hqplayer resampling filter setup guide for ordinary person

[hdwater]

now I use the trial versions of hqplayer, there is another 20 days for me to try this trial versions.

 

I notice there are different oversampling filter and noise-shaping selection. so it is a little bit diffcult for the normal person such as me to find the best combination.

 

I wondering if there are some easy understanding instructions for different combination of hqplayer resampling filter and noise-shaping selection even just in design pursose.

 

i am glad to find the software developer Misk in this forum, so I hope if it is possible for Miska to give a easy understaning introduction for the combination of these different filters. the pdf manual of the guide has too many technical words for me to get a good understand.

 

I know everybody has his own music device and own favorite. but can you give a general introduction of thes combination output design purpose.

 

or any suggestion for my system

 

my computer is windows7 x64 operation system, 8G memory, I7 processor.

 

My soundcard is LYNX L22.

 

minimax tube pre-amplifier, plinus sa-100 amplifier, sonus faber amati homage speaker.

 

I like the sound which hqplayer provide and i hope i can find the best combination, I try something and find just by my ears it is not so easy.

 

thanks and rgds

 

hdwater

[Miska]

I've been trying to avoid too much technical jargon in the manual, but balancing it is quite difficult. I'll try to explain things a bit further, but please ask if you would like to have some areas covered further.

 

First a bit explanation on time and frequency domain, please excuse me for some technical jargon. Frequency is signal change as function of time. Thus a signal has presentation in both frequency and time domains. "Linear phase filter" is a filter where all frequencies pass with same time delay. "Minimum phase filter" is a filter where all frequencies pass through as fast as possible, higher frequencies faster than lower ones. Longer/steeper filters change faster from passing frequencies to not passing frequencies as function of frequency. Shorter/gentler filters transition more slowly or "gently" from pass to stop as function of frequency. More accurately the filter wants to detect frequencies and transition pass/stop faster, longer time the filter has to "look" at the signal. This has side effect called "ringing" or rather "time blur". On the other hand, extremely short filter like a one that looks only at single moment cannot filter anything at all, because it sees only single point of time at once without any history or future (so it cannot detect any frequencies as those are a change over time). Linear phase filter takes equal amount of history and future into account during calculation. The problem in this is that it is kind of unnatural for something that is going to happen in future to affect already the present. Minimum phase filter on the other hand considers only from present to past, so it doesn't reflect things that are coming in future. This "ringing" is already in most RedBook recordings, since in most cases the ADC has gone through down-conversion and possibly another round at mastering from 24/96 or similar to RedBook. "Apodizing" filter is one that replaces or modifies this original ringing with it's own - that can be less than the original. All the filters explained below are more or less "apodizing" unless otherwise noted.

 

Why is "filtering" needed? Because otherwise upsampling/oversampling produces alias (distortion) components in frequencies above the original one. In down-conversion case it is even worse, because those components are produces below the original ones. D-A conversion also produces these components above half of the sampling rate frequency, and those are then removed by the analog reconstruction filters. Higher the sampling rate seen by the D-A conversion stage, simpler the following analog filter can be. Digital filters can easily outperform analog ones. Removing those spurious frequencies by filtering is called signal "reconstruction".

 

So if I go from left to right on the main window...

 

First is the filter selection, most of these can perform either up or down conversion, depending on what is needed.

- So "IIR" the first one is how a steep analog filter would sound like, I don't recommend using it for anything else than upsampling and only at 2x or 3x ratios, although it can do higher ratios or down-conversion too. I think this is mostly useful to hear how "extreme analog" would sound like. Some DAC chips have slightly similar output stages.

- Then there are three types of traditional "FIR", these are similar in construction to those ones used inside most DACs, the "asym" one being somewhere between linear and minimum phases, only taking "near future" into account. So a traditional design made as good as possible.

- "FFT" is a special kind in that it performs it's work in frequency domain and is also fairly steep. This is technically closer to how audio codecs work than how upsampling is traditionally done. I don't know if any hardware oversampling implementation would use similar technique.

