Does he need volumes to be at 100% to playback this way? I’m not sure just a thought.
Nope, on Windows you use NativeDSD for this kind of sample rate. Generally, if you have the ASIO driver installed you should use NativeDSD as playback method.
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Yes, the volume needs to be 100%.
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This issue was discussed a few days ago. Reason has nothing to do with it
If his DAC is DSD512 capable, he will get the best sound at this rate. That’s why he does it.
A USB-2 cable is sufficient to transfer audio at this rate. But a short USB cable is always better.
For USB connection, I always use 30-40 cm cables.
It’s a driver issue, most likely.
“Best sound” must be qualified, beyond just a subjective opinion… If we talk about resolution, we must take into account the resolution of the hearing neurology… Yes, natural sound has a defined resolution and this is our absolute reference, however, when a quantified 16bit/44.14kHz or higher resolution PCM signal is recorded, inherently, there is a limited amount of information from which to modulate the PCMxxx signal to DSD512 or DSD1024… A higher-sample rate above DSD128 will not impart tangible harmonic and dynamic information that is perceivable in this context… In the context of a natively recorded DSD256 or DSD512 product there will be tangible dynamic and harmonic information captured in the A/D modulation.
It has nothing to do with all this.
The reason for which people upsample to the highest DSD rate of their DAC, DSD512 in this case, is because the player can run with the CPU of the computer better upsampling algorithms than the chip of the DAC. Otherwise, it’s the chip of the DAC that will upsample the sound to DSD512 with its low quality algorithms, and the sound will be less plesant.
@Cloclo
This is not necessarily true… I can perform PCMxxx to DSDxxx modulation in an offline process and then play this file back in the native modulated sample-rate… Either way, the source data has been quantified as PCMxxx resolution… even a 768kHz PCM A/D signal cannot capture the level of dynamic and harmonic detail of a PDM 1-bit A/D DSD128, DSD256 or DSD512 recording… At best it boils down to cognitive bias and physical biases of the hearing neurology and the playback system capabilities and the acoustic environment (speakers in a room or headphones).
You don’t understand.
The upsampled sound, to whatever rate, is not better than the sound of the 16-44.1 source file. It does not matter if it is upsampled offline or on the fly. It cannot be better, because it cannot create an additional audio data that does not exist in the source file.
But with a Delta-Sigma DAC, it just sounds better, because these DACs are built to proceed with the digital to analog conversion at a given DSD rate. The highest DSD rate for which they are capable.
If the upampling is done with better algorithm than the low quality algorithm with which the chip of the DAC works, the analog sound that you’ll hear will be more resolved and more detailed.
That’s why people upsample to these rates. And people who want to go further use a dedicated upsampling player that has very high quality upsampling algorithms. These algorithms require a very powerful computer.
In the context of your R2R device this is probably true…
In the context of a smart modulation algorithm that imbues interpolated data points… there is more dynamic and harmonic and phase information being filled-in between the PCM data-points, because there is space for the 1-bit samples in the increased Fs.
This is too narrow in interpretation… The PDM signal is a reinterpretation of the modulated PCMxxx data signal and this corroborates my statement regarding modulation of PCMxxx beyond DSD128…
Excuse me, my friend, but you elaborate on a topic on which you are completely ignorant.
What I say is true only for Delta-Sigma DACs.
With my R2R, I do not upsample at all, because I hear no tangible improvement in doing so.
And people who do it with their R2R DACs obtain only very little improvement.
I told you all this a few days ago.
@Cloclo
Look into the RDOT-Neo algorithm used in the newest TEAC DAC’s
I previously to used XiSRC to modulate PCMxxx to DSDxxx offline and it’s algorithm used interpolation to flesh-out the PCMxxx data, so to put more “flesh on the bone” to the DSDxxx product.
Pffff… How can the chip of your DAC run an algorithm comparable to Sox or r8brain?
And these are not the best upsampling algorithms. To go further and have a better sound, you need to use a player that is specialized in upsampling, and who has heavy-duty algorithms.
But I don’t want to elaborate, because I don’t want to name competing players on the site of Audirvana.
It’s very simple… a dedicated SOIC doing one operation (modulation and interpolation) in concert with the DAC platform… I’ll take a dedicated SOIC system over a playback-engine trying to do it all, every time… This is digital-audio design 101 at the hardware platform level…
malheureusement non
Which DAC do you have?
It’s true that a dedicated chip is more efficient for a given task than a general purpose CPU.
But to run in real time the best upsampling algorithms, you need a computing power that it’s completely out of reach by the chip, as efficient as it may be.
Just to give you an idea, to upsample to DSD512 with a specialized upsampling player while running the most demanding algorithms, Dither, and modulator, you need a gaming PC with a powerful NVIDIA GPU board for CUDA offload.
You can do the same with a less powerful Mac, because macOS exploits more efficiently the hardware of the Macs. Nevertheless, you need at least a M1 Mac.
en effet je trouve que le son est meilleur, et déjà testé avec les sur echantillonnage sur off et rien ne change, je reste persuadé que le problème viens du nouvel ordi;
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