TwoEyedJack
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- Jun 2, 2022
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Hello, I am trying to reach a conclusion regarding what settings should one use in windows audio settings (I understand that most of the things I am going to ask will probably be inaudible to me but i just want to know for the sake of knowledge). DAC's are know to produce intersample-overs and clipping when trying to upsample 44.1khz signals.
Do DAC's always try to upsample their input signal before converting it to analog? Is this just their operating principle while trying to interpolate the rest of the "analog samples" in-between the digital ones?
Does windows/software up sampling also suffer from this besides the rest of the IMD/Aliasing distortion it introduces? Apparently it used to although testing done here suggest things have gotten better.
Are the Intersample-overs affected by the sample rate of the input signal? Meaning are there fewer cases of clipping if the signal is getting up sampled from a higher base sample rate? If so would it beneficial to use software upsampling to bring it closer to the target upsample signal of the dac? This comes from the debate about whether you should go back and forth to change the sample rates of windows depended on the sample rate of the audio file you are playing. The advice given here was to just set it to 96khz since the windows upsampling was improved in the past years.
But I am skeptical about this since it's a given that you wont get any benefit from playing back higher sample rate files (in contrast to mixing an producing) , and usually-depending on your equipment- you might hurt performance . In my case while my dac is capable of 24/96khz I can hear loud intermodulation distortion if i use the 24/96 setting on windows and playback the test tones (supersonic tones).
Lastly do you get the same issues when downsampling? Going from a 96 source file (or a 48 movie) down to a 44.1 windows output, could this hurt performance -not loss of hypothetical supersonic fidelity- in a meaningful way like distortion, clipping or aliasing?
Do DAC's always try to upsample their input signal before converting it to analog? Is this just their operating principle while trying to interpolate the rest of the "analog samples" in-between the digital ones?
Does windows/software up sampling also suffer from this besides the rest of the IMD/Aliasing distortion it introduces? Apparently it used to although testing done here suggest things have gotten better.
Are the Intersample-overs affected by the sample rate of the input signal? Meaning are there fewer cases of clipping if the signal is getting up sampled from a higher base sample rate? If so would it beneficial to use software upsampling to bring it closer to the target upsample signal of the dac? This comes from the debate about whether you should go back and forth to change the sample rates of windows depended on the sample rate of the audio file you are playing. The advice given here was to just set it to 96khz since the windows upsampling was improved in the past years.
But I am skeptical about this since it's a given that you wont get any benefit from playing back higher sample rate files (in contrast to mixing an producing) , and usually-depending on your equipment- you might hurt performance . In my case while my dac is capable of 24/96khz I can hear loud intermodulation distortion if i use the 24/96 setting on windows and playback the test tones (supersonic tones).
Lastly do you get the same issues when downsampling? Going from a 96 source file (or a 48 movie) down to a 44.1 windows output, could this hurt performance -not loss of hypothetical supersonic fidelity- in a meaningful way like distortion, clipping or aliasing?