Yes, the dx300max tonality reminds me of my dx300 using the amp11. (An analog-like tonality)
that changes when using the dx300 w/ amp12 (has more "solid state" or "digital" like tonality. That sounds vague, but the two have very very different presentations in practice. Very obvious when you try it out).
But in aspects like dynamics, timbre, decay, etc 3MAX is an upgrade over the dx300/amp11. But the dx300max is diminishing returns over the dx300/amp11 if not using DSD512. Just shows how amazing the dx300 is.
Bravo! You guys got that correctly, but some people preferred it that way, and take it for being musical and realistic. Being subjective, we can not debate against other people preferences. However, based on technicalities models, there are objective matters, and explanations.
The reason why R2R is with those bloom and warmth, started out at the foundations.
Digital have 2 main stages, Quantitative and interpolations. Then at the final topology, we have Non over sampling and over sampling. Basically put, Non over sampling is 2X sampling of the human hear-ability which is 20-20Khz and that is Nyquist, 44.1Khz. Then whatever is doing a 2*1 is actually Non Over Sampling, and Over Sampling is actually 4X, 8X, 16X. What are these X ? Those are the calculations based off 32 bits length and multiplication factors. That resulted in DSD. So why the 32 bits length ? Nyquist Theory ! 16 bits and 44.1Khz. When going into Twice the sampling , that is 44.1/2 = 22.05Khz (cap out human hearing range), why not the 20Khz ? Well, let’s say that is head room. So, factor 2X to meet Nyquist sampling, that makes it 32 bits. Now, if you are Non Over Sampling, you do not further sample the frequencies any further, just apply interpolations and convert the bit lengths down to 1 bit, and into DSD to the final stage of Square Wave, which will then be Filtered out digital filters and analog filters, then finally become Analog sinuous musical waves. But there has never been DSD32 right ? However, there are DSD64/128/256/512, and those are from the factors of 2/4/8/16 multiplications as explained
Resistors ladders are being used for Quantization of binary info into electrical values, with resistors, there are precision accuracy, with that there are room for errors. Then with mathematical model, the correct tolerances should be 0.004% but, the modern technologies only allow discrete resistors of 0.01%, and that isn’t enough. However, there are ways to cheat. I can cover this later when needed.
So why does most R2R sounded exactly as you observed ? Because these precision tolerances results in quantization errors, that means a “Non Linearity” values from the binary input in references to the resistor to convert to voltage value output. This will involve error corrections algorithms, and it resulted in Amplitude being decreased to avoid clipping and distortions, errors in general.
This actually coincidentally explained why MicroSD can achieve better Dynamic. If you told me this 5 months ago, I wouldn’t believe you, and also those picture from Sony that shown noises is misleading, but somewhat educative for why it can happen to alter sound performances. From my recent experiments and observations, mind you, I don’t have bat ears, just loving music and high-fidelity stuff, the better the MicroSD, the better the Dynamic being achieved, and then other related performances could be observed together with it. The reason is that the bits and binary information are being retrieved with less “errors”, this means more “amplitude” from the original recordings , and that is ”better dynamic”
The reason why different playback app will have different sound signatures is also due to this, the so called ”variations in information received and processing”. This is why I always been sayings , there is no such thing as “bit perfect”.....because there is always “errors”, how it is being corrected, and how it is being retrieved will results in “variations of signatures”. This is why I adored DSD directly from studio, because everything were done by the studios, and into the final chain of 1 bit lengths that only need to pass over DSD-filters and analog filters to become musical analog waves. There are not much rooms for errors
The reason why I loved offline conversion of DSD is because OverSampling is DSD, and doing all of this conversion ”LIVE” will involve in too many elements that have errors over and over is just bad. So, why don’t we just use a built PC, an offline algorithms, and process PCM to DSD ? Yeah, that way you get the most out of digital over sampling technologies without too much errors.
However, and lately, I found out that when PCM Live conversion is being done right with the modern technologies , DAP , the end result is actually beautiful to listen to. This is because different DAC has it own modulations that effects the signatures, and different playback apps...etc....and I like those differences. Just so as long as it doesn’t have those Losses from Amplitude values .....or just as long as You have it done correctly and have great Dynamic delivery, that is what mattered most In My Opinion
Noises in Digital information retrieved (that resulted in Quantization errors, non linearity errors) should result in errors that look like this , and not audible noises like Sony is showing. This is exaggerating form of it, but you get the reason why you have bloomy textures and dark signatures don’t you!
