Tks
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Just a few quick questions for a few things I can't get my head around. I know it's a lot, but tackle whatever you see fit if you can some time. It's just been bothering me not knowing, and not knowing how to seek proper answers to these things. I'm sorry if I am asking questions that I should better be served learning prerequisites for (like someone asking about the Big Bang's actual state, without never having taken a physics course for example).
First off, how does one precisely read linearity graphs? I recall in the old measurements there were the linearity graphs as they are now, but would also accompany by telling us how man "bits of resolution". Is there a way to extrapolate this data somehow by looking at the modern linearity graphs, and also.. bits of resolution? Brings me to my next question..
Does that have anything to do with bit depth of an audio file? Or is that the hard-limit of what a device is able to reproduce? Like if some older devices I seen measured a while ago for instance had 20-bits of resolution, does that mean the noise floor is effectively 120db (or something?). And if so, does that mean any device that doesn't have the capability of reproducing something like 16-bit.. you should never bother seeking out higher bit-depth content (not that I do, nor does anyone, but I am asking theoretically does it make sense for a device maker to bother with something like 24 or 32 bit support in-hardware.. aside from just marketing?). Also, does linearity have an overall effect on the final SINAD calculation directly or indirecty?
Another thing I am wondering, that I recently ran into was when playing around with DA filters on my DAC, and using the Non-oversampling filter due to seeing someone say in passing "it basically would produce the best impulse response". The problem being is supposedly hardware has to work harder(think MC from RME mentioned this, I'm not sure though) and the high frequencies begin to fall off (not too bad considering I can't hear 19K or higher). But what I did notice when a handful of songs when I had the volume cranked up for "fun listening".. is there was like some sort of wobbling ringing effect. No idea what that's all about.. Was wondering if anyone has anything about these sorts of topics for a laymen to grasp?
Lastly.. I always hear of DAC's doing their own thing with respect to sampling (like they do on-the-fly conversions and reconversions to reconstruct the signal or something because supposedly having a DAC process an "upsampled" 192kHz file is far easier for the hardware, than having it "natively" process something like 44.1 (which is why I assume no modern DACs support some 32kHz sample rate I see in settings page that asks me to check-off what rates my DAC supports). Also "upsampling" or "downsampling" ..How do I know it's even occurring? This only multiplies my confusion when we take Windows sampler into the equation. By this ordeal, would it make sense just to go into the driver page and set my DAC to 96kHz or 88.2kHz so whenever I'm not using ASIO or WASAPI (like when using Firefox, or playing a game since you don't have the option of NOT setting a sample rate) the DAC can just upsample to those "easier to process" sample rates for the heck of it and possibly less heat generation or whatever?
What's up with PC motherboard SPDIF Optical-output.. and what's up with "disable all effects" settings in the driver menu? (Ill elaborate now with this question)
When I had Equalizer APO running, and I had the Output of Foobar set to the default Windows Mixer optical out (with no ASIO or WASAPI or anything like that) I would instantly hear some insane distortion (sounds like IMD) compared to when I used ASIO or WASAPI push. I had to uninstall Equalizer APO and enable "Disable all Effects" in the windows driver menu.
So getting back to the whole sampling and SPDIF Optical thing.. I seen Roon has a sort of "chain tree" where you can see exactly what's being processed and how. Barring that, how would I know what is happening with the audio I am playing. What on Earth Windows Mixer is doing (this I hear resamples music to what the default sample rate on the DAC is set to through the driver panel in Windows). Is anything happening to my audio stream being fed from the Optical-out from my motherboard vs USB (I seen old posts online talking about how the audio is processed by the built-in "sound card" (or audio codec supplied with the motherboard for example). If so, what is Windows doing that wouldn't be done if I went the USB route vs Optical since it seems Windows Mixer will resample anything regardless that isn't being controlled by a company driver? Can this "mixer" be replaced with something like "PulseAudio" driver native to Linux (and whatever runs Mac) theoretically? Seeing as how this fellow demonstrates some differences, and interesting conclusions. Especially interesting how he goes into NOS DACs and describes what I propose is the same annoying ringing I heard as well with the NOS DA Filter.
Bit-perfect.. What is the point of this outside of studio use for whatever reason? What does bit imperfection even look like or sound like?
Windows/System volume VS AMP or DAC volume. (I hear this isn't even a question anymore in Windows 10 or something for some reason. but lets say for the sake of the argument we're in the Vista era), but yeah, what's worse? Lowering the volume on my AMP/DAC or Windows System (seeing as I can't run both at full blast). I would guess leaving the AMP or DAC at the volume where there is the least THD is the best route, and then lowering software volume as you see fit?
Seriously, what's the point of DSD? (PCM I can somewhat grasp my head around with respect to what finally goes on a CD and such). I've read two things about this ordeal, here is the first and the second one I can't find, but argues the opposite basically in some fashion I can't recollect (I know this question is almost ridiculous but don't need a true answer outside of just people's opinion based on understanding how each is handled).
