Hi jkorten,
I am disregaring a trivial source of differences between your tests.
Using loudspeakers, that have 10-100 times worse THD/IMD than analogue equipment, and frequency/phase response nowhere near as good as typical alnalogue hardware for comparative testing of digital atrefacts - is this going to work?.
Let me explain. My measurements on the codec show SNR=110dB and THD=0.003%. It would only be fair to assume, that SPDIF connections would not produce worse results than the above figures. So both techniques produced inaudible artefacts.
The codec results are particularly interesting, because during my comparison tests with SPDF, the codec approach actually inserts an D/A/D "bottleneck" running at 48kHz with 24bit depth with it's own jitter. I could not detect the presence of this bottleneck.
These results are similar to the experiment described in "Audibility of a CD-Standard A/D/A Loop Inserted into High-Resolution Audio Playback". E. BRAD MEYER, AES Member AND DAVID R. MORAN, AES Member.
Their bottleneck was much worse than mine. Here is their summary:
Claims both published and anecdotal are regularly made for audibly superior sound quality for two-channel audio encoded with longer word lengths and/or at higher sampling rates than the 16-bit/44.1-kHz CD standard. The authors report on a series of double-blind tests comparing the analog output of high-resolution players playing high-resolution recordings with the same signal passed through a 16-bit/44.1-kHz "bottleneck." The tests were conducted for over a year using different systems and a variety of subjects. The systems included expensive professional monitors and one high-end system with electrostatic loudspeakers and expensive components and cables. The subjects included professional recording engineers, students in a university recording program, and dedicated audiophiles. The test results show that the CD-quality A/D/A loop was undetectable at normal-to-loud listening levels, by any of the subjects, on any of the playback systems. The noise of the CD-quality loop was audible only at very elevated levels.
So, until you can show, that Mayer and Moran are wrong, I would be very sceptical about your results.
Best Regards,
Bohdan
MPD is anything but local. It has more than "several" remote control options. Perhaps you're confusing it with the only year old squeezlite? And the configuration is the standard UNIX-esque text file with plenty of notes t figure things out.
Why is it real? Because they are in essence recreating MPD? Why is it serious? Are the developers German and thus incapable of humor? Grand claims and not even a small paragraph justifying your assertions.
Also, MPD is free. And has been around for over 10 years. And is written in C as opposed to Perl. And MPD is serious as the current lead maintainer is really German! Link: Blog And they prefer stability over whiz-bang features and updates for the sake of updates. But of course you know all this.
They're recreating MPD, poorly. I've been to those forums and have read through all the buggy issues with the Logitech Media Software.
You can even sync those clients during playback.[/quote]
And MPD doesn't do this and more? Much, much more?
Linux supports more hardware than Mac OSX and Windows combined. Has for a long time. Remember USB3?
BTW, how are those native UAC2 Windows drivers working out for those audiophiles who have USB Audio Class 2 DAC and want to play material greater than 24/96? Oh, right. Microsoft never got around to endowing them with native drivers. Much less open source drivers.
@Bohdan - I don't listen with thd measurement devices. I use my ears. I am a musician, so I'm not so interested in the testing as it was conducted in your post. A bad amplifier (solid state class AB) could mask any important differences. So I'm not sure of the value of the listening tests. I am quite sure that high sample rate (192khz, 24 bit) sounds much better than 44.1khz 16 bit. If you can't hear it - you are blessed with no need to spend $ on high end audio.
A 10kHz square wave recorded and played back @ 44.1 is indistinguishable from a sine wave. But at 192 it actually more resembles a square wave.
But I understand your point of view and do not wish to dispute it. I am less fortunate and am chasing what I hear ($).
A 10kHz square wave recorded and played back @ 44.1 is indistinguishable from a sine wave. But at 192 it actually more resembles a square wave.
But I understand your point of view and do not wish to dispute it. I am less fortunate and am chasing what I hear ($).
You are just feeling you are quite sure, without a blind listening test. I have read many conflicting reports about perceived audio quality.
That test was blind, i.e. I trust the results the authors claim, i.e. that their subjects personally could not hear the difference. Was your test blind to claim comparably credible results?
It is not blind. I record my son playing piano at 44.1kHz and at 192kHz. The 192 sounds more like a piano to me. I'm using very nice Gefell MT800 microphones and a Grace preamp along with a Mytek192 ADC.
