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On 4/18/2017 at 11:46 AM, auricgoldfinger said:

I am interested in parametric equalization of my headphones.  Is there a plug-in or some other way to do parametric EQ in HQPlayer?

Sure! I use REW to create a set of filters, then export them as impulse response WAV files for both channels. These are then loaded directly into HQPlayer convolver. The filters are applied automatically when HQPlayer is playing, as long as you have the convolver enabled. Let me know if you need any specific step described in more detail.

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58 minutes ago, Bob Stern said:

 

I've never used REW before, but I figured out the following procedure.  I'd be grateful for your advice on simplifications or improvements: 

 

Prefs > View > clear “Enable mouse wheel zoom” if using a trackpad.

 

Open EQ window by clicking EQ button.

• Gear button:  Clear “Invert Filter Responses”.  Click Gear button again to close dialog.

 

Open "EQ Filters" window by clicking EQ Filters button (top center of EQ window).

• Clear “Always on top”.

• Control > Manual.

• Type > PK for parametric EQ.

• enter Q=1.4 for BW=1 octave.  (0.67 for BW=2.)  http://www.rane.com/note170.html

• close EQ Filters window.

 

Close EQ window, or else bring the main REW window to the front via cmd-tilde.

 

File menu > Export > "Export filters impulse response as WAV".

• Clear “Normalize”.  Mono.  32-bit.  Sample rate same as imported impulse response.

• No minus sign or other non-alphanumeric characters in filename.

 

Optional: Verify frequency response of exported IR WAV file:

• Open the WAV file:  File menu > Import impulse response.

• Click the “SPL & Phase” button above the graph.

• Hover mouse near lower right of graph to reveal 20–20,000 X-axis button and click it.

• Upper left/right zoom icons separately zoom Y-axis of amplitude & phase curves.

 

Hi Bob,

 

Looks correct to me. I do a few things differently, some are due to the fact that I'm trying to correct for measured room response, so I have a measured curve and a house curve I'm trying to achieve. Still, I mostly do manual filter selection to match the curves. For headphones, you'll have to experiment with the filter settings, as I'm not sure how you'd measure headphone output.

 

In the EQ window, I select 'Generic' as the equalizer type. This gives the most flexibility with shelf filters, LP, HP, etc., you are not limited to just PK.

 

When entering manual filter settings, don't forget to set the gain amount, in dB. I never set more than 6dB correction in either direction. And, as I mentioned, I use more than just PK filter type. High pass and low pass filters are extremely useful for adjusting a large portion of the curve before applying the more precise/limited PK filters. 

 

 

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47 minutes ago, Bob Stern said:

@Miska:  Convolution question:

 

The user guide says: "When source material sampling rate differs from the impulse sampling rate, impulse responses will be scaled to the source material's sampling rate. This can have a huge impact on CPU load ..."

 

It seems to me that HQP would resample the impulse response before it begins playing the audio, save it in memory or disk, and then begin playback by convolving the saved (resampled) impulse response with the music source.  This would cause a delay before playback begins, but it would not affect the CPU load during playback.  Am I missing something?

 

As far as I can tell, HQP resamples and convolves in near real-time. The only delay might be due to a small buffer, such as the one you can enable with some async audio drivers and with NAA. Other audio players such as Audiorvana+ have an option of decoding large chunks of files into memory, and then playing them from there. This does result in a potentially large delay before the start, but perhaps, a smoother playback. I don't see such an option in HQP.

 

I assume that convolution is an expensive operation from the CPU perspective. And if you are upsampling, then HQP must apply it to many more samples per second, resulting in a greater CPU load. But this is just my guess.

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14 hours ago, auricgoldfinger said:

I do have one quick question before I dive into the rabbit hole:  will my DSD files be played in native format or converted to WAV files?

 

The PC output will be sent to the DAC in the format that you chose in HQP, DSD or PCM. The filters are applied to the data before the output is generated.

 

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9 hours ago, acatala said:

Hello guys,

 

I would like to know if there are any known issues with HQPlayer (running on Linux and/or MacOS) with the DAC embedded in the preamplifier Rotel RC-1590. I am interested in playing PCM and DSD64/128.

 

Thanks!!

Don't know of any issues with MacOS and HQP. That's how I'm using it in my system: Macbook-Pro to USB to DAC. I feed my DAC with either PCM 384KHz or with DSD256 using DoP. My DAC is not a Rotel preamp, though.

