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Bit perfect software changing sound. How?


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2 hours ago, March Audio said:

@kumakuma

Manueljenkin has asked that I do not respond in his thread where he is promoting a certain audio player. 

I haven't blocked you from replying or posting, but requested you to not try to inflict unintended deviations. I have kept your first post as is for reference because you put effort to check things. I have made some observations that are different from your assessment, and I have added that also to the topic. The player does load the music into the RAM as I've described in my post with the evidence. Windows task managers may not reflect all changes, and you'll have to observe carefully or try with a bigger file.

The rest of your comments were all off topic, hence I removed them since it was deviating from the core point of the thread.

 

2 hours ago, March Audio said:

@kumakuma

 

You have posed some objective questions in that thread which in theory is not allowed as its the "subjective" section.

I haven't found any issue with his questions so far. They were genuine and intended to check things. If it were actually asking for measurements or other objectivity oriented question, I myself would have directed those posts here.

 

2 hours ago, March Audio said:

@kumakuma

 

Lastly I have tried optimising many files.  Also optimising the same file multiple times as suggested.  I can hear zero difference between original and the "optimised" files.  I already demonstrated in the original thread that the files are identical (as the software author states) and the original and optimised files null perfectly.

Few things. The developer's recommendation is to not make a copy of the file, move it across different partitions and/or physical drives, and not make modifications to the content of the file other than renaming. Certain music playback software make a temp copy before playback and it would nullify the test as per the above. It is recommended to make the tests with the junilabs player if you haven't to make a fair assessment.

 

I'm also curious to know more. Good luck!

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5 minutes ago, kumakuma said:

 

Can you explain walk me though what this evidence is?

 

I looked at the screen caps you shared but don't see anything indicating that a 1 GB file ISO file had been loaded into memory.

Memory usage level changed from 33% (before starting optimizer) to 64% (after starting optimization). My system has 8GB RAM and 30% of it would be about 2.4 GB. No other process was run during this time and did it multiple times to confirm.

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12 minutes ago, PeterSt said:

This thread should not have been started by someone with insufficient knowledge of the Windows OS.

The forum is an open place. Let them have their posts. His assessment has flaws because of the unreliable probes he has used, but as can be seen, it can be refuted quite easily.

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8 minutes ago, kumakuma said:

 

So what's the answer to the question posed by the OP?

How it optimizes is uncertain, but the flow of actions that the dev described looks true so far. The assessments by march audio on how the tool works have many flaws - it actually does RAM loading as the software dev mentioned, it doesn't seem to defrag unlike predictions from @Currawongand few others. And the assessment of fidelity/changes through foobar ab-x also break the usage conditions of the tool as foobar ab-x makes a temp copy of the files before playback which the dev advices against. I can only give sighted experiences at the moment and I'll reserve it to subjective thread. For an assessment to March audios requirements, it would be suitable to try the same comparison with the junilabs player but with helper to properly ab-x (need to make sure the helper doesn't do anything that breaks ab-x) so that both the player requirements are fulfilled and ab-x test is also fulfilled.

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9 minutes ago, March Audio said:

I find it ironic that Manuel requested admin rights in his thread to prevent me from responding in it, yet claims freedom of speech here.

I requested admin rights to ensure that the thread stays on topic and not to block any particular individual. A flawed analysis is fine, and I have kept your original assessment on the thread and made my observations refuting it. Comments that try to deviate the thread, use weasel words and passive aggressive tone, are the ones that get removed.

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2 minutes ago, March Audio said:

 

 

I will respond to this.

 

The floor is yours to come up with some kind of cogent explanation as to how it works.  Please propose a theory of how it might work.

 

Can you explain why making a copy of the file breaks the alleged "optimisations".  We  know it doesnt change the file, from the null test and the authors own testament.  So whats left?  File reading position and continuity on disk?  Sorry but as even you agree, that is irrelevant with SSDs.  Yet you still claim you hear improvements playing off your SSD.  So what is it doing???????

 

I have just proposed some methods of analysis.  Isnt it quite comically convenient that the "usage conditions" prevent such objective analysis?  Conditions that so far have no rational explanation or justification.

