Hi, when I originally asked this question ChrisU replied that most 2960’s have a good power supply inside. Care to share which one you are using, and how much difference it made?
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I experimented with several Catalyst models, some with onboard PS and some with external SMPS. I noticed an improvement with the latter models with external SMPS. The old blue 2960 did the worst of the bunch and the newer 8PT and 8TT POE models did the best and that’s what I’m using now.
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Hi Simon,
Just to be clear, I am not talking about noise shaping or distortion tuning. I am talking about noise and distortion reduction, such that the final digital music stream/parcelling is as high a spec as possible in terms of noise content, vibrational losses, cable losses, timing and voltage transition perfection.
The goal isn’t filtering, its removal of unwanted filtering, unwanted noise, transmission losses and distortion.
I am aware of the techniques used in music production to enhance our perception of the sound, basically shaping the sound by introducing certain effects and filters that appeal to our listening tastes. Indeed almost all recordings have a degree of this, but I wouldn’t want the filters applied twice, which is what would happen if you applied them via the replay chain. You would also get the same effects on all recordings, some will benefit but some will not. For me digital is all about getting the accuracy correct and letting the recording look after itself. There seems to be a ton of really good vibes on a wide range if recordings….its just a matter of making sure our replay captures the whole content and presents it with minimum limits, edits or errors.
In my experience, the structure and degree of perfection of the bit stream, parcel stream, radio stream or light stream and all the conversions between have a massive effect on the final sound and presentation of the music. There’s also considerable opportunity in controlling, optimising and minimizing network traffic, noise and processor interruptions
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Sure, but D to A conversion is necessarily only an approximation, a close one but in the limit of micro dynamics and other subtleties an approximation… and lossy at that…. So there are design choices an engineer has to make to choose his or her poison as it were… one really obvious design difference choice one has to make is to adopt either IIR or FIR reconstruction low pass filtering… both have strengths and weaknesses. These differences can, by various other aberrations (distortions) may appear to be in harmony or not….or that the distortions introduced by a method are outweighed by the benefits with the chosen implementation.
Naim’s choice of IIR reconstruction using Analog Devices SHARC is a good example.
Several other contemporary manufacturers use FIR reconstruction with large filter response kernel sizes (taps)… which of course would be irrelevant fir Naim’s choice… and yes their implementations do sound different to me with different characteristics… quite noticeable with planar headphones and quality live recordings for example. Also the reconstruction balance is in my opinion pretty consistent with IIR, but less so for FIR with different sample rates.
The i2v circuitry is another interesting design choice… ultimately a compromise, but different approaches to linearity are made as design choices.
Reading your posts I suspect you know most if not all this… and therefore I am sure you appreciate the design choices and compromise choices to make (distortions or approximations) (and their interactions) in what is ultimately a lossy compromised process.
But yes I agree it’s about getting one’s audio to sound the best/most pleasing in a particular setup, but alas some productions will be mastered to ultimately sound less pleasing on certain high end setups… as that will not be aligning with the most common target system capabilities. How often have we heard that?
But honestly network (Ethernet) traffic (http tcp payloads) is a country mile away from precision timed bit streams feeding DACs. They are dealing with different concepts … and of course that is where components like the impressive AD SHARC processors help prepare conversion and separation between the two…
But yes as I have said before in a closed system, which essentially a combined streamer DAC is, one can’t perform a function to an input without introducing an error in the real world… good old fashioned system and control theory. So in such a system changing the input will almost certainly affect the output… but highly likely to include artefacts that are nothing to do with digital signal data.
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Hi Simon,
Thanks for the interesting reply. Again, just to clarify my position regarding the network, what I am talking about is making the feed into the DAC as accurate, compliant and perfect as possible; ideally no noise, no jitter, no cable losses, no EMI addition, no vibration/resonance losses, no extraneous non-music related network traffic, perfect voltage transitions with no noise, timing anomalies or stray currents through poorly implemented screening…in other words as close to the desired specification as possible. I am talking about making sure that the DAC, with whatever engineering choices were made by the designer is fed as pristine and perfect a digital stream as possible.