 

My biggest problem was that with the Mac audio driver HQP doesn't provide a setting for buffering of data. This caused me problems since I was reading data from a NAS drive over Wi-Fi. The solution was to use HQP NAA on the same laptop as HQP (allowed me to turn on buffering) and also to put a WiFi bridge next to the system which improved wireless performance. 

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6 hours ago, mooflotic said:

Gents,

 

been a lurker for some time and now need a hint which I haven't found solutions on the forum.

I made a fresh install of HQPlayer on a 2011 i7 macmini (piloted by roonServer on the same machine) but I can't see the rates for DSD (in the settings panel) even if my Luxman DA-06 supports it. Am I doing something wrong? Is, as I think, HQP reading the capabilities of the DAC before presenting you the list of available rates?

Currently if I play DSF it output to 384kHz to DA-06. Switched the "Auto" thing (rightmost) on the frontend to no avail, it just seems to converto to DSD, resample to some 6MHz (???) then outputs to 384kHz (as limited in the PCM panel).

 

Any idea? I'm probably missing something very stupid but I'm hammering it since 2 weeks and no solution yet!

 

 

Go to DSDIFF/DSF settings and turn off Direct SDM. Try that.

Then, try changing Modulator to something different on the main screen and see if the rates show up (not all modulators support all rates).

Also, try different oversampling method if the other two settings don't produce a result.

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13 minutes ago, Jud said:

 

DSD256 is the max for native DSD.  I believe the questioner's DAC utilizes DoP, so you halve that due to protocol overhead and get a max of DSD128.

 

If you are talking about a specific DAC, then the highest rate can be limited by the DAC, no argument there. But the upper limit of CoreAudio on the Mac is DSD256, regardless of what the DAC supports.
 

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2 minutes ago, Jud said:

 

Interesting - I thought I was doing native.  Are you aware of any DAC at all that will do DSD512 with a Mac?

 

It was explained to me that DSD512 over DoP is impossible with current hardware.

 

DoP requires double the bandwidth of the native DSD rate, I assume due to protocol overhead.  So, DSD256 requires the DAC to support PCM 512, while DSD512 would require PCM 1024 support, which no DACs that I'm aware of can handle at the moment. I assume that this rate would also tax any modern consumer CPU when running HQP.

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Just now, Jud said:

 

I understand that.  But native rate should be possible with a driver.  Has no one ever written one?

 

I think CoreAudio is the problem. Nobody bothers to write another driver since that's a big investment in time and money, and you get CoreAudio for free.

 

I have two USB devices I can drive from USB output: an XMOS-based one and an Amanero. Neither one has a dedicated Mac driver, meanwhile both support DSD512 natively when using their provided PC driver.

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21 hours ago, pkane2001 said:

I have two USB devices I can drive from USB output: an XMOS-based one and an Amanero. Neither one has a dedicated Mac driver, meanwhile both support DSD512 natively when using their provided PC driver.

 

All this discussion of DSD512 got me thinking... always a dangerous thing.

 

So I installed Windows 10 on the Macbook pro (via bootcamp). Set up ASIO driver for the Amanero board, and voila! I no longer need DoP -- native SDM all the way, DSD 44.1x512 shown by HQP  and a nice D512 sign displayed on the Amanero DDC.

 

My DAC is playing music, but is obviously confused by the I2S signal (maybe the cable is miswired a bit), as the display shows DSD 384Mhz -- not quite sure what that means, but the sound is excellent!

 

The only other slight problem I have is that at DSD512 with convolution, my I7 4-core Macbook is using about 40-50% of all the cores consistently, which results in some noisy little fans interfering with the music.

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11 minutes ago, tboooe said:

Which USB cord are you referring to?  If you use an NAA you can put it right next to your DAC so the USB cord can be minimized.  You will of course need a way to connect the MAC to the network but maybe wireless will work?

The reference to USB was in regards to the other solution, namely, encasing my laptop in a sound-proof closet while keeping it connected to the DAC via USB :)

 

NAA is great but requires another computer device. What would you recommend as a good NAA device capable of handling DSD512? Wireless is not the best, as I've already found out playing DSD256, but I can do wired if necessary.

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19 minutes ago, Jud said:

 

The CuBox-i can cost you less than $100 with discount, and Miska makes an NAA image for it.  But that image is Linux, so it depends what DAC you have.  Mine will do DSD512 with Linux, but some won't.

 

Edit: I'm not certain whether the CuBox-i will run Win10.  Anyone?

