I suppose it keeps more or less a constant voltage. The result depends on sensitivity.Or do I need to worry about levels like what I read about with the DX3Pro?
I suppose it keeps more or less a constant voltage. The result depends on sensitivity.Or do I need to worry about levels like what I read about with the DX3Pro?
Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.For God's sake. Why would you want to do that ?
You take a perfectly good 44.1 signal, you alter it by needlessly resampling it to 768, just to have a nice "768 KHz" display in your E30 ?
That is not "Hi-Fi", sorry, that is something else.
You don't really believe that "just because it's 768 it will sound better", do you ?
If you had actual 768 KHz music I wouldn't say anything (except that your ears are limited to 20 KHz in your younger years, which is the reason why 44.1 exists BTW : 44.1 / 2 = 22.05 and 22.05 > 20). But AFAIK such music doesn't exist yet at this point. Or maybe limited so some rare files.
100% of people here and elsewhere on Earth have mostly 44.1 music. In my case I have 1 to 2% of 48/88.2/96/176.2/192 KHz music, but no more.
For 44.1 music, 44.1 is the way to go : bit-perfect output and 44.1 on your DAC. That is "High Fidelity".
Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
We have ears to evaluate.
Hahahaha now you've gone and done it , abx Gestapo arriving in 3.....2.......Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
We have ears to evaluate.
Hahahaha now you've gone and done it , abx Gestapo arriving in 3.....2.......
Hahahaha now you've gone and done it , abx Gestapo arriving in 3.....2.......
Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
We have ears to evaluate.
Funny how things only go one way, "smoother". While on a NOS DAC (a holo spring for example), setting it to oversampling or DSD mode of course makes it.. brighter. As opposed to the very smooth NOS mode! Almost like your very own bias is causing you to hear one way or another ...?Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
Hahahahaha.Oh dear. Please stop. At least pretend to science it up a bit. This is how nonsense propagates and becomes received wisdom.
Sorry but then you are speaking about science but don't understand how delta sigma DAC works and what does it mean direct DSD. There are technical reasons why software upsampling and software delta sigma modulation can easily beat in quality the hardware oversampling and delta sigma modulation in DAC chip. Generally, with delta sigma DAC chips, for PCM source content it is possible to bypass first stages of hardware oversampling and for DSD source content it is possible to bypass complete oversampling and delta sigma modulation (direct DSD mode). For delta sigma DACs the PDM (Pulse Density Modulation) signal (not PCM signal) is their native type of signal which is finally converted into analog by low pass filter. DSD signal is one bit two level PDM signal so it is native type of signal for Delta sigma DAC, which (in contradiction with PCM signal) does not need to be complicatedly processed before it enters the D/A conversion stage itself (no oversampling, no modulation).Oh dear. Please stop. At least pretend to science it up a bit. This is how nonsense propagates and becomes received wisdom.
You literally have no evidence suggesting that software oversampling improves performance. In real world testing, no matter what source sampling rate is performance stays roughly the same. Or in contrast, the best performance is at 44.1k and 48khz. Science is evidence based. We need evidence, not pseudoscience.Sorry but then you are speaking about science but don't understand how delta sigma DAC works and what does it mean direct DSD. There are technical reasons why software upsampling and software delta sigma modulation can easily beat in quality the hardware oversampling and delta sigma modulation in DAC chip. Generally, with delta sigma DAC chips, for PCM source content it is possible to bypass first stages of hardware oversampling and for DSD source content it is possible to bypass complete oversampling and delta sigma modulation (direct DSD mode). For delta sigma DACs the PDM (Pulse Density Modulation) signal (not PCM signal) is their native type of signal which is finally converted into analog by low pass filter. DSD signal is one bit two level PDM signal so it is native type of signal for Delta sigma DAC, which (in contradiction with PCM signal) does not need to be complicatedly processed before it enters the D/A conversion stage itself (no oversampling, no modulation).
You and your colleagues are repeatedly attacking people who carry about sound quality. You have to educate yourself before you write a nonsense and call it "scientific". I am watching this "science" forum a week and I am seeing that it is rather a "science" kinder garden.
Still no evidence. In measurements we've done, native DSD doesn't provide better performance. In contrast, the DSD bypass has constraints. For example, you can't have high output mode in ak4499 which potentially reduces DNR performance.An usual personal computer has much higher computational power than resource constrained DAC chip. It can perform upsampling and delta sigma modulation calculations in considerably higher quality in 64bit floating point precision than it is done in few dollars costing consumer DAC chips. Thus feeding the delta sigma DAC with 44.1k / 48k PCM signal, when it disposes with DSD direct mode, is the worst thing you can do IF you have the possibility to perform high quality software upsampling and software delta sigma modulation. It is about the possibility to substitute the majority of DSP which is performed in typical delta sigma DAC chip with software based solution.
High quality software based upsampling, dithering and delta sigma modulation can be performed in HQPlayer Desktop or Embedded in real time during playback, or PCM to DSD conversion can be performed offline in HQPlayer Pro and other professional tools like Saracon. There are also some free or lower cost alternatives (foo_dsd_asio for Foobar2000 or for example the solution of Jriver MC), which don't reach the quality level of HQPlayer's algorithms, but still can provide good result.
So it is no nonsense to upsample or convert PCM to DSD. And there is no base for a rule that it would be best to feed a DAC with 44.1k / 48k PCM signal.
Evidence? Measurements? Facts?So it is no nonsense to upsample or convert PCM to DSD. And there is no base for a rule that it would be best to feed a DAC with 44.1k / 48k PCM signal.
If your source is 16bit/44khz, upsampling can only, at best, end up with the same signal quality. You are adding no new information. Everytime a signal is converted there is entropy, information is lost. It is like making a copy of a copy.
Upsampling to max always sound smoother. And if you do it to DSD that even much smoother.
We have ears to evaluate.
For God's sake. Why would you want to do that ?
You take a perfectly good 44.1 signal, you alter it by needlessly resampling it to 768, just to have a nice "768 KHz" display in your E30 ?
That is not "Hi-Fi", sorry, that is something else.
You don't really believe that "just because it's 768 it will sound better", do you ?
If you had actual 768 KHz music I wouldn't say anything (except that your ears are limited to 20 KHz in your younger years, which is the reason why 44.1 exists BTW : 44.1 / 2 = 22.05 and 22.05 > 20). But AFAIK such music doesn't exist yet at this point. Or maybe limited so some rare files.
100% of people here and elsewhere on Earth have mostly 44.1 music. In my case I have 1 to 2% of 48/88.2/96/176.2/192 KHz music, but no more.
For 44.1 music, 44.1 is the way to go : bit-perfect output and 44.1 on your DAC. That is "High Fidelity".
I'm trying to layman-out the possibilities.