One damn good explanation of error-driven non-linearities resulting to loss of amplitude, sir! DS topologies have, then, higher precision/lower tolerance built-in to their implementation than R2R?
Coupla questions:
1. Have you heard the Tera player (pain to use, but sound really great)? Or read the maker's notes about it being bit perfect?
2. So are you suggesting that the Sony MicroSD card is better for audio files?
claiming bit perfect has it plausible explanation, but it doesn’t exists in the real world.....nothing is perfect
no, I had Sony card, and it only does as good as Sandisk pro
One damn good explanation of error-driven non-linearities resulting to loss of amplitude, sir! DS topologies have, then, higher precision/lower tolerance built-in to their implementation than R2R?
1/ Oversampling with noise shaping , because interpolations can never be enough, so the more the over sampling , the higher the prediction of the frequencies given as the result
2/ DS is nowadays, mostly are known as being a DAC chip, or IC chip. Known that R2R also can be built in a chip, and DS can also be built on a discrete platform. But the top performers as ESS and AKM are chip based. Chip based has it own advantages due to Wafer silicon technologies, the precision can get as high as 0.005% and that together with over sampling up to 8X, virtually DS at the moment is superior
There are ways to cheat by using R2R discrete and While appear being good at it, and that is to use the resistors array as Quantization steps (the more is the better). Then using OverSampling interpolation down the chain. Cayin N6ii Ti is utilizing this way. Then many others are doing this as well
Also, there are R2R chip based, the Analog device chips being used in Luxury&Precision LP6. This player has 2 of them and the special version Titanium has 4x of them (remember what I said ? The more is the better ....it will approach to a point that you are laying it out in an array with as many counts as oversampling, after all, over sampling started out from that idea)
So, the OverSampling is pretty cool stuff actually , for example, if you read the book once and try to tell the story, you have more errors than if you were to read it 8x before you tell the story
So, the OverSampling is pretty cool stuff actually , for example, if you read the book once and try to tell the story, you have more errors than if you were to read it 8x before you tell the story
However, Oversampling actually has less sampling frequency at any given sampling occurrence than NOS, doesn't it? Is it because by doing it repeated times (keeping bit-size constant) they develop better algorithms for guessing noise interpolation than just doing it once (but with greater sampling frequency)?
Audiophile card reader existed, and that is your DAP. Remember ? Previously we were having problem with the way AndroidOS re mapping and resampling the data retrieved, and as a result, a dedicated OS will sound better ? Which was proven and observed ? Then we also had the problem of different SOC that would introduce errors and noises into the sound and degrade it ? For a while we couldn’t go anywhere near the “modern and powerful smartphone SOC” ?
Then suddenly we can all use it and benefit from it ? Thanks to the Chinese innovation! It is called FPGA-Master, SOC and it own noises with data, phases, clock ....etc...errors can all be corrected before it get further. This started out by MSB, then Sony, but then no one else could take it up a notch, now all China stuff do !! (MSB is pretty much the leader in digital music innovations)
Remember the strange effects that the higher end the DAP previously, the better the dynamic as a performance (out of many reasons though) ? Also, firmware would effect playback quality so much ? Even Unintentionally ?
Firmware will still effect sound quality because of the whole system and how it works, but without intentionally doing so, the differences is very subtle and mostly negligible, unless intentionally changed by the DSP (Digital Signal Processing) path, at least with the new innovation that we are having, the FPGA-Master
But then, a reader can only get as good as the given source get. With the 2MAX and especially 3Max, DX300, you will observe this effects from these new MicroSD cards which recently discussed. I don’t have bat ears, and so I believe the obvious effects can be picked up by about just anybody. If you can pickup cables differences, you will easily pickup MicroSD performances
Audiophile card reader existed, and that is your DAP. Remember ? Previously we were having problem with the way AndroidOS re mapping and resampling the data retrieved, and as a result, a dedicated OS will sound better ? Which was proven and observed ? Then we also had the problem of different SOC that would introduce errors and noises into the sound and degrade it ? For a while we couldn’t go anywhere near the “modern and powerful smartphone SOC” ?