- "poly-sinc-*" these are the ones I use most and recommend the most, these can perform conversion from most input rates to outputs rates in a single pass and with a very low CPU load. Single pass approach maximizes the filter precision. (those who will eventually ask, these are synchronous converters)

- "sinc" is a true asynchronous converter and can perform conversion practically from any rate to any other rate. Although it is quite high quality, it has fairly high CPU load too and not recommended unless the "poly-sinc-*" ones cannot do the needed conversion.

- "polynomial*" is not a filter as such, but just polynomial interpolation approach to upsampling. These look only at small number of samples to calculate a new one and thus don't "ring", but on the other hand the filtering performance is poor too. These kind of filters typically also cause premature treble roll-off (roughly 3 dB or so at 20 kHz for RedBook material, starting from ~10 kHz). These are the controversial upsampling "filters" some people like a lot while others don't like at all. (non-apodizing)

- "minringFIR" this is a single-pass filter that is very similar to the polynomial interpolators above in that it is really short and looks only at very brief period of time, while still performing better at filtering and not having such treble roll-off issues. Not recommended for other than 2x/4x/8x/etc ratios. (non-apodizing)

 

Then to the next item, dither and noise-shaping. This is needed whenever any processing is performed. Reason is that calculations can lead to results that have more precision than can be expressed in the resolution supported by the DAC. Just truncating or rounding the result to fit the DACs precision causes distortion that is directly related to the signal. Dither hides this rounding error into very low-level non-audible constant noise (a bit like thermal noise) - then it's no more related to the signal. Noise-shaping takes this further by moving this noise to less- or non-audible frequencies. Especially multi-bit converters but to some extent others too also benefit from noise-shaped upsampling in improved linearity.

 

I don't recommend any noise-shaper for 44.1/48 kHz output rates, because there is no proper frequency space available where to park the noise.

 

There are number of noise shapers:

- "NS1" is a first-order shaper, just tilting the noise floor so that it increases towards higher frequencies and it has a bit of extra "against-the-wall" high frequency noise too. Not really recommended for anything, but included for completeness sake.

- "NS4" is fourth order shaper that has a gentle step to move a bit of lower frequency noise to ultrasonic frequencies. The only shaper that I would say is useful at 88.2/96 kHz rates.

- "NS5" is fifth order shaper that has been designed to be used at 352.8/384 kHz output rates or above. This one moves aggressively roughly 40 dB worth of noise from low frequencies to ultrasonic range.

- "NS9" is ninth order shaper variant for use with 176.4/192 kHz, the step from low to higher noise is more clear, but otherwise similar to the "NS5"

- "RPDF" this is just plain white noise, not really recommended, but also included for completeness sake.

- "TPDF" is industry standard flat triangular dither, good for any case, especially for 44.1/48 playback cases. Doesn't generate practically any CPU load either.

- "Gauss1" is Gaussian noise dither, should be more "perfect" than TPDF, but also loads the CPU more. Works for all cases too.

 

Third selection is set of available output sampling rates, computed based on what the hardware and selected filter are capable of, in combination.

 

Generally, I recommend choosing between "poly-sinc-*" filters and using highest possible sampling rate. Dither or noise-shaper chosen based on above description, "NS9" for 192 kHz output, "NS5" for 384 kHz output and "TPDF" or "Gauss1" for any lower rates...

 

To be continued, I'll make two other posts. One for the DSD->PCM conversion and maybe other one for PCM->SDM (DSD) conversion.

 

Hope this helps...

[Miska]

I would put things roughly this way for filters:

- Minimum-phase filters for studio productions of non-classical music

- Linear-phase filters for classical and other music recorded in real acoustics with minimal miking

[mitchco]

Miska, fantastic explanation and recommendations! Thank you.

 

Cheers, Mitch

[Miska]

OK, then a bit about configuration of DSDIFF/DSF (DSD) content conversion to PCM... 2.8 MHz DSD64 content is converted to 176.4 kHz PCM and 5.6 MHz DSD128 content to 352.8 kHz PCM. These intermediate formates can be further converted to any other supported output rate using methods outlines in my earlier post.

 

In DSDIFF/DSF Settings -dialog there are two selections, conversion type and noise filter.

