AK4499EQ - Best DAC ever

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Hi Bill,
Maybe I wasn't clear enough about what I thought would be obvious from the AD graphic. Here is the way I see it: Most modern ADCs digitize into something called Raw mode, its a few bits at a high sample rate. From there it can be converted internally or externally into PCM, DSD, or whatever.
so? this is a DAC thread.
What I think matters most in the context of the present discussion has to do with the quirks of dacs. Some sound better when operated in DSD256 (or higher) mode rather than they do in PCM mode at any sample rate.
FOR THE SAME INPUT DATA. whether you feed in PCM or DSD the master recording was the same. It's just been processed, which is exactly what the DAC onboard DSP does. So I repeat, you PREFER an outboard filter. You cannot say on a sample of one old pair of ears it sounds better.
That's the only reason I suggest conversion of PCM to DSD256 or higher. If you don't hear it then you don't hear it. Fine. Some people do hear it and they aren't all hallucinating. Similarly, some people using some dacs can hear the difference between high res and CD quality. If you don't hear it, fine. Doesn't mean no one else can hear it. Again, this is the way I see it, my opinion on it for now. Also, my opinions are subject to change as new information becomes available. Please have a nice day.
SO you actually mean you prefer outboard upsampling to what is in the chipset as the end result fed to the modulator will be some form of single or low bit high rate signal.
 
So, I agree with much of what you say. I agree that up to the point of passing through the dac as long as the DSP is done well the underlying information should be unchanged.

Where we either disagree or I have still not been clear enough has to do with the tradeoffs between DSD dac operation and PCM dac operation. To wit, DSD is more linear but is more sensitive to jitter. PCM suffers from element matching problems. For reasons such as those DSD can sound different in some ways from PCM. IME mostly at higher frequencies. In that regard DSD can sound smoother and less grainy.

Regarding audibility in general, please consider that with the hysteresis distortion Bruno and Lars described, only a few customers noticed the effect. It was audible to one or more people, but not all people. Probably not most people. If that can be true, then why can't it be equally true that some people can hear dac filters, DSD versus PCM, etc.

To look at audibility from yet another perspective, it has been claimed that something referred to as 'expectation bias' applies to audio perception. It there is such a bias, then it may affect people who expect to hear no difference (i.e. they hear exactly what they expect). Maybe that applies to you, or do you maintain that is impossible?
 
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Regarding audibility in general, please consider that with the hysteresis distortion Bruno and Lars described, only a few customers noticed the effect. It was audible to one or more people, but not all people. Probably not most people. If that can be true, then why can't it be equally true that some people can hear dac filters, DSD versus PCM, etc.

To look at audibility from yet another perspective, it has been claimed that something referred to as 'expectation bias' applies to audio perception. It there is such a bias, then it may affect people who expect to hear no difference (i.e. they hear exactly what they expect). Maybe that applies to you, or do you maintain that is impossible?
Try to stay focused. Nobody but you has mentioned audibility in this thread recently.

I have a suggestion. I'm willing to agree that someone may be able to hear whatever you claim provided that you stop bringing this stuff up in every possible thread. I bet there are many others willing to join me.
 
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Where we either disagree or I have still not been clear enough has to do with the tradeoffs between DSD dac operation and PCM dac operation. To wit, DSD is more linear but is more sensitive to jitter. PCM suffers from element matching problems. For reasons such as those DSD can sound different in some ways from PCM. IME mostly at higher frequencies. In that regard DSD can sound smoother and less grainy.
And this is where I raise the BS flag. The modulator gets a 1-7 bit signal whatever comes in if it's a SD converter. Linearity has nothing to do with it. BUT your fancy pants HQ resampler will give a different output to the onboard chip as can be seen from the impulse response. as for higher frequencies, you've admitted 5 octaves are lost to you...
Regarding audibility in general, please consider that with the hysteresis distortion Bruno and Lars described, only a few customers noticed the effect. It was audible to one or more people, but not all people. Probably not most people. If that can be true, then why can't it be equally true that some people can hear dac filters, DSD versus PCM, etc.
It was anecdote. good marketing anecdote...
To look at audibility from yet another perspective, it has been claimed that something referred to as 'expectation bias' applies to audio perception. It there is such a bias, then it may affect people who expect to hear no difference (i.e. they hear exactly what they expect). Maybe that applies to you, or do you maintain that is impossible?
I just want a plausible theory, which yours isn't for a SD converter.
 
