AAAAAAA

iTrader: (7)

^I think he might be refering to the fact that one only needs to make sure the FR is flat enough to make devices indistinguishable sonicaly. Parts might affect things such as reliability but a DAC has now become a tool for marketing and is no more about accurate (as far as the ear is concerned) reproduction. Non defective DAC's will have no sonic benefit or detriment.

In other words, there is no proof that multi bit DACs have any advantage over single bits as far as the human hearing is concerned.

dogbaker

As for what he's thinking, I will wait for his clarification,

As to your comment that all DAC's perform alike - they don't - not subjectively nor objectively.

In the same way as amp, preamp and speakers don't. If they did every sound system would sound the same if they were played in the same acoustic environment etc... And they clearly do not...

The differences between them can objectively be found in dissecting their individual qualities of transfer function - most paramount: signal Magnitude and phase.

There's a lot of free places on the net that help bring this into focus for you....

AAAAAAA

iTrader: (7)

^Simply find and post ONE article\research\test depicting people in a blind ABX test being able to pick out different receivers or amps or cd players that are leveled matched. Just find me one, should be simple thing to find if what you claim (as many have and do) is true. And then we can continue.

AAAAAAA

iTrader: (7)

From this

You can't be talking about an equal delay throughout the entier FR so.... what do you mean exactly? Explain this delay you speak of.

dogbaker

Start with the speaker kit and have a listen - if your not satified finish the door...

dogbaker

Not equal but different is what I am saying...

Most people believe that at the outputs of an amp, preamp, speaker surface etc that the frequencies are being emitted all at the same time – in other words, that an electro-mechanical or electronic device outputs: each and every frequency, at precisely the same time, which they don’t. They do so at deferent times, for each frequency, primarily because of impedance fluctuations caused by increases in frequency, within in purely electrical circuits (but not solely as we will learn – the digital domain also incurs delays). This in turn leads to a diminishment in stereo image reproductions, frequency and transient response - at the bare minimum!

So what devices are most troublesome?

No. 1: Speakers have the most errors up to 600ms at low frequencies - that’s over a half of a second
No. 2: Then preamps & amps are next in line with up to 10ms, at the bottom and top of the bandwidth
No. 3: Then Source units, when measured at their digital SPDIF stage or output – CD Players – iPods etc come in at up to 500 micro seconds in many entry level types (single bit), (up to 10Ms at analog outputs) and up to 500 nano seconds in most quality devices (multi-bit), (up to 3ms at analog outputs), some esoteric products are ultra fast, coming in with mere pico latencies (6K for one of these babies), (up to 3ms at analog outputs)...
No. 4: Then Cabling: Which is dependent on length and cable quality – in a car micro-nano latencies are common.

As a system, these sum and in fact, can create additional timing errors due to poor electrical compatibilities etc...

Digging in!

So let’s look at a source unit...I wish to keep this at a low level , so i will talk in broad terms and attempt to paint a picture of sorts...

I will attempt to reveal to you that there are in fact several time errors, which affect the quality and usability of the frequencies, outputted by electronic and electro-mechanical devices and in this case – so called digital devices...

This is a small part of the picture, but as time goes on – all aspects will be brought in.

Let's use a typical CD - Head-unit

It has a transport in which the CD is load into, from that point on, a laser assemble is guided across the CD emitting a measured wavelength of light. Diffractions from this light are collected as they reflect off the surface of the CD and are converted from light onto electrical pulse codes.

So let's pause here and review:

The disk is spinning, but not at a perfectly constant rate, light is pulsing at close to a perfect rate, but is traversing the gap at a constant speed. In that we have some mechanical elements integrated into our data acquisition, some very small but important time errors occur and they need to be compensated for, so a common approach is to incorporate 'buffers' which store/pause the data stream then release it to other stages (a little more on this in a bit).

These compensations and other processes take place after the photon pulses are converted to electron pulses. This conversation process introduces more time errors (as well as other distortions).

So at this juncture, we have converted photons pulses into electron pulses and we have a host of errors / distortions, but what’s primarily at the root of most of them (in the digital domain), is timing errors and of course low sample rate issues...

So let’s get back to our main flow...

Once conversion has taken place, buffering, oversampling and other digital process take place within and at various stages, but all in the digital domain, and all working towards one goal, the reproduction of a linear mathematical equivalent of an analog signal.

Now, this is where DAC’s topologies play a big part, as they each have pro’s and con’s, in terms of trades offs. Some have fewer stages which typically reduce distortions (Time and others), some have higher sample resolutions or faster thread poles, some require the implementation of analog filters post digital processing and the list goes on. As a rule though, Multi-bit topologies currently offer the best blend of compromise and overall value, in terms of dollars (yes – marketing has a say, unfortunately).