 

Conversion type, this defines how the delta-sigma modulation is converted to lower rate PCM format:

- "normal" is a traditional multi-stage way to do the conversion, this is very similar to how modern delta-sigma ADC chips do the conversion to PCM output

- "single-steep" is a single-pass brickwall-conversion, this has technically very accurate results.

- "single-short" is a single-pass more gentler conversion, introduces much less ringing than "steep". I'm using mostly this one when I need conversion.

- "poly-*" these are similar to the "poly-sinc-*" ones mentioned in my earlier post, performed in single pass, but requiring really powerful CPU...

 

Noise filter, this is used to reduce the ultrasonic high frequency noise of delta-sigma modulation:

- "standard" this is the SACD/Scarletbook standard one. Recommended for most cases.

- "low" this has lower transition corner than the standard one, useful for equipment that is sensitive to the ultrasonic noise of delta-sigma modulation. Good alternative for the standard one.

- "slow-*" linear- and minimum- phase gentle noise filters. A bit more aggressive noise filter than above.

- "fast-*" linear- and minimum- phase steep noise filters. The most aggressive noise filter of these.

 

Then there's the "6 dB gain" setting, since DSD is specified to have max -6 dB of the theoretical maximum level in use, the content may sound quiet compared to PCM after the PCM conversion. This setting enables 6 dB gain to match maximum specified DSD level to maximum possible PCM level. However, this should be used only for content that doesn't exceed the specified maximum. There seems to be also content out there that exceeds this maximum level and would thus result in overload. Whether this leads to limiting actually depends on the HQPlayer's volume setting...

 

In the same dialog there are two settings related to native DSD playback.

- When "DirectSDM" is enabled, the internal delta-sigma processing engine for DSD content is bypassed. The engine is now capable of all the same functionality as is offered for PCM, including volume control, delay and convolution engine for digital room correction. But since DSD is about "Direct Stream Digital" (or probably originally Direct Sigma Delta before marketing department jumped on technical jargon), a direct path setting is provided.

- "Direct playback type" is selection to output different kinds of DSD-over-PCM packing methods supported by some playback gear in order to play native DSD. These days, the equipment is almost always "DoP marker" compatible, unless there is a native ASIO DSD driver (in which case "Native / none" is enough).

[Miska]

Then to the last and short post so far..

 

Regarding SDM (1-bit delta-sigma, or DSD in other words) output modes. So the player can perform all kinds of processing for PCM -> PCM, DSD -> PCM, PCM -> DSD and obviously DSD -> DSD.

 

When SDM output mode is selected in the main window, filter selection changes to control "oversampling" modes, matching descriptions of the earlier post. And dither selection switches to control the choice of actual delta-sigma modulator. Output sampling rate selector just shows SDM rates in MHz range, instead of PCM rates in kHz range.

 

There is also possibility to perform conversions like 192 kHz PCM to 2.8 MHz DSD. Or 2.8 MHz DSD to 5.6 MHz DSD (or vice versa). When processing is performed from DSD input to DSD output, all processing is performed at native rate, not at any low-speed PCM. The DSD -> PCM conversion now applies only when PCM output mode is selected and file being played back is DSD...

 

Important note! When performing any processing to DSD output, do not try to push the volume to max, use something reasonable like -3 dB setting max. DSD works better when it's not pushed to the max.

[1audio]

Miska:

I find this very helpful as well. Can I bundle this into a PDF and add it to the one already in the Auraliti players and on our web site? Many users are lost with the available options and no understanding of why or which to select.

[Miska]

I find this very helpful as well. Can I bundle this into a PDF and add it to the one already in the Auraliti players and on our web site? Many users are lost with the available options and no understanding of why or which to select.

 

Demian, yes you can. I will also use this feedback as guidance to improve the manual in future.

[hdwater]

Dear Miska:

 

Thank you so much for such a detailed setup guide.

 

There are so many information that i am not know very well, but i would try my best to understand what you want to tell me.