There's nothing preventing objective analysis within the boundary conditions mentioned. I have mentioned a suitable test in my previous comment.

 

I cannot predict how it might work, I am also curious of the same, since it does work as the dev claims on my system. The above conditions just rules out those particular predictions (defragmentation) on how it works. There's an entire other sea to explore when it comes to electromagnetism, and access noises. It doesn't change the "data" in the file, but it does create a new file (ie it does a change), as evidenced by the fact that it copies to RAM before optimization.

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22 minutes ago, March Audio said:

There is nothing passive aggressive about objectively looking at the software.  Thats just your erroneous assertion. I understand why you want to attack an alternative view that contradicts yours.  You have been directly offensive towards me in the previous thread. You even called me a liar at one point.  You have repeatedly used ad hominem attacks towards me in multiple threads.  So dont go there.

 

This is an objective thread.  Do you have any contribution to explain how this software might work to improve sound?

 

From my objective perspective so far this appears to be firmly in magic crystals and homeothapy territory.  My subjective analysis also concludes it does nothing to improve the sound.

I didn't remove your "assessment" post in the original thread fyi. I have also made my own tests showing that your assessment is flawed and it is also present there.

 

I only removed your other comments that were nothing but a culmination of weasel words and passive aggressive attacks. Even your "assessment" posts had such references but I have let it be.

 

And fyi: 

 

In every other thread most of my comments directed to you were only pointing out flaws in your assessment and assertions. You had been stacking up conjectures on the "gaming measurements" thread where you inserted so many uncorrelated phenomenon and often misunderstandings of - assumption of types of mics used in phones, assumption of human hearing perception with respect to cocktail party problem, etc. Almost everyone in the thread has shown you immense patience and tried to correct where your assumptions were wrong. If you felt that to be an ad hominem attack, I guess we'll need some class on literature/grammar.

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10 minutes ago, PeterSt said:

 

This means that you must have your measurements correct. Your OP does testify of the contrary.

You thus put all in a perspective of attacking the author and THAT with so-called objective assessments ?

 

Your thread thus derailed in the first post already. Especially in the objective board you should have your facts right. But as usual, you just shout around and love yourself with it. What about toning down and have a real discussion ?

Exactly. So far the posts by op on this thread has only been a poorly researched, personally structured attack on the author, disguised as something "objective".

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5 minutes ago, March Audio said:

Well explain how it works Peter.

So your objective is not to find if it works or not. But to extract how it would work. Well in that case, your requirement is completely unrelated to what is called as an objective analysis. An objective analysis can be done in a black box method and testing if the output is as claimed, the black box need not be fully known. It is a functionality testing method, doesn't need to know how it is accomplished.

 

I've tested it for my use case and it works convincingly for me. If you wanted an objective assessment, I have shown you the method to do it as well.

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3 minutes ago, March Audio said:

I have subjectively tested it and my experience is that it doesnt improve the sound.  You have a contradictory experience.  Therefore its perfectly reasonable to objectively look further and, amongst other things, analyse process of operation and assess reasons for differences in conclusions.

Then why are you merely ranting instead of just actually trying to assess it the proper way. 😅 I am not the one who asked for proof or started objective fi thread. But I do know if a personal rant is disguised as an objective thread 😬.

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4 minutes ago, March Audio said:

There is no ranting.  Its a simple situation:

 

  • I tried the software after you promoted it.
  • I heard no improvement in sound.
  • I currently do not understand any mechanism by which it could improve the sound
  • So yep, im sceptical but open to ideas.
  • I offer observations and opinions on why I dont see how it could work
  • In search of alternative ideas I ask for input from others who might be able to explain how it works
  • Still waiting.............

1. Suggesting something and promoting something are two different things. (They may sometimes overlap but it's not the case here). One more episode of your narrative skewing.

 

2. Good for you. You can happily skip this and move on to other stuff that interests you.

 

3. If you're truly curious you'll try to explore more into computer audio. It's a vast domain, so I can't say anything concisely. Everything from changing buffer to changing audio libraries, to changing Os resulted in audible changes for me (sometimes even measurable, but not always).

