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MAC vs PC -> DAC (Digital to Analog Conversion)


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SSE doesn’t disable Core Audio; any Core Audio processes continue to work in parallel. SSE writes to the HAL or Hardware Abstraction Layer, the same (end of the chain) API that Core Audio writes to. It adjusts the DAC setting, which results in a change displayed in Audio MIDI Setup.

 

Okay... This is the bit that confuses me: how can two things write direct to the HAL at the same time? Both Amarra and (for example) YouTube can be playing audio at once and both come out the same hardware device - yet you say Amarra bypasses Core Audio mixer?

 

Is Amarra not missing a "Hog mode" ala Pure Music?

 

Eloise

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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I think it is designed to allow that so that you can be interrupted by a mail arrival sound while listening to music (for example).

 

But you can also do the reverse.

 

I have an HDMI out and optical out. HDMI goes to my TV and soundbar, and optical goes to my DAC and stereo speakers.

 

I can set the default sound output to HDMI and play iTunes sound (movie, music, etc) through the TV speakers and soundbar, and simultaneously use Play.app to play something completely different through my speakers. All within the confines of the core audio.

 

 

 

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I think it is designed to allow that so that you can be interrupted by a mail arrival sound while listening to music (for example).

 

Surely in this case, iTunes and Mail (for examples) are 2 inputs to the Core Audio mixer not writing direct to the HAL: or are you saying there is no mixer in the same way there is in Windows?

 

Eloisr

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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Eloise wrote:

This is the bit that confuses me: how can two things write direct to the HAL at the same time? Both Amarra and (for example) YouTube can be playing audio at once and both come out the same hardware device…

 

In our meat world, things appear to happen simultaneously. Yet, in any operating system, data transfer request are buffered up and time domain multiplexed (chopped up in time, then interleaved) based on the order they are received, their priority, etc. The result is two or more audio sources playing “at once.”

 

Is Amarra not missing a "Hog mode" ala Pure Music?

 

Rather than “hogging,” SSE tries to play nice with the other children.

 

Surely in this case, iTunes and Mail (for examples) are 2 inputs to the Core Audio mixer not writing direct to the HAL: or are you saying there is no mixer in the same way there is in Windows?

 

Core Audio does have a panner/mixer as well as other fun stuff like a generic HRTF for spatialization and a well implemented plug-in architecture, AU.

 

BTW wgscott, love that marsupial avatar!

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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In our meat world, things appear to happen simultaneously. Yet, in any operating system, data transfer request are buffered up and time domain multiplexed (chopped up in time, then interleaved) based on the order they are received, their priority, etc. The result is two or more audio sources playing “at once.”

 

Yes I understand computers don't do anything simultaneously, it's the illusion created by things happening in little time slices however: ... is what you are saying that when another application (inmy example Safari) is trying to play music simultaneously with Amarra that you get...

Amarra sample1:Core audio sample1:A.s2:Ca.s2:A.s3:Ca.s3:A.s4:Ca.s4:...

well you get the idea... output from the audio device? Or as it would have to be

Amarra.sample1:CoreAudio.sample2:A.s3:Ca.s4:A.s5:Ca.s6

as otherwise you'd be halving or doubling (can't work out which) the playback sample rate. I think I was happier with the idea of Amarra and Core Audio feeding two audio streams into a mixer.

 

The more I think of this the more that exclusive use mode actually is better idea! Why do you want SSE to "play nice" and be able to mix in with Safari sounds, etc.?

 

Can I also ask - what association do you have with Sonic Studio and Amarra?

 

Eloise

 

PS. I'm really trying to understand this ... but I can't see how two audio streams can reliably write to the HAL directly. You need a mixer (in my mind). It's not like connecting 2 audio sourses to a single analogue input!

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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Eloise,

 

In my experience using HOG mode and bypassing the mixer and using fixed instead of floating point is a huge upgrade.

 

Just bypassing the mixer alone would be a huge step forward. Sure you specify your internal speakers as your default core audio output and specify your high end dac via the Audio Application. But you are still going through the layers.

