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1 hour ago, jdubs said:

I'll be using them for ultra-accurate RIAA correction.

 

So you'd want to just provide raw biquad coefficients? Tricky part is that those are sample rate specific. Which is of course not so much of an issue for RIAA since you can known the ADC sample rate in advance.

 

But I think the existing IIR EQ components could be useful for the same purpose without being sample rate specific.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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1 hour ago, Miska said:

 

So you'd want to just provide raw biquad coefficients? Tricky part is that those are sample rate specific. Which is of course not so much of an issue for RIAA since you can known the ADC sample rate in advance.

 

But I think the existing IIR EQ components could be useful for the same purpose without being sample rate specific.

 

hq support applelossless formats?

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1 hour ago, jimdukey said:

Is the RME ADI 2 fs DSD Direct Mode also for upsampling PCM to to High Rate DSD?

Not just for Original DSD Files?

Here the user manual https://www.rme-audio.de/downloads/adi2dac_e.pdf

page 32 chapter 17.2

”17.2 DSD Direct
To be able to digitally adjust the volume, DSD data must be converted to PCM. This is done automatically within the DA converter chips. In DSD Direct mode there is no PCM conversion – and consequently no volume control anymore. After having activated DSD Direct in the ADI-2 DAC’s menu (SETUP - Options), the analog signal is available only at the rear outputs, with a coarse volume control via the analog output reference level control. Outputs Pones and IEM deactivated.
In DSD Direct mode the output level for digital full scale is 3.5 dB lower than with standard DSD mode. Therefore the maximum analog output level is 3.5 dB lower than the chosen reference level. For a valid comparison between DSD and DSD Direct the volume of DSD should be set to -3.5 dB.”

… then no upsampling at all, just skip dsd to pcm conversion therefore no volume control 

 

Stefano

 

My audio system

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18 hours ago, Miska said:

 

So you'd want to just provide raw biquad coefficients? Tricky part is that those are sample rate specific. Which is of course not so much of an issue for RIAA since you can known the ADC sample rate in advance.

 

But I think the existing IIR EQ components could be useful for the same purpose without being sample rate specific.

 

 

Yes, I  want to use raw biquad coefficients.  You're correct re: them being sample rate specific but, yes, I know the ADC sample rate in advance.

 

I'm digitizing at 192khz which will remain for the forseeable future so no real issues for me on needing to switch biquads.  

 

The biquads provide a REALLY accurate way to do the riaa correction (significantly more accurate than what can be achieved with the existing IIR EQ components).  So, just really about going for the best result possible!  :)

 

-Jim

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42 minutes ago, jdubs said:

The biquads provide a REALLY accurate way to do the riaa correction (significantly more accurate than what can be achieved with the existing IIR EQ components).  So, just really about going for the best result possible!  :)

 

I'm curious why do you think so? Because the IIR EQ components calculate biquad coefficients for the rate in question and preserve full accuracy of calculated coefficients. There is always some potential for precision loss (not much though) when passing decimal numbers for biquads.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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4 hours ago, jimdukey said:

Is the RME ADI 2 fs DSD Direct Mode also for upsampling PCM to to High Rate DSD?

Not just for Original DSD Files?

 

You should always use ADI-2 in DSD Direct mode when used with HQPlayer upsampling. This also enables you to choose between two different DSD reconstruction filters in the DAC. For best performance, always send DSD256 from ASDM7EC modulator to the ADI-2.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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On 7/18/2021 at 2:39 PM, Miska said:

 

I'm curious why do you think so? Because the IIR EQ components calculate biquad coefficients for the rate in question and preserve full accuracy of calculated coefficients. There is always some potential for precision loss (not much though) when passing decimal numbers for biquads.

 

The deviation from the true RIAA curve of the IRR EQ methodology is significantly (orders of magnitude) higher than using biquads.  There are some threads on diyaudio that speak to it and this fellow Scott Wurcer published an article in Linear Audio which really details it.  Conceptually this whole thing goes back to comp.dsp days and Robert Orban's posts on the subject. 

 

A more recent discussion is here:

 

https://www.diyaudio.com/forums/pc-based/353387-phono-preamp-riaa-eq-using-iir-digital-filters.html

 

The dialog transgresses kind of quickly but a conclusion that can be drawn is that not using biquads can get you "good" results but its not the most accurate approach.

 

I think we should be striving for the best!!

 

-Jim

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1 hour ago, jdubs said:

The deviation from the true RIAA curve of the IRR EQ methodology is significantly (orders of magnitude) higher than using biquads.

 

How is that possible if IIR EQ is using biquads? Both are processed using same algorithms.

 

Quote

The dialog transgresses kind of quickly but a conclusion that can be drawn is that not using biquads can get you "good" results but its not the most accurate approach.

 

But HQPlayer's IIR EQ is using biquads... It is just collection of functions that design IIR biquad filters based on given parameters.