 

I born in China and 10 years before I immigrated to new zealand. I'm keeping looking for the hqplayer setup guide in all the website especially chinese language in the past 10days(and it is the reason why i can find you at this forum), and I get some clue but compare to what you said, it is all not so accurate and correct if just in theory. so I wondering if you not mind I would like to put your explaination in some famous chinese language PC audio forum togewith my try best language translator. I hope more chinese pchifi can get the most detailed guide of how to setup the hqplayer from the software developer, as the HQplayer is one of the best music player software in earth till now as many pchifi player said.

 

before I do this job I would like to give my final understanding about how to setup the HQplay for the common use, which I means the 44.1khz/16bits or 48khz/16bits music file.

 

1. first we would like to check about our DAC, what the highest upsampling it can support and what is the highest resolution the dac can support. for example as the LYNX L22, Which can support 192KHz/24bit.

 

2. the second step is what the kind of music we would play, roughly we say three kind of music types which are classic music, pop/rock/electronic and jazz/blues. and each music type we can say to sub-types, one is live which is recording in the music hall and the other is recording in the audio recording room.

 

3. the third step is check the capacity of the PC hardware, if it is a quick CPU with big memory such as I7, I5, fore cors processor, more than 4G memory, or these low frequency cpu such as intel pentium or similar.

 

4a. this step I am not very clear that is whether in theory the higher oversampling would give a more smooth music playback or it is depend, I means if the dac can accept the 384khz oversampling signal we would try 384khz instead of 192khz, if the dac can accept the 32bit we try 32bits instead of 24bits or 16bits.

4b. I have another not very clear, such as for my soundcard, it is LYNX l22, can support 192khz/24bit signal, but if in “DAC BITS” selection, I select 32 Bits, the soundcard still can play music, do the lynx l22 down the 32bits signal which up-bits by software of hqplayer down-bits to 24bits which lynx l22 can accept, or some other process between the hqplayer software with the lynx L22 DAC chips. And for LYNX L22, in the “DAC bits” selection, which is the best in theory, the highest 32bits which LYNX L22 can not accept but still can play or the 24bits, the highest bits the LYNX L22 can acknowledge. We just talk about in theory not the real world, since the real world is too hard to talk about.

4c. So in the DAC bits selection, there are three choice, the first is we would select the Highest DAC bits such as 32bits, the second is the highest DAC bits the soundcard can play, such as for LYNX L22, is the 32 bits even the lynx l22 in theory can just accept the 24 bits, or the 24bits, what the lynx l22 can acknowledge in theory.

5. After the step 4, we would go to the set up of buffer time, in this part, is there any theory about the butter time length, does it means in theory the more short buffer time would means more quality sound or it is depend on humans ear or something else. If the answer is “yes”, then in this part, we just try the shortest buffer time the computer hardware can accept, otherwise, we just believe our personal ears and feeling to make the final selection or just the default setting?

6. now we go to the filter selection.

6a. We would pay high attention in the poly-sinc series filter firstly since it is your most common use and recommand, I ever notice your own setup of the hqplayer is poly-sinc-short-mp and NS4 @ 192khz for the pop/rock, poly-sinc-short and NS4 @192Khz for the classic, however, in my memory, you talk about another setup in another thread which is NS9 instead of NS4.

I notice that you use filter Poly-sinc-short instead of Poly-sinc, does it mean that the poly-sinc-short in theory can provide the high quality sound than the poly-sinc or the ploy-sinc-short would need more cpu power, so the default would be poly-sinc instead of poly-sine-short, or this choice is just your personal like, I menas you like the sound from Poly-sinc-short more than ploy-sinc but you can not give a exact decribe of sound playback difference between the poly-sinc and poly-sinc-short.

