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Miska

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  1. Miska

    HQ Player

    Using DSD is not really related to this much, it is more related to other aspects. Only notable time domain aspect of DSD is when the source content is also DSD, because then you don't really have much frequency domain band limiting which means that you avoid most of time domain effects of that frequency domain band limiting. Longer impulse response is result of more taps, which gives you a have sharper cut-off in frequency domain. So yes time domain response suffers from length of "ringing" or "settling time" (which MQA calls "blur") perspective. It is same with analog filters too, there you just talk about order or Q of the filter instead of length/taps. However, you cannot make that aspect of time domain performance better by making the filter shorter (less taps) forever, because at some point it begins to roll-off frequency response early which in turn slows down transients and thus making time domain response worse again (MQA being one prime example of such). Overdoing this "settling time" aspect also leads to more images/aliasing - less accurate reconstruction. If we consider time domain performance as whole and talk about "transient response" or "step response" we note that it has two aspects. Settling time (amount of ringing, related to Gibbs phenomenon) Steepness/slope of the rise Location/timing and shape of transient If you try to improve (1) too much, then (2,3) suffers, or vice versa. Bound of (2) is ultimately defined by the sampling rate - fs/2 (Nyquist) frequency. We need to remember from the Gibbs/Fourier series that steepness of the step is defined by number of harmonics included in the series and accuracy of their relative levels. Lowest sampling rate involved defines how many harmonics can be included, but filter responses (both digital and analog) define correctness of the relative levels and possible limitations to the steepness/timing. When MQA talks about time domain performance they talk about (1), while when Chord talks about time domain performance they talk about (2,3). And then they take things to the extremes. Also if you overdo (1), the reconstruction accuracy also suffers due to severe images/aliasing and you instead just get a lot of distortion (MQA being example). If you overdo (2,3) you have transients that never settle. "xtr" and "ext2" are still quite far from the "mega" filters. I've been putting a lot of effort in making filters that optimize all aspects, from all three time domain perspectives as well as frequency domain perspective, by using some nice math tricks. Then you can choose different weightings on these aspects with filter selection. But also all extreme approaches are supported (well, for DSD upsampling some "filter" choices like polynomial are not available at the moment). I also want to strongly emphasize, that in any case, for recorded content (instead generated test signals), you always have effect of the ADC filters included in the source data. Or if it was recorded in hires and converted to RedBook for distribution in software, that software converter's digital filter fingerprint. Using non-apodizing (like closed-form) or all-pass digital "filter" just means that the source's digital filter fingerprint is passed through. So even if your playback filter wouldn't "ring", the source data already contains "ringing" from the production phase. To deal with this aspect of source data, apodizing filters were created. In recorded material, you won't find test signal -like pulses or transients because they have already passed some filters. Also note that the filter "rings" only when it is limiting spectrum of the signal. If you have hires recording where sampling rate is high enough that no signal harmonics are reaching the beginning of filter's transition band (start of roll-off), the filter doesn't "ring". For example with MQA, it's filters begin to roll-off already around 30 kHz for 96 kHz sampling rate (48 kHz Nyquist), so their filters are much more likely to begin "ringing" than steeper filter with cut-off starting somewhere around 47 kHz. Because content is much more likely to have stronger harmonic at 30 kHz than at 47 kHz. In addition, it also means that slope/steepness of the rise is severely affected because of lower corner frequency (less harmonics included and their relative levels affected more).
  2. Miska

    HQ Player

    Yes, that's what it means. More taps/longer the filter is, longer it "rings".
  3. Miska

    HQ Player

    Yes, regardless of output format. These reasonings are based on differences in the source content.
  4. You shouldn't connect two active devices (sources) together, that will likely dramatically worsen performance of both. You could get a passive selector switch, there are such available (multiple input, single output). I have couple of such switches too. These are also inexpensive and generally don't degrade audio because they are only mechanical switch, wires and connectors in a metal box. But things should work quite well with just Chromecast Audio alone too, because most iOS applications also support output to Chromecast (Spotify, Tidal, etc).
  5. Only connect shield at the source end, but do not connect shield at both ends!
  6. Miska

    HQ Player

    ...fixed now for next release...
  7. Miska

    HQ Player

    Only modulator applies, and some of the settings from "DSD Sources" dialog. So the PCM-to-SDM filter restrictions shouldn't apply. I'll check this case that it is not unnecessarily applying the PCM filter limits.
  8. Miska

    HQ Player

    My theory (and ears) tell that filter choice tends to depend on source material. Technically, especially newer source material needs apodizing filters to clean up some of the mess created by half-band digital decimation filters used in ADCs and some production software. Other than that, subjectively I find minimum-phase filters sound good with things like older prog-rock (Pink Floyd etc) recordings and other such multi-track mix studio productions, also with some modern pop tracks. These don't contain any real acoustics at all, only little bit of artificial reverb. (curiously iPhone seems to use minimum phase filters for the headphone output) While I find linear phase subjectively good sounding on recordings made with few microphones in real acoustic spaces. Other than that, recently my preference has been poly-sinc-ext2 which is a linear-phase apodizing filter.
  9. Miska

    HQ Player

    Nice looking setup! I'm also planning to get a HPA4 for my headphone listening while working. OK, I see. Usually DACs give best performance at a certain input format and that could be used for everything. But the option to use "[source]" is there when needed/wanted. First rate conversion step from RedBook material is anyway the most critical. Couple of reasons for using DSD is that it allows higher rate conversion factors with digital filters, and many times replacing on-chip modulators with more direct path to the D/A conversion stage. But DSD128+ would be preferred for that. Yes, I think in your case the display gets updated before filter lists from the server arrive. I've hopefully already fixed this for the next release.
  10. You could try the HQPlayer OS image. It runs network interfaces bridged. Yes, instead of default "auto" in HQPlayer configuration you can set it to an interface name. It is just one of those options you need to manually edit in the configuration file, because it is something supposed to be configured by the hardware manufacturer.
  11. Miska

    HQ Player

    4.9.0 is latest at the moment... What kind of hardware are you using if it doesn't have built-in ethernet? If you are on Apple hardware, it usually has some peculiarities that may make some things not quite work. Many times SD cards do work though. I'm not sure if the Apple's Thunderbolt-to-Ethernet adapter would work, I have not looked into that, but it could be one possibility.
  12. Miska

    HQ Player

    Hmmh, it certainly is: Are you on the image? Default username is "hqplayer" and password is "password". You can then change it later. Either from the web interface, or alternatively from the command line with "hqplayer -s username password". You can look for instructions how to configure wpa_supplicant and enable it for systemd-networkd, but since it is a little bit of command line magic to do, I don't advertise it much... Wired is simple plug-and-play. There are many places with instructions, such as: https://wiki.archlinux.org/index.php/WPA_supplicant
  13. Miska

    HQ Player

    That was really ugly hack in HQPlayer 3 and with addition of input support it became too bad to maintain. And it doesn't work over to the Client anyway. So I took it out. Architecturally there really shouldn't be such link between driver/audio backend and GUI layer.
  14. Miska

    HQ Player

    Use of "[source]" is a bit unusual curiosity and generally not recommended. But it should show the active settings there. It has been showing to me during my testing, but it could have some timing issue with messages, I'll look into this.
  15. Miska

    HQ Player

    Maybe that's the text underneath changing size, so you likely get same result also with mouse. I need to play with size policy of vertical size of the text widget... All numbers should be equally tall, but looks like it is not the case... If you have input backend disabled (set to none), it just falls back to defaults. If you have some device selected, it should stick. Only practical effect of DoP for input is a little bit of extra CPU load when running with inputs because of the DoP detection code.
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