 

The more layers the more processing time the less quality. That has been proven.

 

If you look at the processes in the terminal application (don't bother with the damn activity monitor it tells you nothing). You can see a drastic reduction in overall processing time and application processing time when using HOG mode.

 

Thanks

Gordon

 

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In my experience using HOG mode and bypassing the mixer and using fixed instead of floating point is a huge upgrade.

 

Just bypassing the mixer alone would be a huge step forward. Sure you specify your internal speakers as your default core audio output and specify your high end dac via the Audio Application. But you are still going through the layers.

 

I think I'm agreeing with you ... that bypassing the mixer is good. What I'm saying is that I can't see how Amarra can be bypassing the mixer as you can be playing a file from Amarra and also playing another from Spotify at the same time and they are perfectly mixed (same as if you are playing iTunes and Amarra at the same time).

 

Am I missing something - I haven't found how to specify a different device for output of Amarra (unlike Pure Music).

 

Eloise

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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You wrote:

Yes I understand computers don't do anything simultaneously, it's the illusion created by things happening in little time slices however: ... is what you are saying that when another application (inmy example Safari) is trying to play music simultaneously with Amarra that you get...

Amarra sample1:Core audio sample1:A.s2:Ca.s2:A.s3:Ca.s3:A.s4:Ca.s4:...

well you get the idea... output from the audio device?

 

Hey Eloise,

 

Software writes to the HAL and HAL takes care of the gory details. That’s what abstraction layers do for a living. They behave as an idealized “real” version would so, in this case, the HAL would behave as actual audio hardware would. All modern OSs abstract audio and graphics subsystems for lots of good reasons. The gory details of how Mac OS’ HAL actually processes incoming data is probably available from Apple's developer documentation, or not.

 

The more I think of this the more that exclusive use mode actually is better idea! Why do you want SSE to "play nice" and be able to mix in with Safari sounds, etc.?

 

Sorry to answer with a question but: why not, especially if it gets the job done without “breaking” other services, like user alerts?

 

Can I also ask - what association do you have with Sonic Studio and Amarra?

 

Sure, I’m a consultant and Sonic Studio is one of my clients. I work with several audio manufacturers, you can read way too much about my company on my site.

 

PS. I'm really trying to understand this ... but I can't see how two audio streams can reliably write to the HAL directly. You need a mixer (in my mind). It's not like connecting 2 audio sourses to a single analogue input!

 

Definitely not like analog…My guess is that, inside the HAL there is simple arithmetic summing of all inputs, in floats, which would be the equivalent of the most basic of mixers. The result of that summation, after further processing, is what is presented to the designated output at each clock cycle.

 

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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Hi Oliver,

 

Definitely not like analog…My guess is that, inside the HAL there is simple arithmetic summing of all inputs, in floats, which would be the equivalent of the most basic of mixers. The result of that summation, after further processing, is what is presented to the designated output at each clock cycle.

 

How you that work if sample rates are different ?

 

Elp

 

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My guess is that, inside the HAL there is simple arithmetic summing of all inputs, in floats, which would be the equivalent of the most basic of mixers. The result of that summation, after further processing, is what is presented to the designated output at each clock cycle.

 

So at the end of the day there IS a mixer inside the HAL is what you're saying!

 

Elp asked about different sample rates - I guess this is why the Core Audio interface tracks the sample rate, so that that converts everything to whatever rate the music being played is at.

 

Eloise

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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My take on this is that - where does Core Audio begin and end? Somehow the HAL layer mixer is "better" than one in Core Audio?

 

If the higher - bit-depth processing is important during the playback chain, it's then OK for the "HAL" to mix two audio streams together using (presumably) the same level of arithmetic as Core Audio?

 

If it's a mixer, it's doing some processing. Ergo, the claims about Amarra bypassing the entire audio stack are weak at best.

 

NB I'm not saying there's anything wrong with Amarra in and of itself, but claims that it bypasses Core Audio completely wouldn't seem to stand logical analysis...