6b. Now we talk about the filter from the top to the end, the first is “IIR”, I am not very understanding about “I don't recommend using it for anything else than upsampling and only at 2x or 3x ratios” , do you mean if the oringinal music file is CD format, which is 44.1khz/16bits, for those who would like or can only upsampling to 88.2khz or 128khz and expecially like the analog sounds, such as the sound from the LP, or the sound like the tube amplifier provided, a kind of fat, slow and dark sound then we would select the filter “IIR”, otherwise we would avoid such filter. (but as you know, many person expecially the traditionary hifi player perfer the analog sound, it is the reason why we can still buy the LP(vinyl album) and LP player and just few days before I take part in an hifi party, where they prefer the LP and play the LP)

6c. Once we talk about the “ FIR” filter, it is an digital fiter which used in most DAC’s in the market, the reason while these DAC can provide the good sound is just beacuse they upsampling the signal from the cd player and the upsampling procedure is made by the software in the DAC which packed in the DAC hardware. R right now, what the DAC hardware can do, the Hqplayer can do also, but the Hqplayer is done via software by the computer CPU. In the three FIR filter, the “FIR” suitable the classical music especially recorden in the real world such as the concert hall and the” AsymFir” is good for Jazz/Blues which recorded in the real world, while the Minphase FIR is good for Pop/rock/electronic, since most of these music is made in record studio room or just made in the computer.

From some other website, i found some pchifi player said that they prefer the “FIR Gauss1 @ 176.4khz” for the classic and “AsymFIR Gauss1 @ 176.4khz” for the jazz, since most of this kind of music is recored in the real world.

7. “FFT” is a special kind of filter and it is so special that even i know every words in your reply but i still have not get a good understanding of what the “FFT” was and what kind of music most suitable for “TTY” or what kind of sound the “FFT” would provide. It looks like a new technology or a very special way since there are just few or no DAC use such kind of way. It looks that the “FFT” belong to another kind of filter way.

8. the poly-sinc sires filter, which is what you prefer and according your own setup, it looks that you like the poly-sinc-short and poly-sinc-short-mp more than the poly-sinc and poly-sinc-mp. Although i do not know why, if possible, please give me some explaination, i read the guide and your reply, but it looks i still not find the right key between the poly-sinc and poly-sinc-short. From the software guide, in the poly-sinc-short-mp, it said the most optimal transient reproduction, if it is a optimal transient, why you recommand the poly-sinc instead of poly-sinc-short. What i guess is the ploy-sinc need less cpu power and is more suitable for common computer, is it right??!!

9. now we go to the filter “sinc”, I get the information from a 40 years hifi experience music fans, he said if setup as “sinc TPDF @ 176400” he can enjoy the most beautiful human sound, which is very similar as the hi-end CD player with high end DAC. The advantage of Sinc is sound quality like the “FIR” which can provide the balance and good sound and can suitable for any rate of upsampling such as even 44.1khz to 76.8 khz or 44.1khz to 118khz for example. But since the sinc would heavy occupy the CPU, so you not recommend the SINC filter unless the Poly-sinc can not do the filter job, then we take account of “SINC” such as we meet an very strang upsampling ratio such as 2.3x, not a normal 2x,3x or 4x, then the SINC would think about.

10. for the last two, since you do not recommand to use, so just forget these two filters.

After filter, we go to the dither and noise-shaping.

1. none, for these dac hardware can support 32bits signal, we would select none, otherwise we need some noise-shaping process. My question is for these person who do not do the upsampling and even not change the DAC bits, since they trust do nothing is better than do something, at that suitation, the none filter and none dither is suitable to them, is that right?

2. NS1 as an example of my LYNX L22, if i oversampling to 176.4khz/192khz, then I can select the NS1 as the noise-shaping way, but you do not think it is a good way, so you not recommend this way.

3. for the upsampling is 88.2/96khz, you prefer NS4 as the noise-shaping way,

4. NS5 is for future use, it is not suitable for my LYNX L22 sound card, since its dac maximum just support 200khz/24bits. So only these hardware can support high than 352.8/384khz.

5. NS9 is suitable for 176.4/192khz, it would be the best suit for my LYNX L22 if I upsamling to 176.4/192khz. And it is the reason why you preference setup is “poly-sinc-short-(mp), NS9 @ 192khz.

6. RPDF looks do nothing except the white noise, so just forget it

7. TPDF is a easy job even for the low power CPU and it is the industry way to shape the noise and since the most common industry standard are 44.1khz for cd and 48khz for computer, so it is only useful for these who do not do any upsampling, such as oringinal cd format 44.1khz and do not do any upsampling such as 88.2khz/96khz/176.4khz/…., then we can select the TPDF as the dither noise-shape.