 

4. I'm skeptical about your openness considering your whole saga has been to derail any discussion on this topic, and successfully closed one topic too! Even this thread and your comments on my original thread often begin with derogatory remarks from an incomplete assessment.

 

5. If a subset of tests can't show difference, it doesn't mean there is no difference.

 

6. Inputs were given in other threads which you conveniently ignored. You are just looking for opinions that try to reinforce your prejudice.

 

7. I hope you get your answer eventually.

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35 minutes ago, March Audio said:

 

I can read.  It didnt answer the question.  How is that relevant to the audio file?  Doing what to the audio file? 

 

Also, the author claims you can keep on improving the sound by repeating optimisation over and over again.  Are you suggesting that you can keep on infinitely reducing CPU load by repeatedly changing low level machine code?  That would be a neat trick.

The file optimizer likely has to do with access noise during the file access. Every bit is stored as a set of charges in a cell (typically a floating gate nand cell), and the scenario in which the write action happens can likely manifest in differences in the structure of charges and magnetic fields stored in the cell that the next access after optimization may have either lesser noise or lesser correlated noise.

 

Digital circuits work just with thresholds. Above a certain threshold it is 1, below it it is 0 (or vice versa in some implementations), and there are boundary conditions which the designers have to work hard to ensure data integrity is maintained. This is the reason why you don't magically get infinite clock speeds. There's more to it in modern devices (they are multi, triple layer cells etc) and there's a lot of algorithmic stuff that goes on to it. There's a lot of hardwork in making a reliable working digital system, but it's even harder when you get into analog systems.

 

Now the problem with analog/mixed signal systems though is that it's not merely working on thresholds. A fair amount of noise may be mostly harmless in a digital system but will cause significant issues with an analog/mixed signal systems as every single flaw/deviation will cause deviations in the analog circuit (the dacs) and later get amplified in the buffer and amplification stages.

 

So any of the activity you do has a potential manifestation in the analog circuit, and any task that reduces noise at source can be beneficial. You can claim optical isolation but it is more fairytale than reality. They have their own jitter and noise footprints and any attempt to correct it will have its own jitter and noise footprints. The circuits can always be tweaked to fake numbers to specific scenarios while not being truly capable in other scenarios, and hence measurement charts get unreliable.

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12 minutes ago, March Audio said:

OK,

 

First question. Are you suggesting this player operates without any memory buffering?  That it reads continuously and therefore increadibly slowly off the disk?  So that this alleged noise is continuous during playback?

 

Second question.  Have you any objective evidence of this noise mechanism causing problems?

 

I cannot make specific remarks on what this software does unfortunately. I haven't explored the code yet.

 

The above post is generic to whole of mixed signal systems and yes many of these issues are well known (search ground plane noises, working of optocouplers etc).

 

12 minutes ago, March Audio said:

Sorry to quote my own post but the edit time has expired.

 

Third question here.  The author also claims that the process is "sensitive to the electromagnetic environment".  Can you explain how the EM environment affects machine code?

I haven't claimed electromagnetic spectrum in the environment affects machine code as data, but the the electromagnetic spectrum does induce differing noise on anything that can act as an antenna (including fingers and skin), and pcb lines and transistors would be no exception. Again if your entire process is going to be fully in digital domain, just passing the cut offs would be enough, but for analog and mixed signal systems, may not be, especially if you're looking for very high fidelity.

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1 hour ago, March Audio said:
  • You heard an improvement in sound.  Good for you. perhaps you can move on instead of opening another thread to promote it?  As an aside, how many others have come back in this forum to say they also hear an improvement in sound?
  • Im truly curious about THIS software.
  • It was you and PeterST that thread crapped with significant amounts of ad hominem for good measure. My questions were and still are perfectly legitimate and reasonable.
  • There were no cogent explanations provided in the other thread.
  • Well it appears that answers arent exactly forthcoming from those who claim to know how it works.

1. I've had enough friends who have reported me that it improved sq for them. Give it some time and you might see similar results in the forums as well. Not everyone is free all the time.

 

2. Nice. I hope you get enlightened.

 

3. I'm sorry, if you look at the first post, you didn't start off normally but you made a remark about me, then made assertions with your "analysis" which I've shown to be incorrect.

