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StreamFidelity

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  1. This is a model from 2016, which was built until 2019. The iPhone 12 has been around since last year. Is the outdated technology the reason?
  2. Thanks for the clarification. The application Low Frequency Interpolation has to do something else, otherwise a music track would not be recalculated for hours.
  3. Dear @Nenon, I always enjoy reading your posts. I don't always agree with you, but I can understand your thoughts. Regardless of percentages, most will agree that the room is a very important component. I can understand that from my own experience. For example, if you measure with a non-calibrated microphone, you will correct the errors of the microphone afterwards. ๐Ÿ˜‚ But if you follow a few principles, you will get very good measurement results. Good software enables psychoacoustics and windowing to be taken into account. For example, a smooth freq
  4. That's a good point. In fact, I have to given one important piece of information: I also use the HQPlayer with low frequency interpolation files. Namely DSD256 with the Modulator ASDM7EC and an ingenious new filter poly-sinc-gauss-xla. ๐Ÿ‘ Now most of you will think that is nonsense. The thesis behind this is, if a file ripped from the CD in WAV format sounds better than FLAC, why shouldn't processing with more tabs also increase SQ? Just as with WAV it is no longer necessary to unpack and therefore possibly ensure a better SQ, the significant increase in the tab points c
  5. I did not manage to deposit the correct link. Presumably our forum has an aversion to audio โ€œscienceโ€ review. ๐Ÿ˜‚ I know KSTR from German-speaking forums. He writes well-founded. I am always grateful when I understand the technical construct. But at the same time humble, because not everything can always be explained with today's technology. My ears always win. The named sample jumps are wanted by the developer (it was written somewhere in the developer blog). Besides some measurement results, did KSTR "hear" the sampled music? There is nothing about that in t
  6. Tips for refining music files on the hard drive with resampling introduction to the topic Anyone who saves their music files on the hard drive has a fairly large playing field in front of them. For example, it's about the sampling rate (CD format 44.1kHz or HiRes from 88.2kHz) and also the bit depth (CD 16Bit or HD 20-32Bit). While the sampling rate expands the frequency response, the bit depth is about the dynamics. Sampling rates: PCM 44.1: 2 Hz -20 kHz PCM 48: 2 Hz -22 kHz PCM 96: 2 Hz -40 kHz PCM 192: 2 Hz -80 kHz PCM 384: 2 Hz -100 k
  7. Many Thanks. It actually made that clear. To work out the difference: I tested the combination Taiko Audio EPS (powered by 1 Keces P8 20V) in combination with 1 Keces P8 12V. Instead, you had 2 Keces P8 with 12V each for the CPU EPS and another with 19/20V. That could make all the difference. ๐Ÿ‘
  8. Ok i did it. ๐Ÿ˜‰ Of course, this is only possible with one Keces P8 (20V). It works. Unfortunately I cannot say whether the motherboard accepts the 4 pin CPU LPS but I suspect it will. In a direct, quick listening comparison, however, 2 Keces P8 are still better. The bass is more accentuated and the space opens up better.
  9. I just see that my Taiko circuit diagram looked different. EPS 2 was still called PCIe and EPS 1 was called CPU. So another use case would be the connection with Taiko ATX EPS 1 to 8 pin CPU EPS & EPS 2 to 4 pin CPU EPS?
  10. We may not be talking about the same things. I also use two Keces Single P8. But I only use the 8 pin CPU EPS. In my opinion, the question revolved around using the 4 pin CPU EPS as well. So ultimately to use the CPU EPS over 12 pins in total. That is only possible with an LPS. Sure it's possible. I use an 8 pin Molex cable on both sides for this. 4 pins remain free on the side of the motherboard. But I only did this for my experiment with 2 Keces P8. Otherwise, of course, I took the 8 pin on the motherboard. What is the point of only using the 4 pins?
  11. It would have been nice if it worked. I had done this experiment before and have now repeated it with Taiko Audio DC-ATX and 2 Keces P8. I use ROG MAXIMUS XI FORMULA. The nice thing about the Keces P8 is the display of the voltage and power consumption. This means that there can be no mistake about where the electricity comes from. 1. Use case: - Keces P8 20V to Taiko Audio DC-ATX - Keces P8 12V to 8 pin CPU EPS - Taiko Audio DC-ATX 12V to 4 pin CPU EPS No power was drawn from the Keces P8 12V! The audio PC booted successfully anyway. 2. Us
  12. Interesting! You made a lot of effort. With this setting you have at least reduced the latency. MHz: 2800 CL: 17 = Calculated nanoseconds: 12.14 ๐Ÿ˜‰
  13. I'm not sure if Apacer with ECC should be built into computers without ECC. They have disadvantages because of the high latencies: MHz: 2666 CL: 19 Calculated nanoseconds: 14.25 And they don't have a heat sink. There are good alternatives, for example G.Skill RipJaws V black DIMM kit 16GB, DDR4-3200, CL14-14-14-34 with 8.75ns. I consider low latencies to be one of the key factors behind high SQ. But everyone may see it differently.
  14. Interesting! I have problems with Euphony playing NAA with HQPlayer 4 Embedded 4.24.0. NAA T + A SDV 3100 HV USB are displayed correctly. Roon does not show any errors, but the progress bar does not change. And there is no music. ๐Ÿค”
  15. Me too! ๐Ÿ‘ In image processing, Gaussian filters are used to smooth or soften the image content. In fact, I get the impression that the music is smoother. Even so, the instrument separation and resolution is excellent. The bass is wonderfully accentuated. I like it best for 44.1/48 poly-sincgauss-xla: Apodizing extra long Gaussian polyphase sinc filter with extremely high attenuation. Optimal time frequency response. For SDM outputs, processing is two stages with 16x intermediate rate. The computing power is quite high for the DSD256 / ASDM7EC, but still in
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