8. “Gauss 1” is a good way for noise control but a heavy cpu occupied, so only suitable for these with the best hardware computer. I know why some audiophile prefer the classic sound via setup “FIR Gauss1 @ 176.44khz” or human voice at “ sinc Gauss1 @ 176.4khz” , for these strong computer, these hard cpu job filter and noise-shaping can provide the good quality sound.

My final question in this part is in theory for these files with 44.1khz/16bits, which is the best upsampling rate, the 176.4khz or 192khz for my LYNX L22, since somebody said the 176.4 is eaual 44.1x4, so it is the better upsampling ratio than the 192khz, while somebody said the higher the better, so if you hardware can support the 192khz, then you select the 192, whaterver the original format is 44.1khz or 48 khz.

My conclusion is in theory, suas as an example of my LYNX L22 in an I7 Cpu, 8G memory, since LYNX L22 can support the 192khz/24bits. And the I7 has 4 cores with 8 G memory, So the best setup of the sqplayer for the normal format such as 44,1khz/16bits or 48khz/16bits files is

Buffer time ???===waiting for your answer

DAC bits ???===== waiting for your answer

176.4khz/192khz ???====waiting for your answer

Upsampling filter and dither are: poly-sinc-short-(mp) at NS9 according to the types of music, the real world recording use poly-shin-short, the audio studio select poly-sinc-short-mp.

For the human vocal voice, maybe we can try

“sinc at Gauss1”

And for these who do not believe the upsampling and insist use the orignial music file format, such as 44.1khz/48khz, we can select “TPDF” or “Gauss1” as the noise-shaping and for the filter we should select “none” since we not do any upsampling or we still need a selection between “poly-sinc-xx sires” “sinc” .

And finally for those prefer the analog sound, maybe we can try “iir” and the three “FIR” is not a bad choice for those traditional audiophile since they are the normal way in the most DAC.

Tomorrow I would like try the DSD file.

Any reply would be highly appreciated!

Fred ([email protected])

[hdwater]

Dear Miska

 

Since yesterday is too late and too hurry for the reply. so there are many spelling mistakes and somewhere is hard to be understood.

 

today i carefully read your reply and get a much clear understanding of how to setup the hqplayer.

 

but some questions is still comfuesed me and I re-mark it as follows:

1. the best Buffer time setup

2. the best DAC bits setup

3. the best upsampling rule, it is 176.4khz or 192khz to an original 44.1khz/16bits music file.

4. difference between poly-sinc and ploy-sinc-short in the sound quality playback in easy understanding words.

5. talk about the DSD files, when the "Directsdm" is selected, then the setup in the SDM default woudl be in function and if we ignore this selection, then the Pcm default setup would be in function, if right, what is your prefer choice in the "Directsdm".

6. for the 2.8mhz DSD 64 file, if I ignore the "Directdsm", then it would be downsampling to what my PCM setup, and in this suitation, what is the best re-sampling rate, the 176.4khz or 192khz if we talk about the LYNX L22.

7. in the Dsdiff/Df settings , when the convertion type is in working function, does it mean if the directsdm is selected, and my default setup of SDM sample rate is same as the oringianl DSD file, then there would be no convertion happen, otherwise, if the oringinal file is 2.8mhz and my sdm default setup bits rate is 11289600, then there would be convertion happen, if so, what is the best bits rate choice in the SDM default setup.

8. in my understanding, if we ingore the Directsdm playback, then the PCM default function would be in working status, and in my case, since my pcm setup is 176.4lhz or 192khz, so the convertion would happen. and in my case, it would be better to select the Directsdm and keep the default setup of sdm re-sample rate same as the DSD files. is it right?

 

i have translator some of your important point in chinese and if you like i can pass to you and you can use it in the chinese version setup guide if you think it is necessary.

 

hdwater

[Miska]

1. the best Buffer time setup

 

This depends on the computer and hardware used. 100 ms is good value to start with. It is mostly choice between responsiveness of the volume control (how fast changes are heard) and tolerance to CPU load spikes that cause gaps or stuttering. Low values give best volume control responsiveness, while large values give best safety against drop-outs.