 

OTOH, RIAA EQ function used to do the vinyl cutting equalization is anyway based on analog electronics where accuracy is several orders of magnitude worse than anything digital comes up with... Even your player's speed will be off enough to have errors several orders of magnitude worse.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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Miska, I've set up my system such that the output is through a Gustard X16 via ASIO and the input is the analog line inputs of a Motu M4.  I see the input sample rate go from 192khz to 48khz and am not sure why.  This is an excerpt from the log:

 

  2021/07/20 18:45:36 Initialization complete, starting audio engine
  2021/07/20 18:45:36 Teams: 1
  2021/07/20 18:45:36 Places: 1
  2021/07/20 18:45:36 Parallel threads: 12
  2021/07/20 18:45:36 Nested parallelism: 4
  2021/07/20 18:45:36 Parallel pipelines: 4
  2021/07/20 18:45:37 ASIO output channels: 0 in / 2 out
  2021/07/20 18:45:37 ASIO output channel map:
  2021/07/20 18:45:37     0: Analogue 1
  2021/07/20 18:45:37     1: Analogue 2
  2021/07/20 18:45:37 ASIO output buffer sizes: 4096/262144/1048576 granularity: -1
  2021/07/20 18:45:37 ASIO output using ASIO default buffer size
  2021/07/20 18:45:37 ASIO output using ASIO buffer size: 262144
  2021/07/20 18:45:37 ASIO output latencies: 286720/331776
  2021/07/20 18:45:37 ASIO output not using ASIO output ready notifications
  2021/07/20 18:45:37 ASIO output engine started at 24576 kHz, 2 channels, 262144 sample buffer (2 channels)
+ 2021/07/20 18:45:37 ASIO output engine running at: 24576000
+ 2021/07/20 18:45:37 WASAPI input engine running...
  2021/07/20 18:45:37 WASAPI input period time set
  2021/07/20 18:45:37 WASAPI input open audio endpoint GUID: {0.0.1.00000000}.{5945ba1a-e30b-4a48-992a-e93d090574ad}
  2021/07/20 18:45:37 WASAPI input currently using: 'In 3-4 (MOTU M Series)'
  2021/07/20 18:45:37 WASAPI input type: Default
  2021/07/20 18:45:37 WASAPI input engine initialized
  2021/07/20 18:45:37 WASAPI input device period (default/min, ms): 10/3
  2021/07/20 18:45:37 WASAPI input using device default WASAPI period size
  2021/07/20 18:45:37 WASAPI input trying to use 10 ms for WASAPI period size.
  2021/07/20 18:45:37 WASAPI input initialize audio device using 96000/24 (32), 2 channels
# 2021/07/20 18:45:37 WASAPI input failed, trying another format (if available)
? 2021/07/20 18:45:37 WASAPI input no formats available - trying shared mode
  2021/07/20 18:45:37 WASAPI input mix format: 48000/32/2
  2021/07/20 18:45:37 WASAPI input sampling rate: 48000 (48000)
  2021/07/20 18:45:37 WASAPI input buffer size 1056
  2021/07/20 18:45:37 WASAPI input engine started at 96 kHz / 24 bits / 2 channels, 960 frames buffer (2/2 channels)
  2021/07/20 18:45:37 WASAPI input engine starting at: 48000
  2021/07/20 18:45:38 Rate or blocksize change triggered
  2021/07/20 18:45:38 Rate: 48000, block size: 5120, frame size: 640
  2021/07/20 18:45:38 Block size: 5120 (sample: 4)
  2021/07/20 18:45:38 Oversampling: short min phase poly (light)
  2021/07/20 18:45:38 Modulator: adaptive fifth order 1-bit
  2021/07/20 18:45:38 Integrator: IIR
  2021/07/20 18:45:38 Playback engine ratio: 512
  2021/07/20 18:45:38 Set volume: -35 +
  2021/07/20 18:45:38 Convolution engine: overlap-add
  2021/07/20 18:45:38 Convolution gain compensation: 0

 

Any thoughts?

 

Thanks!

Jim 

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11 hours ago, jdubs said:

Miska, I've set up my system such that the output is through a Gustard X16 via ASIO and the input is the analog line inputs of a Motu M4.  I see the input sample rate go from 192khz to 48khz and am not sure why.  This is an excerpt from the log:

 

  2021/07/20 18:45:37 WASAPI input currently using: 'In 3-4 (MOTU M Series)'

...

  2021/07/20 18:45:37 WASAPI input initialize audio device using 96000/24 (32), 2 channels
# 2021/07/20 18:45:37 WASAPI input failed, trying another format (if available)
? 2021/07/20 18:45:37 WASAPI input no formats available - trying shared mode
  2021/07/20 18:45:37 WASAPI input mix format: 48000/32/2
  2021/07/20 18:45:37 WASAPI input sampling rate: 48000 (48000)

 

This is a virtual device presenting only part of the device. Thus you cannot change sampling rate or exclusively access only channels 3/4 and Windows thus refuses direct mode access and HQPlayer falls back to WASAPI shared mode and the format set for the device in control panel.

 

If there's an aggregate device presenting all channels of the device, you should use that instead.

 

Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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