise Shaping modulators.

Changing these does not help. Just clarifying for those who encounter the same problem.

 

Oh those :) yes, I get settings in HQ Player often the wrong way around, bitter experience teaches well. Glad it worked for you.

 

On the closed form filters.... never had much luck with them being stable, tended to lock up things, so I keep out of trouble and use the poly sinc family filters with very good results.


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On the closed form filters.... never had much luck with them being stable, tended to lock up things, so I keep out of trouble and use the poly sinc family filters with very good results.

 

Truth be told - I'm still trying to hear the differences over here.

I just tried it out since that was one of the new options available in the latest version

 

Are these the same kind of filters Schiit Audio has been hyping recently?


Win10 Transport + Fidelizer 8 + JRMC 26 & HQPlayer | Job INT | Green Mountain Audio Eos HX

 

 

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They have the same name?

 

From the Schiit Audio website ...

 

Closed-Form Digital Filter Preserves Original Samples

Most DACs simply use the stock digital filters embedded in their D/A converters. But even the most sophisticated ones, using their own digital filter algorithms, don’t have what Yggdrasil has—a time- and frequency-domain optimized digital filter with a true closed-form solution. This means it retains all the original samples, performing a true interpolation. This digital filter gives you the best of both NOS (all original samples retained) and upsampling (easier filtering of out-of-band noise) designs—and it is only available on Yggdrasil.

 

Have not found a description from Sygnalyst on the new filter.

Was just wondering if this is the same or similar thing.


Win10 Transport + Fidelizer 8 + JRMC 26 & HQPlayer | Job INT | Green Mountain Audio Eos HX

 

 

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Hi, just want to pass along that HQPlayer plus REW is an awesome combo. The room EQ file REW created fixed the bloated bass I was hearing and now the music is so much clearer, especially with vocals! It was a cheap and effective way to improve my system and so I thought i'd share my experience.

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How do you down sample DSD256 to DSD128?

 

You need to uncheck the "Direct SDM" setting in DSDIFF/DSF Settings dialog. This opens up all the DSP functionality for DSD sources too.

 

My DAC can play DSD128 max. With the earlier version of HQP I could limit the bit rate to 5644800.

Now with the Auto feature it follows the source file settings in the bit rate drop down menu.

So with DSD256 material HQP defaults to 11.2 MHz and higher in the Transport window.

 

If you have set default/limit to 5.6M in Settings dialog and have Auto in the main window, it shouldn't go beyond the limit.

 

However, if you have DirectSDM checked, the default/limit doesn't apply to DSD sources. If DAC allows the source file DSD rate, it is used and if not, playback is stopped.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Is there a keyboard shortcuts for, play, stop, forward and backward? Where can I find them?

 

Yes, there are two. Primarily the multimedia keyboard play, stop, forward, backward, etc buttons (also through IR remotes, etc). Secondary are double-mapping to Apple keyboard ones so:

F7 - previous

F8 - play/pause

F9 - next

F10 - mute

F11 - volume down

F12 - volume up


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Was just wondering if this is the same or similar thing.

 

Similar yes, certainly not exactly the same because at least my implementation uses more "taps" and probably has other implementation differences too (64-bit floating point, conversion to MHz rates, etc).

 

Which also means that it is not apodizing and thus doesn't reduce ringing from ADC/mastering decimation filters. So by definition no "de-blur" for closed-form.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Apologies if this has been addressed before, but I wasn't able to find a clear answer from my searches. I am trying to use Acourate filters with HQPlayer's convolution engine. Is the best approach to generate a 192khz filter, split the .wav into two Mono tracks (one for each of the left and right channels), and add the .wav files as the Left and Right IR files in the convolution screen? If I am playing only PCM and upsampling to PCM 192, will I need to select "Expand HF"? What if I upsample to SDM?