 

2. the best DAC bits setup

 

It limits how many bits are output, using the selected dither or noise-shaping algorithm. It has two uses:

1) To tell the player what kind of bit depth the connected DAC has, mostly applies when unidirectional interface like S/PDIF or AES is used. So if for example 16-bit DAC is connected via S/PDIF, it should be set to 16.

2) When linearity measurements are available to tell the player about linear range of the DAC. So if DAC is linear for example to -110 dB, you can set it to 18 bits. This can be useful when noise-shaping is used.

 

With USB DACs and sound cards with on-board D/A-converters (1) is automatically detected and only (2) becomes useful.

 

3. the best upsampling rule, it is 176.4khz or 192khz to an original 44.1khz/16bits music file.

 

For use with poly-sinc filters, 192 kHz or what ever is the highest rate supported by the hardware.

 

4. difference between poly-sinc and ploy-sinc-short in the sound quality playback in easy understanding words.

 

"poly-sinc" measures a bit better, while "poly-sinc-short" probably sounds a bit better...

But I recommend just trying out and listening all those four variants. What sounds best may also vary depending on music material.

 

5. talk about the DSD files, when the "Directsdm" is selected, then the setup in the SDM default woudl be in function and if we ignore this selection, then the Pcm default setup would be in function, if right, what is your prefer choice in the "Directsdm".

 

"Direct SDM" is partially unrelated to the PCM and SDM default settings. If you don't need any processing (rate conversion up/down, volume control, multichannel speaker adjustments or convolution engine for DSD) select "Direct SDM" - it will just pass the data through from a file as-is.

 

I will return to the "Defaults" in a separate post.

 

6. for the 2.8mhz DSD 64 file, if I ignore the "Directdsm", then it would be downsampling to what my PCM setup, and in this suitation, what is the best re-sampling rate, the 176.4khz or 192khz if we talk about the LYNX L22.

 

Conversion is controlled by selection of "PCM" or "SDM" output mode in the main window and independent from the "Direct SDM" setting. "Direct SDM" will only disable any DSD -> DSD processing, it makes DSD playback chain "bit perfect".

 

If you play a DSDIFF or DSF file and "PCM" output mode is selected in the main window, conversion to PCM will be performed as configured in the DSDIFF/DSF Settings -dialog.

 

7. in the Dsdiff/Df settings , when the convertion type is in working function, does it mean if the directsdm is selected, and my default setup of SDM sample rate is same as the oringianl DSD file, then there would be no convertion happen, otherwise, if the oringinal file is 2.8mhz and my sdm default setup bits rate is 11289600, then there would be convertion happen, if so, what is the best bits rate choice in the SDM default setup.

 

If "Direct SDM" is unchecked (disabled), then 2.8224 MHz DSD can be upsampled for example to 11.2896 MHz (DSD256). If you have hardware that is capable of this output rate, I recommend using it.

 

8. in my understanding, if we ingore the Directsdm playback, then the PCM default function would be in working status, and in my case, since my pcm setup is 176.4lhz or 192khz, so the convertion would happen. and in my case, it would be better to select the Directsdm and keep the default setup of sdm re-sample rate same as the DSD files. is it right?

 

Whenever output is set to PCM, "Direct SDM" setting doesn't have any effect.

 

i have translator some of your important point in chinese and if you like i can pass to you and you can use it in the chinese version setup guide if you think it is necessary.

 

You can do it, but please try to ensure the information stays as correct as possible.

[Miska]

This is a post about "PCM Defaults" and "SDM Defaults" in the Settings-dialog.

 

In "PCM Defaults" it is possible to select the default filter, dither/noise-shaper, and sampling rate used when the application is started up or a new transport is selected. The sampling rate setting is not strict, other than it is used to limit the maximum auto-selected rate. In other cases the output rate is automatically selected using following logic:

- Source file sampling rate is read

- Selected filter is asked if it supports conversion to the chosen output sampling rate

- If the chosen output sampling rate is not supported for the source file's input rate using the filter, then next lower supported sampling rate is selected

 

In "SDM Defaults" it is possible to select the default oversampling filter used for the delta-sigma modulator, as well as select the default modulator algorithm and output sampling rate / limit with same logic as above.