 

I'll have SU-1 DDC connected to the NAA. Not sure if SU-1 has Linux drivers, but I'll check.

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1 hour ago, mirekti said:

Could someone try to explain why would one prefer XTR over closed-form filter, please?

 

I can't give you the theory, but I did prefer the closed-form filter for a few days. It seemed as if things sounded clearer, crisper. But, I ultimately decided to switch back to using XTR filter after playing some piano and orchestral pieces using closed-form: these sounded less natural to me, as if distorted. I would suggest that you try these filters on complex passages and see if you can A/B test them. 

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8 hours ago, Rasmussen said:

I am currently running Roon on a Mac mini (mid 2011) 2,7 Ghz intel Core i7, 8 GB and it often stumbles a bit when doing upsampling to DSD128. I was wondering whether using my iMac (late 2012) 3,2 GHz Intel Core i5, 32 GB RAM would make a difference? I would need to connect the latter with a firewire cable around 4,5 meters long and also use a thunderbolt to firewire adapter. But I hope that will work despite the length of the cable.....But do you think it is worth it - will the iMac do a better job with HQplayer?

 

I have an old Macbook Pro with a 2.8 GHz I7 and it can play DSD256 without stuttering. I don't think it has to do with more memory. Could be some settings you can adjust in HQPlayer (such as multi-core, different filters, etc.) Check CPU utilization while the files are playing. If it's not more than 50%-60% for all the cores, then it's probably not the CPU causing this.

 

Also, check your USB cable. I started by using some generic USB cable and it caused stuttering and drop outs. Switch to a different, quality cable to see if it makes a difference, also try a shorter cable if your current one is long. Try a different USB port on your Mac, as sometimes ports stop working properly.

 

Consider where the music files are stored. A slow hard disk or a network drive could easily result in stuttering and drop-outs. Try creating a RAM drive and copy a file there and see if it plays without stuttering.

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  • 2 weeks later...
16 hours ago, mirekti said:

 

I do have REW, but no idea how to create filter for my needs.
Is there something simple like plugins for miniDSP? There I put the frequency, Q and gain, and that's it. 
In my case I would need to export it as a wav file I guess.

 

You'll need to have at least one measurement loaded into REW. Then go to the EQ screen and click on EQ Filters button to create filter settings manually. When done entering, you can export the filters from the main window by selecting File->Export->Export Filter Response as WAV. Here was a more detailed thread on how to do it:

 

https://www.computeraudiophile.com/forums/topic/19715-hq-player/?do=findComment&comment=654147

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2 hours ago, Miska said:

 

My i7-6950X can just do poly-sinc-xtr oversampling from 44.1k to 22.5792M (stereo), with about 90% load. So I'm waiting to get my hands on the 18-core i9 beast, to see what it can do. At the moment, there's no machine that would run poly-sinc-xtr from 44.1k to 24.576M. Or maybe couple of more channels from 44.1k to 22.5792M. Like poly-sinc-xtr from 44.1k to DSD256 with 8 channels maybe?

 

 

This confuses me. Does poly-sinc-xtr-mp-2s produce a much lower load than poly-sinc-xtr? Because with my I7 4790K, 16GB build I get an overall load of about 25% using poly-sinc-xtr-mp-2s upsampling 44.1k to 22.5M with convolution enabled. If I turn on CUDA offloading (780ti) the CPU load goes down to about 21%.

 

Is this unusual? I see everyone recommending much faster processors and faster GPUs. Maybe I'm doing something wrong?

 

pc_dsd512.thumb.jpg.f96f43b07da3a7f0238f34fc394f61cc.jpg

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Just now, bibo01 said:

The "2s" filter version is much less demanding in terms of CPU resources.

So there is nothing wrong. If you test the "non 2s" filter version, you will see that numbers go up considerably. 

Makes sense. I'll have to see what kind of load I get with the non-2s filter. Is the non-2s filter significantly better? I didn't notice a huge difference between the two.

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9 minutes ago, mfsoa said:

I'd like to try a Singxer SU-1 into my Gustard A20H, but I'm not sure if I can output 512 to it.

 

Almost certainly. You'll need the latest firmware for SU-1 (2.22) and a Holo Spring XMOS USB driver that supports native DSD up to DSD512. SU-1 will need to be connected to the DAC by a short I2S cable.

 

That's the configuration I'm using with X20Pro and SU-1 and DSD512 works great!

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