 

Also, when convolution is enabled, is it normal to have a 5-6 second delay when beginning playback (or switching tracks or moving to a different section of a track)? Note that there is no delay when the track changes at the end of track. During this 5-6 second period, the CPU load is at around 50%, and then drops back down to 6-7% during playback. I'm using Roon with HQPlayer, and my system is a NUC i5 with Windows 10. Will a faster computer alleviate or possible eliminate this delay?

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Apologies if this has been addressed before, but I wasn't able to find a clear answer from my searches. I am trying to use Acourate filters with HQPlayer's convolution engine. Is the best approach to generate a 192khz filter, split the .wav into two Mono tracks (one for each of the left and right channels), and add the .wav files as the Left and Right IR files in the convolution screen? If I am playing only PCM and upsampling to PCM 192, will I need to select "Expand HF"? What if I upsample to SDM?

 

Yes, it is best to produce filters with highest rate possible using the filter design tool. Acourate can do the equivalent of "Expand HF" when it is generating the filters, so you can create filters at high rate there and leave the "Expand HF" disabled at HQPlayer side. When possible it is better to get best possible filters generated in one go, instead of HQPlayer having to modify the filters later on.

 

Convolution is available also when you upsample to SDM, if source is PCM then the convolution filters are processed at source PCM rate before upsampling. If source is DSD and output is SDM, then convolution filters are processed at native source rate. If source is DSD and output is PCM, then convolution filters are processed at the intermediate 1/16th PCM rate. Convolution filters are adapted to the processing rate as necessary.

 

Also, when convolution is enabled, is it normal to have a 5-6 second delay when beginning playback (or switching tracks or moving to a different section of a track)? Note that there is no delay when the track changes at the end of track. During this 5-6 second period, the CPU load is at around 50%, and then drops back down to 6-7% during playback. I'm using Roon with HQPlayer, and my system is a NUC i5 with Windows 10. Will a faster computer alleviate or possible eliminate this delay?

 

Yes, it is normal to have initial start delay with all or most cores fully loaded - this is the initialization phase for the processing. The delay also happens when source sampling rate changes during playback.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Is there a way to get HQPlayer to output to 2 NAA's at the same time?

 

No, not at the moment. Since DAC has the master clock of the system, there can be only one master. Two NAA's would mean two DAC clocks.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Yes, it is best to produce filters with highest rate possible using the filter design tool. Acourate can do the equivalent of "Expand HF" when it is generating the filters, so you can create filters at high rate there and leave the "Expand HF" disabled at HQPlayer side. When possible it is better to get best possible filters generated in one go, instead of HQPlayer having to modify the filters later on.

 

Convolution is available also when you upsample to SDM, if source is PCM then the convolution filters are processed at source PCM rate before upsampling. If source is DSD and output is SDM, then convolution filters are processed at native source rate. If source is DSD and output is PCM, then convolution filters are processed at the intermediate 1/16th PCM rate. Convolution filters are adapted to the processing rate as necessary.

 

 

 

Yes, it is normal to have initial start delay with all or most cores fully loaded - this is the initialization phase for the processing. The delay also happens when source sampling rate changes during playback.

 

Thank you. I have noticed that using a 176khz filter alleviates the long initial start delay when loading 44.1khz files. However, when I use the 176.4khz filter, 96khz files and 192khz files eventually get distorted. I have tried having "Enable HF" both selected and not selected, and tried both convolution engine settings.

 

When I try using a 44.1khz filter with 192khz material, I get the same distorted sound. With the 176.4khz filter, the sound starts out fine, but will eventually distort from 20 seconds to several minutes into playback. 44.1/88.2/176.4 material is fine with the 176.4 filter. So far I haven't encountered any distortion with the 192khz filter with any files, but would like to avoid the initial start delay.

 

Perhaps a workaround would be to allow different filters for 44.1/88.2/176.4 material and 48/96/192 material?

 

My settings are PCM, poly-sinc, NS9 and Auto (max 192khz) but I have tried various PCM settings.

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Perhaps a workaround would be to allow different filters for 44.1/88.2/176.4 material and 48/96/192 material?

