Surely we are now talking of two different things? Reproduction closest to original with high fidelity, ie as close to the original if you played it back in the same studio conditions, and I would include high resolution in whatever form. The second which I think we are talking about is how close to that we want ‘our’ sound to be at home. Do we actually strive to hear it as the studio engineers and band heard it, or do we strive for our own sound? I prefer the sound of Naim with Spendor speakers, both have couloration. I try and balance that with the best source that gives me clarity and silence between notes. It’s just one approach, there are others.
I’m not technically aware enough to understand the design of electronic components, and I include speakers and crossovers in this, but if I were to design any component I’d probably try and make its best operating window in the middle of its range rather than at one end or the other. That means for me that a large speaker utilising a very small movement of the volume control at the bottom of its range is just as bad as running a small speaker with the volume permanently set to loud as possible. Room size, or more correctly air volume, therefore influences the speaker/ amp combination.
So many variables!
I think I need a hifi guru to come round and design a system for each room 
Sorry I wasn’t clear I was referring to a studio microphone chain through to replay focussing on technical constraints and current equipment. There was no production sound manipulation or aesthetic processing in this example, which yes is mostly used in commercial music/recordings to make the end result sound as appealing as it can by often create a ‘feel’ or vibe to a mix… and that is a different thing as you point out.
Personally I like to hear the production as it was intended to be heard where ever possible… that is part of the aesthetic with recorded music and part of the interplay of musical sounds…
One of my favourite examples of this is that is sooo poignant… Martin Hannett’s production of Joy Division’s Unknown Pleasures. A masterpiece of dark emotion oozing out of the music that can drive me to tears when in certain moods. Compare that to the earlier recordings and productions on the Warsaw album of many of the same songs but without Hannett’s superb recording and production engineering skills and artistry… good as they are they just don’t have the same effect.
Bear in mind 4 phenomena.
- With long wavelengths, the energy from the loudspeakers tends to be more omni-directional, i.e a woofer’s polar response is 360 degrees.
- Boundary reinforcement. Sound waves hitting a wall, floor or ceiling will be reinforced by the reflection
- The ratio of sound pressure level to distance. Based on the Inverse Square law, a doubling of distance reduces sound pressure by -6dB
- Absorbtion vs frequency. The higher the frequency, the thinner the material required to absorb it. A typical, comfortably furnished room will absorb more midrange and treble and little bass. Thus its easier to achieve the ideal reverberation time (RT60) at higher frequencies
In a small room, both the speakers and the listener are closer to all boundaries. Thanks to its polar response the deeper bass is bouncing off more boundaries than the midrange and treble frequencies, the boundaries are closer together so there are more reflections at higher SPLs and the bass energy is not as easily absorbed as the midrange and treble so the RT is longer. The effect is that in a small room bass SPLs are attenuated less than higher frequencies while being reinforced more.
In a larger room, the distances are greater, so there is greater distance attenuation and fewer boundary interactions at lower SPLs so therefore less reinforcement.
The problem with larger rooms is that the bass nodes are at lower frequencies and more discretely spaced, so a lot more obvious.
And of course whilst for some people their preferred sound is different, for others it is indeed largely as the studio engineers heard.
With music recorded live (as opposed to substantially manufactured/assembled in the studio), it might be expected that the studio engineers will have tried to get it to sound like the live sound, so sounding like that might be expected to be as close as one can expect to get to hearing it live, some I believe is not uncommon for hifi enthusiasts to hope to emulate as far as possible within the constraints of their home. But then of course it also depends which version of a studio engineer’s listening - main monitors, replaying “as it is”, or mini monitors replaying as the mass listening public would through basic radios, cheap “stereos” etc…
Just wire your chosen end point to a wireless access point or a mesh wi-fi satellite unit. Pay attention to the ethernet cable and power supply/grounding of the access point/mesh unit and you benefit from not having a relatively noisy wi-fi receiver active in your end-point.