 

your friendly neighbourhood idiot

 

 

 

 

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I'm glad I'm not the only idiot who thinks that doing mixing = processing /= bypassing Core Audio completely.

 

I guess it depends on if your definition of "Core Audio" includes the HAL.

 

Eloise

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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You said:

 

My take on this is that - where does Core Audio begin and end?

 

Hey iSavant,

 

My take is that Core Audio ends at any and all of the transmitters designated as audio outputs.

 

Somehow the HAL layer mixer is "better" than one in Core Audio?

 

An summing node, whether analog or digital, is not a “mixer” by most people’s definition. Mixers usually have gain staging and panning, at the very least and, by modern standards, that a pretty lame mixer. So yes, I would say the accumulator in the HAL is better for what we all are trying to accomplish: better SQ.

 

If the higher - bit-depth processing is important during the playback chain, it's then OK for the "HAL" to mix two audio streams together using (presumably) the same level of arithmetic as Core Audio?

 

Simply put, no. The Core Audio mixer is a “real” mixer, with gain, pans, routing and redithering and Apple only knows what else. Complex convolving, like plug-in processing, is probably done both before and after the mixer, since it depends on where the processing has been instantiated. So, audio passing through normal Core Audio paths must pass through a very complex “process,” in the true sense. Even though some coefficients may be set to unity, others are not, and they all act as operators on the audio data. And, this is floating point not fixed point arithmetic, which means the low order bits (the low amplitude data) are sacrificed, again and again, on the altar of gain staging.

 

If it's a mixer, it's doing some processing. Ergo, the claims about Amarra bypassing the entire audio stack are weak at best.

 

Never said SSE bypasses Core Audio entirely. If that were the case, you’d never get audio out of the computer! It bypasses everything right up to the output spigot.

 

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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There are many on here who claim to have bit perfect output. However, 99% of those are unable to absolutely confirm their claim. Quite sophisticated test equipment is required for that and I'm sure that most of us do not possess it. While we would all like to have the best possible input into our Dacs, its really a case of trying different options/drivers etc until you find one that suits.

 

MacMini 8Gb OSX > Pure Music / Bitperfect / Amarra / iTunes > Synology DS215J NAS > Schiit Wyrd > Stello U3 > Naim Uniti Atom, Harbeth P3ESR. Meier Corda Arietta Headphone Amp > Sennhieser HD650 Phones (Cardas rewire). Isol-8 Powerline Axis. Isotek GII Orion Power Conditioner. Cardas Clear USB Cable. Tellurium Q Black Speaker Cable. All other cables by Mark Grant.

Vinyl still has it's place. Technics SL1200. Modified with Mike New Bearing, KAB Strobe Disable, MCRU 2 box PSU, Isonoe Feet, SME M2-9 Tonearm > Goldring 2400 >Rothwell Simplex Phonostage.

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It's important to bypass the "Core Audio" mixer because we don't know what level of processing it does, but it's OK to use another part of Core Audio to mix together the two streams, even though we don't know what level of processing that does?

 

Any kind of mixer is a multiply. This must have an arithmetic width, and will result in the output increasing in wordlength, so again, some kind of dithering/truncation is probably required at the output of the mixer.

 

Additionally, if you bear in mind the fact that iTunes has been proven on here to stick the same bits out of the audio output are as in the file, and Amarra/SSE puts out the same bits, there can be no arithmetical advantage to SSE, unless you are using some of the processing (i.e. gain, EQ) - which I would agree SSE should do better than Core Audio.

 

As a side note, if Amarra/SSE is super low-latency, why is it that when you increase the iTunes volume, the iTunes version comes out before the Amarra one? NB I don't think latency makes any difference for playback...

 

 

your friendly neighbourhood idiot

 

 

 

 

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"since I have the best playback for realistic sound reproduction"

Peter, can you share with us the setup of hardware and software that allows that claim..?

Peter would be referring to his own software and own designed DAC...

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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iSavant said:

 

So...

It's important to bypass the "Core Audio" mixer because we don't know what level of processing it does, but it's OK to use another part of Core Audio to mix together the two streams, even though we don't know what level of processing that does?