[hdwater]

thank you Miska

 

i am almost know how to setup the HQPlayer.

 

 

regarding the chinese version translatation, if you think it would be helpful to you, I would like to translator the chinese back to english and then you would know what I talk about.

 

best rgds!

 

hdwater

[Miska]

hdwater,

 

To answer your earlier question, in case you want just "bit perfect" PCM without modification to the data in form of upsampling or such, select "none" as filter and "none" as dither and make sure volume is either turned up to 0 dBFS or disabled (by setting min and max to 0 in the Setting dialog). "none" dither selection shouldn't be used for any other case, only for this particular one.

 

regarding the chinese version translatation, if you think it would be helpful to you, I would like to translator the chinese back to english and then you would know what I talk about.

 

At least with Google Translate from Chinese to English I feel it usually ends up with too much "lost in translation" to be really useful for evaluation. So I don't think double translation is worth it.

[Miska]

Update on some recent changes regarding 2.9.1 and later:

 

There are now "poly-*-hb" filters for both up/down-sampling and DSD->PCM conversion. These are steep so called "half band" linear phase non-apodizing filters. These are in many upsampling cases completely reversible interpolation. Not really recommended for general purpose use, but interesting to play with.

 

The "poly-*" DSD->PCM converters (in DSDIFF/DSF Settings) have been heavily optimized and are now much faster while also having even more accuracy, so these are now actually one of the lowest CPU load while having best quality, most recommended choice now for the conversion.

[dallasjustice]

Thanks. Subscribed.

[dallasjustice]

I know I am about to be expelled from the NOS DAC club for what I'm about to say. :-)

 

I love the filters with my NOS DAC! I seriously think HQplayer is a perfect match with a true NOS dDAC because its like starting with a clean slate and selecting the parameters that work best for the specific recording. No one size fit all fillter. Can I eat at the grown-up's table now? :-)

 

Thanks Jussi!

[MarkS]

I agree that the manual is pretty technical and am looking for a simple way to upsample everything to double DSD. I've always been interested to try this player.

 

Has anyone compared this player to Audirvana/PM/Amarra?

[flatmap]

I'm also in the evaluation period and this thread is very helpful in getting started. Thank you.

[ted_b]

I know I am about to be expelled from the NOS DAC club for what I'm about to say. :-)

 

I love the filters with my NOS DAC! I seriously think HQplayer is a perfect match with a true NOS dDAC because its like starting with a clean slate and selecting the parameters that work best for the specific recording. No one size fit all fillter. Can I eat at the grown-up's table now? :-)

 

Thanks Jussi!

 

+1

[ted_b]

I'm also in the evaluation period and this thread is very helpful in getting started. Thank you.

 

I seriously think it's time for another set of screencast videos, this time HQPlayer. Problem is I don't know it like I did when I produced the JRiver ones. Anybody wanna help?

[sig8]

I seriously think it's time for another set of screencast videos, this time HQPlayer. Problem is I don't know it like I did when I produced the JRiver ones. Anybody wanna help?

 

Ted: That will really help me, and I am sure many more.

 

There are few people on HQ Player thread who seem to know more than me, who may be able to help. So you may want to post there as well.

 

Thanks.

[sig8]

Somebody mentioned pdf guide for HQ Player. Any url?

[wwaldmanfan]

Somebody mentioned pdf guide for HQ Player. Any url?

 

Google search: "Signalyst HQPlayer User Manual"

[jroyer]

I am new to HQPlayer. I like what I ear so far but to compare with my Audirvana+ setup on which I oversampler to DSD128 with no problem, I still need to figure out how to oversample to DSD128 with HQPlayer. Untill I crack the code on this one A+ sound better.

 

I am actually able to upsample to 5.6M with HQPlayer but something must be wrong in my setup because I eard noise or should I say 1 sec of sound then noise then 1 sec of sound etc...My DAC has no issue when I upsample to 5.6M using A+. Anyone has insights?