Good thing you or another told me... I cant spend the time sorting thru all the possibilities. So, thank you for making it easier to get something to work properly or well. .
🙂 😎
-Richard
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You and the others here are why we created the Auraliti player. It was designed so you did not need to pay much attention for the file types. Unfortunately it was the wrong business model. We should have built $10K players instead of ones uner $1K. As it was we got little respect for them.
Hello Mr Marsh.
I use an old laptop with external USB Hard Drive storage and USB Dac using Foobar with ASIO driver.
With correct plugins Foobar plays pretty much any file at the original file format/sampling rate and seamlessly bypasses Windows SRC issues with Foobar and ASIO applet reporting currently playing file sample rate.
You ought to be able to get this kind of setup running pretty easily on pretty much any old cheap PC or cheap laptop and ASIO capable Dac, and in my experience this setup sounds perfectly good.
Using a laptop can also provide local network connectivity and control, also access to streaming radio/music services.
How were you using your Benchmark 2 Dac ?.
Max.
Yes, real time
But not DSD, my recollection is that there is a modulator fed by a random signal involved in the construction of the resultant audio, not deterministic. (my memory is hazy on the subject)
Cheers
Alan
Finally had a look a that link. Sadly doesn't cover the ASRC chipsets currently in use, or the standard linux resampler. But does show Foo_DSP to be good. Of course my use case is going 44.1 to 96k which is the other way around. My use case is odd as I use minidsp at 24/96 to gain a few more bits for digital volume control. I could use the minidsp inbuilt SRC or resample before delivery. Currently get the server to that, but intrigued to know what the amateur can do to test the efficacy of this approach...
AFAIK DSD was originally developed by Sony as a way to store content in their archives, independent of change in future technology.
Then came the idea to bring this "almost analogue" Direct Stream to the market, but there are two reasons why this never happened.
1) To perform the necessary Edit and mix DSD while recording, 1 bit recordings have to be converted into PCM and reconverted back into DSD (this results of course inevitably in some conversion losses).
2) Progress in technology made higher multibit sampling rates no longer an issue, outperforming the 1 bit 2.8Mhz DSD concept.
This all caused the DSD to become a mere marketing issue and real Direct Stream is nowhere in existente at all.
Apart from an added surround sound feature, there is no advantage in DSD or SACD, but you get a lot HF noise to cope with.
This article goes into somewhat more detail.
https://www.grimmaudio.com/site/assets/files/1088/dsd_myth.pdf
Hans
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Bill, pretty sure the miniDSP input ASRC is always in the loop, even with matching sample rates.
I'm sure it is, but is there any benefit to having it convert or run 1:1. I don't know...
Is there more than one master clock in the system? If more than one clock domain then you would need something, typically ASRC, to bridge between them. Of course, some people use a big FIFO instead.
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"Some have suggested that this HF noise may change the behaviour of capacitors, cables and electrical contacts. We have not done any research into this ourselves, but it should be relatively simple to set up a listening test featuring a third file: the original 192/24 file but with added HF DSD noise taken from a silent DSD recording. If the ultrasonic noise is responsible, this file should sound ‘better’ than the original 192/24 without the added noise."
🙂
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🙂
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You miss the point. The reason they are used in DSPs often is to only have to use one set of coefficients.
I do not know how it clocks it into the shark. My interest is what could be done without needing an AP/digiscope/RS or similar. As there is an SPDIF out from the box and I can record that I wonder if there is 'something' that can be measured.
Yes I believe that's the game plan with the miniDSP stuff. I suppose it also allows a bit more flexibility with master/slave clock arrangements, and can theoretically provide input jitter reduction (and hopefully the downstream hardware doesn't just make more of its own).
Yeah, it has a lot of benefits. I should clarify that Mark wasn’t wrong either, but I see that as the primary motivation in a lot of commercial applications.
No argument ASRCs are often used with DSPs because of that.
However, wasn't Bill as asking about interfacing two things already at the same sample rate, whether or not ASRC would be still be needed?
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No I was more asking if there would be a measureable difference between SRC from 44.1/16 to 96/24 in the music server and letting the SRC in miniDSP do it. Internally miniDSP works at 48kHz or 96kHz depending on the plugin.
The challenge for me is what to measure if I am comparing bits in with bits out.
The challenge for me is what to measure if I am comparing bits in with bits out.