 

4. I'm not responsible for your inability to comprehend.

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1 hour ago, idiot_savant said:

Righty-ho, please nobody attack me :)

 

I've kind of not really read this thread as it appears to be degenerating rapidly, but this *is* the objective-fi forum, so let's see what we can objectively ascertain, no?

 

1: So we have some playback software, that claims to benefit SQ on playback. Assertions have been made to both the positive and negative as to this, but as far as I can see, we don't have any measurements, making this a purely subjective claim

2: An Additional feature is this "optimise" feature, that can be run more than once, that creates improved copies of the original source material. These improvements are very fragile. Again, subjective claims to positive and negative, and the files have been nulled successfully ( although I'm not sure anyone's done a simple file compare or a CRC )

 

So, does the claimed improvement only work with the associated software player, or with all players? Is it PC specific? Can more than one OS be used?

 

Now, if the improvement only works on one player, it is feasible that the contents of a file could be restructured whilst maintaining the actual reconstructed audio, and this is in some way beneficial to that player - anyone who has ever attempted to write an audio file parser will understand this ( typically, there are a bunch of "chunks", which can contain audio, metadata, other guff ).

 

Actually, what file types can you optimise? WAVs? DSF? FLAC? All files?

 

If it is attempting to interact with the file system, does it require administrative rights?

 

your friendly neighbourhood idiot

 

 

 

 

 

I think you should be able to optimize any audio track (it even accepts non audio files!). I think it does ask for some admin rights (I saw a warning message when I run the optimizer). It is supposed to be paired with his own player but any player that does memory playback by directly fetching the file should do I guess (without making a temp copy in hard drive, which unfortunately is what foobar2000 ab-x does hence not being compatible). I had good luck with both Junilabs player and XXhighend when using the optimized files.

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51 minutes ago, Confused said:

I have a question.

 

For those that hear a difference, are you running a PC to DAC via USB?  

 

For my own set up, I use a PC, but this is then streamed via Ethernet to a network attached endpoint.  In addition to this, I do have a desktop setup that is PC to DAC, for headphones or desktop speakers.

 

Just thinking out loud a little, I can imagine that some kind of software optimisation might yield an improvement when running a PC via USB to DAC.  A reduction of noise in the PC, less noise making to the DAC.  With Ethernet streaming to an endpoint, I cannot see how this could be a factor.  (I might be wrong)

 

Keeping an open mind, I should be able to try this "both ways" at the weekend, to see if I can discern any audible improvements. 

 

 

Yes it was done through PC to DAC via USB but I don't see ethernet being any magical fix. At best, it might de-sensitize it to the improvements made upstream and replace with a noise and jitter of ethernet's own backend. Doesn't guarantee global best performance. I'd be curious to know your results both with USB and also with ethernet.

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8 minutes ago, March Audio said:

Ethernet solves the specific problems I noted earlier as it provides galvanic isolation.  Its built into all ethernet adaptors.

Nice fairytale. Now explain me how "galvanic isolation" fixes every single possible source of noise. There are two types I know of

 

1. Transformer based (if you peek closer into the actual EM waves equation and also actual physical structure of transformers, ie magnets used etc, you'll know it is not a guarantee of full isolation for all types of noise).

2. Optical - Well optical has problems of its own, jitter mainly and also most of them are low speed. High speed optocouplers have problems of their own and will need a heavy electronic backend which will cause it's own noise.

 

There's NO MAGICAL FIX. And fyi, the improvements were also heard on ifi micro iDSD black label. Go check out the isolation it has.

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1 minute ago, idiot_savant said:

@manueljenkin - right, so I downloaded the software, and the warning message you get is that it's not been digitally signed, rather than it requiring admin rights. I didn't install it, as there is a very good reason you shouldn't be installing unsigned software ;)

 

 

 

your friendly neighbourhood idiot

I never had that issue. Only when running it showed me a warning. There's only one way to test it though.

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30 minutes ago, March Audio said:

Can you provide your understanding of what happens to a file/audio data from being read from disk to ending up at the DAC?  Sort of step by step process.