Then suddenly we can all use it and benefit from it ? Thanks to the Chinese innovation! It is called FPGA-Master, SOC and it own noises with data, phases, clock ....etc...errors can all be corrected before it get further. This started out by MSB, then Sony, but then no one else could take it up a notch, now all China stuff do !! (MSB is pretty much the leader in digital music innovations)
Remember the strange effects that the higher end the DAP previously, the better the dynamic as a performance (out of many reasons though) ? Also, firmware would effect playback quality so much ? Even Unintentionally ?
Firmware will still effect sound quality because of the whole system and how it works, but without intentionally doing so, the differences is very subtle and mostly negligible, unless intentionally changed by the DSP (Digital Signal Processing) path, at least with the new innovation that we are having, the FPGA-Master
But then, a reader can only get as good as the given source get. With the 2MAX and especially 3Max, DX300, you will observe this effects from these new MicroSD cards which recently discussed. I don’t have bat ears, and so I believe the obvious effects can be picked up by about just anybody. If you can pickup cables differences, you will easily pickup MicroSD performances
Well, i'm usually using integrated microSD slot in my laptop to transfer files as speed is much higher. But to me it was oblivious that mSD slot (and interfaces) must be a part of Snapdragon 660 SoC.. Maybe i'm wrong of course.
Well, i'm usually using integrated microSD slot in my laptop to transfer files as speed is much higher. But to me it was oblivious that mSD slot (and interfaces) must be a part of Snapdragon 660 SoC.. Maybe i'm wrong of course.
By writing onto your MicroSD, digital and bits info is just binary information. It is not much effected by the way of reading/writing cycles, unless the data are corrupted beyond operational. This is related to OS level and whatever stuff that is beyond my knowledge at the moment. But anything digital that are not effected by time domain , such as printing a pictures, reading and writing are not effected , virtually, and that is why Offline calculation and processing for DSD is a more efficient ways to maximize the benefit from digital audio and OverSampling purposes. However, when both is done correctly, both are enjoyable
Writing and reading data that is then presenting back by the OS as informations such as file locations, folders....etc....not time domain related, then you are correct, the interfaces is from the 660 SOC
Bare with me, I try to learn the stuff after I observe the differences, and as I mentioned, not everything can be explained by sciences and technologies (YET), as a human, our understanding and Comprehensive abilities are vastly limited
So, let’s get back to the Dx300Max performances , we have had enough dosages of these digital stuff eh ? May be later down the road I can explain some what Titanium and Chassis materials
But then, a reader can only get as good as the given source get. With the 2MAX and especially 3Max, DX300, you will observe this effects from these new MicroSD cards which recently discussed. I don’t have bat ears, and so I believe the obvious effects can be picked up by about just anybody. If you can pickup cables differences, you will easily pickup MicroSD performances
I totally agree. Currently, I use a Phasure Lush 2 USB Cable in my home hi-fi set up which enables the user to 'Play about' or 'Tune' the component signal from Source to DAC. This has made me very tuned in to variances of sound. I'm not 'Bat' tuned by any means but I can distinguish sounds quite a fair bit.
The SD card scenario has now also surprised me and it really is something to consider. Yes, the Micro SD Cards are more expensive but are crammed with the latest and best resolving micro-circuitry. If you can, give it a go and start with a low capacity Card upwards from 256GB as this seems to be where the upgrades start with upgraded technology. If you like and can distinguish a difference, upgrade to a larger capacity and invest if you wish. My view, after you have decided you like what you hear and go for it, invest in multiples of either 256GB or 512GB or a combo.
claiming bit perfect has it plausible explanation, but it doesn’t exists in the real world.....nothing is perfect
no, I had Sony card, and it only does as good as Sandisk pro
12V charger with 1.5A or 18W minimum . I charge my 2Max with 12V laptop charger that is 12V and 36W
If you have some time, I’d be interested to hear your thoughts about the Tera player DAC approach.
Think I might just pass on the confusing Micron cards and keep using my Sandisk Extreme Pros.
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don’t you!
, we have had enough dosages of these digital stuff eh ? May be later down the road I can explain some what Titanium and Chassis materials 