 

That’s not what I said.

 

Any kind of mixer is a multiply. This must have an arithmetic width, and will result in the output increasing in wordlength, so again, some kind of dithering/truncation is probably required at the output of the mixer.

 

Ah hah, you said it. Even the most basic real world mixers need to have, at the very least, additions (the actual summing operation) and multiplies (level adjustment for each input then post-summing gain normalization), re-dithering (another multiply) and then truncation. And that’s not even considering the fancy stuff. In comparison, a summing node is just a summing node. Arithmetically, a significant difference.

 

Look, Windows, Mac OS and other modern OSs all employ hardware abstraction, and not just for audio. All I/O is abstracted. So, unless you’re anti–computer music, you live with hardware abstraction every time you listen, regardless of who makes your favorite widgets and apps.

 

Additionally, if you bear in mind the fact that iTunes has been proven on here to stick the same bits out of the audio output are as in the file, and Amarra/SSE puts out the same bits, there can be no arithmetical advantage to SSE, unless you are using some of the processing (i.e. gain, EQ) - which I would agree SSE should do better than Core Audio.

 

Though you won't hear any manufacturer provide details as to how exactly they manage to get improved SQ from their product, bit perfect output is assumed as an entry level feature in any of the dozens of music playback apps out there. If the above statement were true, why is it that you, I or anyone can head to a dealer or, if their system and auditory faculties are up to it, run any of these apps at home and hear a difference when compared to Apple’s or Microsoft’s default playback? (Have you tried Amarra's bypass button?) It's not, we assume, because Sonic Studio, Channel D or any other manufacturer who states bit perfect operation is lying. It’s because we’ve found better ways to handle audio data that the OS defaults which are, as I’ve mentioned elsewhere in this forum, general purpose systems, designed and built for any and all eventualities.

 

As a side note, if Amarra/SSE is super low-latency, why is it that when you increase the iTunes volume, the iTunes version comes out before the Amarra one? NB I don't think latency makes any difference for playback...

 

Simple: the interapplication communication between iTunes and Amarra, which has nothing to do with audio BTW, is relatively slow and of lower priority. In comparison, iTunes is always given a very high priority so its requests are always serviced very quickly.

 

Regards,

______________________________________________

O.A. Masciarotte - http://www.othermunday.com

______________________________________________

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Oliver ... when I questioned the lack of Hog mode, you commented that Amarra didn't need it because it wrote to the HAL ... but surely the advantage of the Hog mode in Pure Music is that it avoids any need for the mixer being utilised - therefore there IS a need for a Hog more as you (I think) and Gordon have both said that it's best to avoid mixers which (with Amarra) can't in some case be avoided. Likewise in Windows applications where they use WASAPI exclusive mode.

 

Eloise

 

Eloise

---

...in my opinion / experience...

While I agree "Everything may matter" working out what actually affects the sound is a trickier thing.

And I agree "Trust your ears" but equally don't allow them to fool you - trust them with a bit of skepticism.

keep your mind open... But mind your brain doesn't fall out.

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The 'quite sophisticated equipment' could be an AV reciever with DTS decoder. Play a DTS track through the digital output of the computer to the reciever.

 

? MBP ? M2Tech hiFace ? Heed Q-PSU/Dactilus 2 ? Heed CanAmp ? Sennheiser HD650

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Mike,

 

Peter, can you share with us the setup of hardware and software that allows that claim..?

 

I can do that allright, but please notice you should look at it as a "concept". The very explicit concept of

 

1. Pass on the data as much 1:1 as possible;

2. Use speed as one of the means to get there.

 

Since audio is "vague" and almost always a subjective thing to the users of it, I thought to better hunt for a discrete "something", which for me became that 1:1 passthrough of everything everywhere in the chain.

 

This is very contradictionairy, because where our today's chain start at the digital part of the end result of recording (be it redbook or higher res material), it is not *allowed* to pass that on 1:1. We'd have high levels of harmonic distortion when doing so, thus the first step is to approach this as best as can be. This means throw out all of the existing rules, and create some of your own.