Set up:

Connection of USB -> enumeration -> Basic USB handshake ...... (this is for usb 2.0, if it's usb 3.0 its way more complicated).

 

Playback:

Fetching of Music file from PC to RAM. Either done in small chunks through CPU access to RAM (or DMA in rare cases it is supported). These chunks can be small or big (buffer). There is noise associated with every cpu activity and every drive access, every RAM load.

 

The USB controller in DAC will send interrupt frames requesting data depending upon its buffer levels and CPU will need to send this data through USB port as usb cannot do DMA (fetch from RAM, send to PCH -> USB Root Hub -> USB port). In the above case CPU has to switch from accessing data from Drive and pushing to RAM, getting user control from mouse, keyboard etc, look into interrupt command by USB dac and any other task. The alternative would be to store entire music in RAM so that one access and non determinism is reduced.

 

USB asynchronous over isochronous works by buffering and USB dac clock will become the master for the port generally. It will track buffer and request when it needs data through the interrupt command. This buffered data is clocked and sent to the DAC using a dedicated logic. The polling period is 125us.

 

All works out in theory if you assume a fairytale scenario. But reality is more complicated. There's noise (due to lots of digital transactions ie signal switching) and lack of determinism everywhere, a digital signal would have the digital line and a ground line which serves as a reference and ground lines will definitely get polluted by switching noises from motherboard and also the data flowing in the data line (return currents). Every single component from the clocks, the buffering logic etc, can have their own noise that might get worse if the input is not clean enough, and then of course you have your ground. If you try to break the ground by using optocoupler you end up with jitter from optocoupler. High speed optocoupler is even harder to accomplish and have issues associated with themselves too. And measurements can always be cheated.

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8 minutes ago, idiot_savant said:

Right, so before we all get pernickety about proving how smart we are, how does optimising a file reduce this switching noise?

 

 

 

 

 

your friendly neighbourhood idiot

The structure of charges stored in the gates likely (you'll have to push charge through digital circuits and depending upon the scenario of ambient environment, it could change the structure). Even mechanical vibrations can sometimes cause instability in capacitors (and these charge storage elements are tiny capacitor like structures). That's my current guess.

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10 minutes ago, idiot_savant said:

I'm sorry?

There's a very fundamental problem with this theory, that I'll attempt to explain for anyone who cares, and it is this:

The application *cannot* have the level of access to the file that would be required to alter the kind of things you are talking abut. As you have said yourself, there is enormous non-determinism in a PC - there are typically 3 levels of CPU cache, plus a file cache run by the OS, and there are people who's entire jobs are dealing with caches. Because of this, you simply *cannot* guarantee with any certainty when anything is read or written to. 

The whole function of an OS is to provide "abstraction" from the hardware - so that an application doesn't care if the file is on a USB stick, HDD, SSD etc. - or even what brand they are and the OS will actively block any efforts by something attempting to gain lower level access.

You're also missing out several layers of RAM access between the original file and the USB interface ( so the playback software will almost certainly use the OS audio stack, which talks to the USB audio driver, etc etc )

 

You also have to consider how much *stuff* a CPU is doing when you're just looking at a static screen, and any extraneous noise added by reading a few megabytes of data will be swamped in this. I honestly don't doubt that you perceive a difference, but I equally honesly believe that if things were as "fairytale" as you are stating, we wouldn't be having this discussion because the internet wouldn't work

 

your friendly neighbourhood idiot

 

I am aware of the massive amount of layers, buffers and Phys present through the chain. And of course software abstractions, and each abstraction layer = generally longer non optimal code = more processor and component activity = more switching noise, and of course there's more considering the speculative execution etc and these are accounted for with many of these audio software like xxhighend. They try to work at a lower level language with less abstractions, and hence lesser noise (one example is using kernel streaming). So the whole thing actually reinforces the benefits of a customized software system.

 

Yes it is phenomenal that the data storage access noise seems to pass through all these layers but if you consider the path, none of them have anything to compensate for the fluctuations, and as long as it is within thresholds of digital circuit operation it'll be passed through (but analog and mixed signal systems are picky). It indeed is profound that this distinct improvement is not buried within noise generated from the rest of the link.

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