OP here again. There seems to be a bit of a difference of opinion as to whether the transport makes an audible difference if you’re streaming bit perfectly. If you go to a site like Audio Science Review, they will contend that you can’t improve on a bit perfect WiiM Mini as a streaming transport, and all the chatter about clocking, jitter and power supplies in the context of the transport is nonsense. Then I read reviews of very expensive streaming transports made by Auralic and HiFi Rose that suggest these make a huge difference to the sound. Someone in this thread even suggested that the WiiM Pro is audibly better than the WiiM Mini as a transport, but I have not been able to find any support for that. I suppose I have to trust my own ears, but I am none the wiser after more than 200 responses to the OP.
What do you mean by “chosen end point”? Right now I have a SonicTransporter (I guess you’d call it a NAS) that I use as a Roon Core (housing all my downloaded/ripped files), and that is plugged into my cable modem. Sound is provided via a Burson DAC connected to a WiiM Mini via TOSLINK cable, so that the WiiM is used as a wireless streaming transport. Amplification is provided by an SN3. I do have a decent mesh wifi satellite system, but none of these units are plugged into it - everything is wireless but for the Core. Can you clarify which of these devices should be wired to the mesh satellite? Many thanks.
Hi - there might be an mis understanding or lack of understanding by some … bit value accuracy is just 50% of the equation - the other part of the equation is time on a bit stream and that is where the variation typically comes in - it is nothing to do with the bits… but to do with the time consistency.
Now sample bit streams typically use ‘zero order holding’ (that is the sample or bit value is held through the clock period) and therefore timing of the held sample values is determined by the serial stream clock within the digital streamer. Now depending on the sample coupling the impact of that clock variation (jitter) will have different effects on the receiving DAC and associated circuitry… but even assuming the receiver clock was totally decoupled from the sender clock - and given they are free wheeling in most setups this is unlikely to be completely the case - the clock stability from the digital streamer noise or phase noise would at least couple into the receiver as noise power or phase noise power. This phase noise and therefore clock stability is down to the digital streamer’s clock stability and bit stream encoding circuitry - working in conjunction with its front end network transport module. Note this bit stream clock jitter is almost never linked directly to the DAC sample clock stability these days.
This is part of the reason why different digital streamers can induce different signatures into the receiving DACs, albeit the ‘sample numeric values’ are identical , and the more stable the clock the less noise power created. Another consideration is common mode noise coupling - but that is a different topic
Are you saying that jitter-noise is somehow worse - has a more detrimental effect on the final sound produced by the dac - than other sources of noise? (Genuine question)
I don’t think so… I am saying clock phase noise can couple into other sensitive components like DACs… and timing with sample values together define the sampled data in a bitstream.
I’m down a massive internet rabbit-hole about dac sample-value error due to noise. Fascinating. Thank you. 
As you are in Dublin, there surely are hifi dealers. It shouldn’t be too hard to exit the rabbit hole by demonstrating two transports at opposite ends of the cost spectrum. While the “why” is a fascinating topic, if you just want a yes or no answer, the way to get that isn’t too cumbersome.
Reminds of the article from the early Linn/Naim days over 40 yers ago:
High Fidelity - to what?
I.e. should you prioritize fidelity to the acoustic events or to the music.
BTW, I think I badmouthed the Holo RED streamer earlier in this thread.
I changed the I2S-cable to an Audioquest Vodka and the harshness went away (I was using a cheap HDMI 2.1 cable from Amazon).
I now use the RED with the Holo Spring3 KTE and if you like the more natural sound of NOS-DACs it turned out to be a really excellent combination.
no worries - there are many cases of ‘errors’ for DAC reconstructions. The attribute I am referring to here is its clock - and the modulation of that clock can be directly or indirectly though noise caused by other instability of other coupled clocks such as the clock from a digital streamer via its SPDIF or asynchronous USB connection.
If you can picture it an analogue signal is a smoothed out continuous signal of discrete sample values occurring at specific points in time. If that time varies so does that reconstructed analogue signal even though the sample values are bit perfect. Hopefully that simplified picture makes sense.
Currently running an ND5XS2, and have an NDX2 incoming. It’s purely research for my own understanding / nerdiness. Many years since I studied all this in college.
This topic was automatically closed 60 days after the last reply. New replies are no longer allowed.