I understand that within a network, there are multiple formats and many conversions. Again what I am talking about is aiming to have each of those streams, be it light, radio, packet data or voltage streams as perfect as possible. When that’s the case, the conversions are performed more precisely and the resulting sound is better. Take any of the typical network physical formats listed above, improve it by making it quieter, more accurate and better timed and your music will take a leap upward in SQ. When all the streams are optimised, the gains in fidelity, musicality, listener involvement and enjoyment can be really stunning…the equivalent of a very major and profound system upgrade.
Following the above philosophy for the past 3 years, I have uniformly obtained improvements to the SQ. Excellent recordings sound even better and even the less accomplished files, for example 128kbps MP3 internet radio becomes extremely engaging and beautiful. Its very rare for even the worst recordings to sound worse. As a minimum it becomes possible to understand which part of the recording process was poorly performed while still enjoying the composers’ and performers’ skill and artistry. The more I have improved my network, the fewer the number of recordings that I would designate unlistenable. There are still a few but its a vanishingly small percentage.
Regarding your final point, removing noise, jitter, vibration and minimizing losses are not functions that change anything, they are merely improvements that render the same stream more exact, less noisy and less in need of correction and are analogous to changing a microphonic or noisy tube.
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Hi thanks for the reply… I agree having a system introducing minimum additional artefacts is going to be ultimately desirable with other things being equal.
One possibly last thought…. Not saying it’s the case with you… but if one finds you are very sensitive to changes in the audio chain… and if one finds some tracks unlistenable, from the production approach as opposed to musical content 
, one might find in the replay chain most probably with speaker/room coupling, there might be resonances, poor frequency response or comb filtering that can make one very sensitised to changes that can become more prominent or apparent at certain frequencies… to possibly check this it’s worth using high end planar headphones and a good headphone amp to see if one still finds the track unlistenable.
Noise shaping isn’t all in the digital domain…
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What I tend to do is hook up a streamer of some kind with an ethernet cable from my draw of cables, whack on roon, sit back with a beer and enjoy music.
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A very nice relaxed way of enjoying music.
DG…
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Actually to be fair I would say 80% of my music listening nowadays is in my office, so with a cup of tea and a pair of meridian M33s active, some of the best bang for buck speakers I have heard. Hooked up to a raspberry pi into a lovely Gustard dac. Whole set up 650 quid.
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The one I’m trying on my 2960 is a bel HC12-3.4-A. I don’t say it’s the best choice but it did give me a better sq-timing from my NDS.
Interesting to note is that removing the SMPS from the power outlet did seem to have a positive effect on other LPS-electronic.
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I think you may have completely confused and mixed-up digital vs. analogue.
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Thanks frankie, your comment tells me a great deal.
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I am increasingly finding wifi preferable to Ethernet for streaming and audio … there is a whole load of noise related issues avoided with wifi that gets me close to the same noise benefits of fibre but more flexible. Where I can I try and use ac… my APs don’t yet support ax - and that has more advantages that could be helpful to audio streamers. Though still use Ethernet with my ndx2 … but now some of my main music servers/proxies are wifi connected.
However if you are the sort of household that has a single or non overlapping Access Points blasting a signal out, or longish wireless mesh chains, then I would stick with Ethernet.
I also have ensued my wifi equipment supports WMM (wifi prioritisation/COS) but not yet needed to DSCP mark the network frames yet as I really don’t suffer from local wifi congestion.
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Well that’s no fun for the audiophile because there is no wires
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Argh… yes I thought of that one… we might see a market in audiophile antennas in time? But I admit there would appear more money making opportunities with Ethernet for the audiophile…
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What about audiophile air to get a purer signal from router to streamer ?
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Indeed, though radio can pass through and reflect off many things other than ‘air’… but if you are using a single wifi access point built into a router… you are quite probably not using an optimum wifi set up anyway.
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That may well be true Simon…. But it wouldn’t stop some buying it 
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Very true……
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I suppose there is room to use LPS for the AP’s…(unless poe is better…)
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