Lastly, can you forgive the insanely long post :-} ?
First off, how does one precisely read linearity graphs? I recall in the old measurements there were the linearity graphs as they are now, but would also accompany by telling us how man "bits of resolution". Is there a way to extrapolate this data somehow by looking at the modern linearity graphs, and also.. bits of resolution? Brings me to my next question..
Does that have anything to do with bit depth of an audio file? Or is that the hard-limit of what a device is able to reproduce? Like if some older devices I seen measured a while ago for instance had 20-bits of resolution, does that mean the noise floor is effectively 120db (or something?). And if so, does that mean any device that doesn't have the capability of reproducing something like 16-bit.. you should never bother seeking out higher bit-depth content (not that I do, nor does anyone, but I am asking theoretically does it make sense for a device maker to bother with something like 24 or 32 bit support in-hardware.. aside from just marketing?). Also, does linearity have an overall effect on the final SINAD calculation directly or indirecty?
Another thing I am wondering, that I recently ran into was when playing around with DA filters on my DAC, and using the Non-oversampling filter due to seeing someone say in passing "it basically would produce the best impulse response". The problem being is supposedly hardware has to work harder(think MC from RME mentioned this, I'm not sure though) and the high frequencies begin to fall off (not too bad considering I can't hear 19K or higher). But what I did notice when a handful of songs when I had the volume cranked up for "fun listening".. is there was like some sort of wobbling ringing effect. No idea what that's all about.. Was wondering if anyone has anything about these sorts of topics for a laymen to grasp?
Lastly.. I always hear of DAC's doing their own thing with respect to sampling (like they do on-the-fly conversions and reconversions to reconstruct the signal or something because supposedly having a DAC process an "upsampled" 192kHz file is far easier for the hardware, than having it "natively" process something like 44.1 (which is why I assume no modern DACs support some 32kHz sample rate I see in settings page that asks me to check-off what rates my DAC supports). Also "upsampling" or "downsampling" ..How do I know it's even occurring? This only multiplies my confusion when we take Windows sampler into the equation. By this ordeal, would it make sense just to go into the driver page and set my DAC to 96kHz or 88.2kHz so whenever I'm not using ASIO or WASAPI (like when using Firefox, or playing a game since you don't have the option of NOT setting a sample rate) the DAC can just upsample to those "easier to process" sample rates for the heck of it and possibly less heat generation or whatever?
What's up with PC motherboard SPDIF Optical-output.. and what's up with "disable all effects" settings in the driver menu? (Ill elaborate now with this question)
When I had Equalizer APO running, and I had the Output of Foobar set to the default Windows Mixer optical out (with no ASIO or WASAPI or anything like that) I would instantly hear some insane distortion (sounds like IMD) compared to when I used ASIO or WASAPI push. I had to uninstall Equalizer APO and enable "Disable all Effects" in the windows driver menu.
So getting back to the whole sampling and SPDIF Optical thing.. I seen Roon has a sort of "chain tree" where you can see exactly what's being processed and how. Barring that, how would I know what is happening with the audio I am playing. What on Earth Windows Mixer is doing (this I hear resamples music to what the default sample rate on the DAC is set to through the driver panel in Windows). Is anything happening to my audio stream being fed from the Optical-out from my motherboard vs USB (I seen old posts online talking about how the audio is processed by the built-in "sound card" (or audio codec supplied with the motherboard for example). If so, what is Windows doing that wouldn't be done if I went the USB route vs Optical since it seems Windows Mixer will resample anything regardless that isn't being controlled by a company driver? Can this "mixer" be replaced with something like "PulseAudio" driver native to Linux (and whatever runs Mac) theoretically? Seeing as how this fellow demonstrates some differences, and interesting conclusions. Especially interesting how he goes into NOS DACs and describes what I propose is the same annoying ringing I heard as well with the NOS DA Filter.
Bit-perfect.. What is the point of this outside of studio use for whatever reason? What does bit imperfection even look like or sound like?
Windows/System volume VS AMP or DAC volume. (I hear this isn't even a question anymore in Windows 10 or something for some reason. but lets say for the sake of the argument we're in the Vista era), but yeah, what's worse? Lowering the volume on my AMP/DAC or Windows System (seeing as I can't run both at full blast). I would guess leaving the AMP or DAC at the volume where there is the least THD is the best route, and then lowering software volume as you see fit?
Seriously, what's the point of DSD? (PCM I can somewhat grasp my head around with respect to what finally goes on a CD and such). I've read two things about this ordeal, here is the first and the second one I can't find, but argues the opposite basically in some fashion I can't recollect (I know this question is almost ridiculous but don't need a true answer outside of just people's opinion based on understanding how each is handled).
Lastly, can you forgive the insanely long post :-} ?