I'm playing back on a system with a Cary SLP98-P preamp, a set of Audio Mirror single ended triode monoblocks through Harbeth SLH5 speakers.
You are welcome to stop by and listen when you are in NYC!
Jerry
Well - not quite the case. In some part 192kHz does avoid the brick wall filter you need starting at around 19kHz to do the reconstruction filter during playback (@44.1kHz) which will avoid the alias that occurs above the sample rate in the low 20kHz range. But the damage to the original audio waveform has already been done by the anti-aliasing sampling filter during the record process at 44.1kHz. By recording at 192 - you only need to start anti-alias filtering at way supersonic levels (170khz for instance) which does not affect the audio band as much.
Yes I know people don't hear at 19kHz (much) but they do hear the effect that the filtering has on the texture of the sound.
Upsampling is used to avoid the nasties of the reconstruction filter that removes "tones" generated by the sample rate beating with the audio signal when using 44.1kHz. This is due to the sample and hold nature of DACs on playback BTW.
Sampling at a high rate to begin with moves the nasties of the reconstruction filter up to supersonic ranges and avoids the problem at the outset. And also avoids a filter in the audio range (20kHz) during the record process.
It may sound different to you, but unless you confirm with blind testing, the claim be hardly taken as anything more than... just your feeling. Certainly not as an argument in a technical discussion like this. Sorry...
I appreciate your offer but I am not the one claiming to hear that difference
Again, I appreciate your offer. I have done a few blind listening tests myself to know the striking difference between knowing what is going on and just relying on my ears to tell the difference. Try one, it will cost you next to nothing.
I study the underlying technology and technologically improbable claims have never been shown to be audible in a blind listening test, at least I have yet to hear/read about any.
But surely you would listen to a comparison and render your opinion, no?
I have no problems with blind tests btw - used in the gearslutz forum for microphone evaluation all the time.
Oh I would love to listen to your setup, to learn new stuff. I would just not make any conclusion about what difference I can really hear as I know my eyes would guide my ears. I am just afraid I will not make it this weekend from central Europe
But honestly thanks for your kind offer.It is great to hear that
Oh I would love to listen to your setup, to learn new stuff. I would just not make any conclusion about what difference I can really hear as I know my eyes would guide my ears. I am just afraid I will not make it this weekend from central Europe
It is great to hear that
Great - so I will post the files and you can have somebody else select which one to play (hopefully you have a DAC that will sense sample rate automatically, and a player besides iTunes that will play different sample rate files)!
Let me know when you get to NYC.
How were the files created? By decent (up/down)resampling on a computer, or recorded separately on different gear? If the latter is the case, of course they can sound differently.
Thanks
Maybe I wasn't clear, sorry. I will record myself on piano this weekend, using the exact same microphone placement and recording chain. The only thing I will change will be the sample rate during recording. The two files will be different takes, so I can't play exactly the same, but you can hear the change in the quality of the sound.
The recording chain I use is the Grace 101 preamp to the Mytek192 ADC. I use the Mytek192 DAC to create a S/PDIF to USB signal that then is acquired on my laptop using audacity. I will only switch the sample rate on the ADC and Audacity and then I will play the same piece again. No other changes, no resampling algorithms.
The recording chain I use is the Grace 101 preamp to the Mytek192 ADC. I use the Mytek192 DAC to create a S/PDIF to USB signal that then is acquired on my laptop using audacity. I will only switch the sample rate on the ADC and Audacity and then I will play the same piece again. No other changes, no resampling algorithms.
I am afraid if you play/record the piece twice, anyone can hardly conclude the difference (most likely detectable in a blind test too - too different recording rounds) is attributable to the different sampling rate. There will be many key factors different, starting with the human one But I believe it will be a pleasant event and that's what counts
But I believe it will be a pleasant event and that's what counts 
Only half baked test could be to try a downsampled to 44.1kHz version of the 192kHz against each other.
Even then you have to make other tests to rule out the DAC playing it back doesn't prefer 192k by design.
One more thing is to not worry about the filter needed for the 44.1kHz version. A pure piano recording shouldn't have any audible content that can ring or alias that high.
Even then you have to make other tests to rule out the DAC playing it back doesn't prefer 192k by design.
One more thing is to not worry about the filter needed for the 44.1kHz version. A pure piano recording shouldn't have any audible content that can ring or alias that high.
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