 

The first part (but I skip ripping) we run into is the playback software. And yes, it should spit out the bytes as they were read from disk. Although this is subject to the last posts in this thread (about hogging the mixers etc. etc.) this isn't all that difficult as long as you are a(n audio) software developer and know where to get what.

 

Once that's done, the bytes have to be passed on to the D/A converter, and sadly this traject doesn't imply that the bytes are interpreted as intended. It is subject to jitter;

When all has been done in the DAC itself to avoid / eliminate as much jitter as possible, there's still inherent jitter in the DAC itself (I refer to the whole cabinet now), and that jitter is influenced by the software at the other end. It just is, and the verdict on the "exactly how" is still out.

 

So, still being in the software "layer", it is best if that software influences the (jitter) behaviour of the DAC in a positive sense.

This is my personal take on the software part, and I am the only one doing this. I mean, with "control" and sliders for it (knobs these days :-).

This may be 10-15% of the whole 1:1 job, but this 10-15% is in a rather important area of the listening pleasure. Also it has been proven that it works on any system/DAC/etc.; Part of the proof is that the results are consistent over users.

 

Next in the chain is our DAC;

It is here where so many options exist, that chances are 100% that a wrong combination is chosen. There's the whole clocking thing, the input receiver, the DAC chips and types used, the I/V conversion (in-chip or not), the gain stage, and ALL influence eachother so much, that it's just a Gordian Knot to solve when you want to do it right. And then I didn't mention PSU design, PCB layouts which may give you noise at -100dB or over -140dB.

Luckily, if you follow the 1:1 route, a lot of combinations fall out, and measuring becomes a great help.

 

It is here though, where the 1:1 (at least for redbook) is not allowed to apply, so it is here where the chances are. It is also here where it is done wrongly without exception, and thus it is here where all can be gained once you know how to do it right.

Long story short on this part : it can be done (because I have), but it "requires" an overall design where *none* of the parts mentioned are common, or even ever used before.

And mind you, "none of them" is rather drastic, where "one" improves significantly already.

 

Although theoretically such a DAC doesn't need special software to feed it (the music data), in this case it does because of the lack of the needed support by the OS or hardware parts which just don't exist. So, what's lacking on the general side, was filled in by the (player) software side of things. Part of this is the filtering dealing with the so important 1:1 strategy, while another part is about the sheer needed input sample rate to avoid harmonic distortion *when* this is combined with the special filtering (notice that "filtering" uses upsampling as a vehicle).

 

Once we are at the output side of the DAC, bad life may start, because we're now outputting transients unheard of. So, this happens as a result of the 1:1 strategy, which carries as one of its properties that transients are maintained. Notice though that a large deal of this happened at the analogue stage of the DAC (with all kinds of implications on the current surge, capacitors which must be able to follow, slew rates of (active) devices used) ... which now apply the same to the following parts in the chain.

 

The first next part is the amplifier, and certainly not any means of preamplifier or other attenuation means. The first *that* does it killing those transients again, assumed that whatever it will be, will be in the signal path.

 

Now we have to go back to the playback software again, because it will be there where the attenuation must happen (according to my ideas that is), and there's only one 100% legitimate digital attenuation scheme - and this is in my software. It is lossless, which firstly means that out of an attenuated stream the original can be recreated, but thinking somewhat further it implies "bit perfectness" for that area, because no single byte (bit if you want) is inconsistent with the others - hence stays for the inter relations as how it was in the original stream. Only the output volume is lower.

 

Still there ?

 

What we need next is a fast amplifier which is able to follow those transients and what easily comes down to the nessecity to follow a 30V jump within the time of one (redbook) sample.

 

While specs of amplifiers may show you the capabilities in this area, it is not to underestimate what happens if they do "just not" follow what is needed here. Firstly a voltage peak may arrive too late, while before it's at its peak it has to go down already. This is not just square wave behaviour (for measurement), but incorporates the necessary pre-swing, the overshoot and anything that takes time BUT is depended on what happened before. So, a 30V swing may be possible, but maybe not when a -30V swing just preceeded it ...