Try to stay focused. Nobody but you has mentioned audibility in this thread recently.

I have a suggestion. I'm willing to agree that someone may be able to hear whatever you claim provided that you stop bringing this stuff up in every possible thread. I bet there are many others willing to join me.
I was talking to Bill, and to me at least, Bill said something earlier about not 'believing' in dac filters. Seemed to me he was referring to whether or not they were actually audible. If so, I didn't bring up the issue of audiblity first.

Regarding the discussion about dac design, fedde was asking about building a dac equal to or better than OktoDAC at a cost of less than 2,000EU. In response to his request for discussion, I offered some advice based on experience.

Later Mr. tone? asked why I said I personally would not buy Marantz SA-10 because I didn't like the sound all that much. In response to further questioning by tone? I described some philosophical and practical issues having to do with voicing of some high end audio gear that was intended to explain enough about it to help it make some sense to Mr. tone? Again, I framed it as a personal opinion.

What I think people object to the most is when we start arguing. There have been complaints about that before from people not directly involved in the arguing.
 
Bill said something earlier about not 'believing' in dac filters. Seemed to me he was referring to whether or not they were actually audible. If so, I didn't bring up the issue of audiblity first.
Bill said: "Enough DACs offer switchable filters that you can adjust to taste if you believe in that stuff". That has nothing to do with general audibility of filters.

Many people (me included) object to you filling every possible thread with this same stuff.
 
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Many people (me included) object to you filling every possible thread with this same stuff.
There are many 'possible' threads in the digital line forum. I think you will find that I don't participate much or at all in a number of them.

Mostly it seems to be the same few people who don't approve of my views. They sometimes rant, exaggerate, troll, etc.
For such people the forum has a feature intended to solve their problem. Its called an ignore list. That way you can cancel whoever you want.
 
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Christ people.

can you guys stop arguing for a second even?

ok have a different point of view and discuss.

it doesn’t help others here as myself in learning anything.

unless I shouldn’t be on here because I’m not an EE.

what is different about converting to DSD instead of using PCM in a DAC?
Marantz and Phillips aren’t morons, so I assume there is a reason why they did that


thanks in advance to anyone who can explain this to me
I’ve read something that converting to DSD first is more accurate and doesnt use as much math to ‘ guess’ frequencies or samples.
something like that
 
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I’ve read something that converting to DSD first is more accurate and doesnt use as much math to ‘ guess’ frequencies or samples.
something like that
With a band-limited signal, interpolated samples can be accurately predicted based on the surrounding samples, no guessing required. Especially if you realize the incoming samples are not bit-perfect, because, most likely, they have been dithered and probably already incurred several sample rate and format conversions in the mastering process.
 
can you guys stop arguing for a second even?
I sometimes get trolled, attacked, rude disagreement, exaggerations, rants, etc., whatever you might want to call it. Sometimes efforts to find some agreement fail. Other times there is no way to stop the arguing but to either out-debate someone or to go offline for 3-days or so to let other people feel like they won. When there are 3 people at once attacking sometimes the only solution is to take civil discussion to PM.

Speaking of civil discussion, I would be happy to continue. If nothing else PM and or private email can be available options.
 
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You guys just can’t regulate your emotions one bit can you?

it must be the personality of an introverted engineer or something.
some of you guys are super smart , but just can’t regulate your emotions or know social etiquette.

now you can continue to bicker to each other
 
For me, a great DAC needs to hold my attention, draw me into the artists' communication via the recording. It goes without saying that such a DAC will have zero listening fatigue.

As regards opamps and NFB, if you use opamps the NFB comes along for the ride. I doubt there's any practical way to use opamps open loop, they make jolly poor comparators.
 
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For me, a great DAC needs to hold my attention, draw me into the artists' communication via the recording. It goes without saying that such a DAC will have zero listening fatigue.

As regards opamps and NFB, if you use opamps the NFB comes along for the ride. I doubt there's any practical way to use opamps open loop, they make jolly poor comparators.

so the op amp has the amount of negative feedback baked in?

thanks