If you never made a test and that is not confirmed by many other users, how can you say that it's a fact?
Neither you did measure lineatity, freq and transient response and other parameters of 7963 bcf (that would not tell the true story anyway).
So i repeat: you don't know what you're talking about.
Please stop it until you won't perform a serious test.
I don't need serious test, I know that Messerchmitt Bf 109 have no chance against МиГ-29..
Ok then, if you're so sure and full of faith there's no sense in trying to convince you to have a new and better experience. Stay with your opamps output stage, no problem for me, just don't disturb the rest of us.
For all the other "normally open minded" diyers, i think you have enough infos yet, but if you need any more just ask.
Opamp output stage is not mine, DRV603 which I use also..
I don't see anyone disturbed here..
There is nothing wrong in fact that you like sound of tubes more than opamps.. but fact is that AD797 and LT1363 which I used on Joro's board, and DRV603 wich I use now have few hundred times lower distorsion than any tube used as cathode follower..
Ok, that's what i was meaning.
Fortunately, many diyers do understand that distortion is not all in an hifi system, other parameters are much more important at this levels.
In my opinion, maintaining phase and impedance linearity of power supplies is one of the critical things to consider when using a SS design, especially on opamp stages in line level circuits.
Since tube circuits don't strictly require regulated power supplies and due to the nature of single rail operation, phase/Z mismatch of "each rail" doesn't show up with its negative effects.
When considering the phase and impedance linearity of dual rails (say +/- 15V) on SS power supplies, we have to take into account of two independent regulator stages, one for negative, one for positive.
jbau's experiments on sound quality versus phase and Z linearity over audio band is very helpful on this subject.
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Question: I would be interested in whether the difference between 44.1 kHz and 48 kHz / 96 kHz / 192 kHz in the audio recording can be heard.
Answer: No. A higher sampling rate than 44.1 kHz is meaningless since frequencies above 20 kHz can not be heard by anyone. The DIN standard has even set a limit of only 16 kHz.
Question: Why do manufacturers offer these formats?
Answer: Business. Marketing.
Question: What sampling frequency is necessary to accurately reproduce an Hi-Fi audio signal.
Answer: 44.1 kHz
Answer: No. A higher sampling rate than 44.1 kHz is meaningless since frequencies above 20 kHz can not be heard by anyone. The DIN standard has even set a limit of only 16 kHz.
Question: Why do manufacturers offer these formats?
Answer: Business. Marketing.
Question: What sampling frequency is necessary to accurately reproduce an Hi-Fi audio signal.
Answer: 44.1 kHz
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I think you have a bit of confusion about what a sampling frequency is and how sampling works...
Why should it be directly related to what audio frequencies humans can hear? Analog to digital conversion is not that simple...
Do you think that if i have to reproduce a bass instrument of - say - 200 hz frequency, i only have to sample it at a 200 hz rate and it will be correctly and fully reproduced?
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😀
Sampling rate is all about informations density. Low (digital) sampling rate = low adherence to the sampled (analog) waveform of ANY frequency
Please explain. Are you saying that you don't believe in the Nyquist-Shannon sampling theorem?
Yes. I assume we all agree to the validity of the (well-proven) Nyquist–Shannon sampling theorem that states that we need a sample rate at least twice the highest frequency signal we want to sample (if we want to avoid aliasing). The only disagreement should be about how much margin you need to cope with non-perfect filters - and I would like to see well-justified and supported views.