 

HQPlayer is not limited to any particular source sampling rates (file with 12345 Hz sampling rate would be just fine), so there would need to be huge number of alternative filters. So my recommendation is to use 192k filter and leave HF expansion off.

 

When I try using a 44.1khz filter with 192khz material, I get the same distorted sound. With the 176.4khz filter, the sound starts out fine, but will eventually distort from 20 seconds to several minutes into playback. 44.1/88.2/176.4 material is fine with the 176.4 filter. So far I haven't encountered any distortion with the 192khz filter with any files, but would like to avoid the initial start delay.

 

If you are using Windows, check Resource Monitor and on Mac check Activity Monitor to see how your per-core CPU load looks like.

 

How is your volume setting in HQPlayer and does the "Limited" count ever change from 0?


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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How is your volume setting in HQPlayer and does the "Limited" count ever change from 0?

 

Yes, as a convolver user it is very important to manage two things:

volume max at -3db

limit counter stays at zero.

 

This is a counter that shows when HQPlayer clips, which could very well be the "distortion" you describe (various amplifier topologies handle clipping in various ways, but none of them are good). Run your HQPlayer log and see what the convolver is reporting; if say +7db peaks, then set your convolver gain at -7.

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Just read about your efforts for GPU processing a couple of weeks ago as I try to catch up, Miska, kudos on this!

 

2016: the year I get myself a proper HQ Player upsampling + NAA rig!


Dedicated Line DSD/DXD | Audirvana+ | iFi iDSD Nano | SET Tube Amp | Totem Mites

Surround: VLC | M-Audio FastTrack Pro | Mac Opt | Panasonic SA-HE100 | Logitech Z623

DIY: SET Tube Amp | Low-Noise Linear Regulated Power Supply | USB, Power, Speaker Cables | Speaker Stands | Acoustic Panels

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HQPlayer is not limited to any particular source sampling rates (file with 12345 Hz sampling rate would be just fine), so there would need to be huge number of alternative filters. So my recommendation is to use 192k filter and leave HF expansion off.

 

 

 

If you are using Windows, check Resource Monitor and on Mac check Activity Monitor to see how your per-core CPU load looks like.

 

How is your volume setting in HQPlayer and does the "Limited" count ever change from 0?

 

Jussi,

 

Thanks, my volume min and volume max are both set to 0.0dB. I have confirmed that the "Limited" count does not change from zero when the sound changes.

 

I have just had the same issue happen, playing 44.1khz file with a 192khz filter, that was triggered by a remote login to the machine during playback. The resource monitor does not seem to show anything noteworthy when it happens. I will send a .log file to you via email after a few more trials.

 

ted_b: The sound is not just digital clipping. The sound suddenly turns into a scrambled mess until I stop playback. It's not momentary.

 

Edit: Just to provide some details: I am using HQPlayer with Roon, on Windows 10 on an i5 NUC. My DAC is a Devialet 120, using the WASAPI driver over USB.

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Edit: Just to provide some details: I am using HQPlayer with Roon, on Windows 10 on an i5 NUC. My DAC is a Devialet 120, using the WASAPI driver over USB.

 

For WASAPI it is good idea to change the "Buffer time" setting in HQPlayer to for example 100 ms. Practically all drivers default to 10 ms which is quite demanding when other stuff like Roon or remote desktop is running.


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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For WASAPI it is good idea to change the "Buffer time" setting in HQPlayer to for example 100 ms. Practically all drivers default to 10 ms which is quite demanding when other stuff like Roon or remote desktop is running.

 

Miska,

Do you know what the ExaSound (e22) driver defaults to?

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Can I ask a newbie question, out of curiosity? The filters used in HQPlayer, are they generic or are they customized for HQPlayer?

Furthermore, can these filters be licensed to be used in other software?

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Any idea what are those NAA*.img files under /images/ are for?

 

If you read the name of each file, they are images (entire OS + networkaudiodeamon) for BeagleBone, Cubox-i, RaspberryP1 and RaspberryP2.

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