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hdwater

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  1. 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
  2. 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
  3. 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])
  4. 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
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