 

When the amps are fast enough to follow all this, we'll have a problem in the last part of our chain : the loudspeakers. If they can't follow our preciously pre-cooked data, it will be extra-wrong. In that case we better had 2000K interlinks to filter (oh yes) everything in the first place. Or a somewhat more fluffy amp. Or a nice preamp. Or a lousy DAC. Or a flattening player.

 

And so the speakers now have to be ultra fast, and I think this can only be done when they are the most sensitive, hence don't need much to move (their driver's diaphragms).

Here too (out of anything btw) a pre-swing applies before a frequency fully develops. The faster this is, the more the peak (excursion) will be there before it has to distract.

 

Well, if you are only with me that this is a kind of different approach then, say, tubes which nicely distort when they distort, you'll also see that such a comparison with SS is out of order, because nothing should distort in the first place. And nothing = nothing, and stating that some can't be avoided is out of the question. But -as in my case- it may take you 5 very explicit years to get there. And then to think that indeed the software is my own, the DAC is my own, the amps where under my close watching development, as were the speakers for a large part. And mind you, this is all up to the smallest resistor, the DAC being the most difficult, because all is amplified hugely from there.

 

It is all about quite undoable stuff; If I only mention the digital attenuation - thus the amps at full gain all the time - while the combination with the high efficiency speakers just *will* exhibit a blast of noise when it would be at -100dB ... that part alone ... go look for it. Go look for complete silence and see how odd your chances are.

Of course, it is me myself who put the requirements in the first place, but in the end it all can be done.

 

I don't think it is necessary to lay out what I exactly use for amps and speakers, although it is the least of the secrets. Nobody is going to buy that anyway. It is about the principles though, and the necessary knowledge of what goes on and what implies which. In the end it is about data sheets and looking at specs and whether they comply;

No such thing like "an OpAmp measures good but sounds bad" exists. No such think like "the best measuring amp is not the best sounding one" exists. You'd have to understand the implications of that well measuring amp though. It may drive your not being able to follow speakers crazy. Or, it may be able to follow the harmonic distortion it is fed with in the first place.

 

Lastly, there's a whole debate on "how to measure" underlaying. I won't start explaining the details again, but will mention that it is totally useless to follow THD specs on a DAC chip (or DAC) when it includes the filtering. It would be the first mistake to make (following my concept of approach), and we must take it that these figures are useless. Sadly, I know of no other means than understanding the data sheets and looking at the real merits of things, or measure yourself while knowing how to do it (which is out of the question for any normal consumer).

 

Allright. I hope this has been a more useful answer than plainly mentioning a few brands.

Peter

 

Lush^3-e      Lush^2      Blaxius^2.5      Ethernet^3     HDMI^2     XLR^2

XXHighEnd (developer)

Phasure NOS1 24/768 Async USB DAC (manufacturer)

Phasure Mach III Audio PC with Linear PSU (manufacturer)

Orelino & Orelo MKII Speakers (designer/supplier)

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I've heard many 'high-end' systems in the 35-odd years that I've been in this hobby (I started pretty young!), and I've owned systems that most people would consider pretty 'high-end'. But let me say that Peter's system is the most 'true-sounding' system I've ever heard, and certainly beats any system that I've ever put together. (I think his great room plays a big part in this though.)

 

I heard Peter's system ~6 months ago and it looks like he's made some improvements since then.

 

Just my 2c.

 

Mani.

 

Main: SOtM sMS-200 -> Okto dac8PRO -> 6x Neurochrome 286 mono amps -> Tune Audio Anima horns + 2x Rotel RB-1590 amps -> 4 subs

Home Office: SOtM sMS-200 -> MOTU UltraLite-mk5 -> 6x Neurochrome 286 mono amps -> Impulse H2 speakers

Vinyl: Technics SP10 / London (Decca) Reference -> Trafomatic Luna -> RME ADI-2 Pro

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