Regardless of DA topology, they all produce a myriad of distortions, with most of them rooted in sample resolution and processing time errors (but not all). The reason for this is that digital data – binary, is managed by transistors based processors, which can be viewed as electro -mechanical devices that are capable of turning on an off from a few times a second, to billions of times a second and onward (3.2 GHs CPU etc)... So in other words, it takes time to process or create digital data and that time leads to delays and in modern DAC’s they are processing many things at once and combining the results in a buffer, which in turns takes more time to organize the data, into packets (and makes errors while doing so, leading to more distortions). Once these packets are complete, they are ready to be converted into analog signals or to be transmitted as binary via a SPDIF or converted back to photon pulses and transmitted via a TOSlink output (which can add more errors).

So as you can see, there is a lot of room for error - All of the signal goodness is based primarily on sample rate – then processor rate and software/algorithm quality – and lastly circuit and board design.

I realize that I have left sample rate and circuit board element out of this overview until this point, but I am quite sure that I will be ask about them specifically, as time progress and I will address them at that time.

This is a general overview of the key contributing elements that have the most significant effect on phase response at the digital output stages, as such; the other elements are less important and have been left out off of focus within this discussion.

So let’s pause again and review:

DAC’s manage binary signals via transistors based CPU’s, which are in essences micro relays. These devices open and close controlling the flow of electrons etc. All this creates a bottle neck and takes time to process (hence the name processor). DAC’s process many layers of data at the same time and combine the results in buffers, which sort, compile and create the data packets for streaming to the final stage – digital to analog conversion or other. Most processors are running in the MHz range, which holds them down to at best, low-micro– high-nano scale frequency generation, some esoteric products use faster processors and more advances software algorithms, resulting in faster and more accurate frequency production.

Regardless of speed in the digital domain, they all must enter into a final stage of conversion from digital to analog and it is in this stage where in terms of phase they all incur shifts to a very similar degree, which is to say that most implement very similar analog circuits. Depending of what type of DA topology that the incorporated, they may also be required to additional analog filters to diminish some harsh effects of poor digital processing, which in turn can and does lead to significant phase shifts, centered around the filter frequency, increasing delays into the high micro – low milli second range, in some very entry level products.

In closing,

My point was and is that it has offend been argued that all CD Players/DAC’s should and do sound the same – but, that argument began in 1980 and has most been mostly abandon, as we have all experienced different tonal qualities (sonic signatures) from different DAC’s, as evidence by the fact that, year after year, as a myriad of audio companies release new and improved variants, we continue to trade-up. If you have the opportunity, listen to some older DAC’s within your existing system for a few hours - then put yours back in – I am highly confident that you will hear a difference – who cares if it’s an improvement or not – that would be subjective – but you should perceive a difference and I am sure you will, unless the audio system is very poor, as a whole.

If you’re unable or unwilling to attempt the aforementioned then at least ask yourself this: why if most of us have not perceived such, why have we all been gear junkies for the last 5-10-15-20-25-30 plus years... the answer, simple – if technology hasn’t in fact improved year after year, as objectively verifiable advancements in science and mathematics have claimed, then it can be for no other reason than, wait for it – we must be able to hear the difference...

So now ask yourself, where did that difference come from?

Now this is for a few readers specifically: the answer is NOT FROM OUTER SPACE, but from the group effort of Scientists and Engineers!

So what was once argued has passed because the mass major of people have in fact, been able to hear the improvements within each new variant of DAC’s (and other devices) and also that we have been able to start objectively measuring and therefore map more of the contributing: electrical, electro-mechanical and digital factors, which govern the transfer function of the data that is requisite to high quality audio reproduction.

What has been discovered is that a correlation exists that allows for the classification of DAC performance, based on their frequency phase responses. DAC’s which have mere pico latencies in the delay of frequency production (pre-conversion into analog signal) tend to produce sonic qualities closer to analog (post conversion). Furthermore, it has been discovered that delays, which occur and are greater in the mid range, but within the same scale, (pico, nano etc...) produce a sound that is laid back to most listeners. Most DAC’s produce latencies, (pre-conversion) in the micro ranges and have a poor grouping in latencies scores. In the land of electrons, micros seconds are miles and pico seconds are feet (i am not being literal)! Regardless, of the aforementioned digital qualities, the design and build quality of the analog stages are very important, and heavily influence the final outcome of the analog signal, which will next be presented to the preamp – and the story continues!

While we still don’t have the complete picture and may make a few mistakes, while we are still forced to permit some assumption(s), it has become clear that pico scale latencies are in fact, important to ultra high fidelity audio reproduction – not low!

As to the question, can we actual hear such delays – if we couldn’t, they wouldn’t be - becoming, an important consideration in ultra high fidelity audio design.

But let’s end with this – while this was in fact a low level overview, to some, it has been an esoteric one, which is to stay that it was meant for and understood by only a few...

In practical audio design, milli-seconds are all that matter and nothing more, and your head unit, cabling and even your amplification are not your biggest enemy(s), in this regards, it’s your loud speakers! So focus 99% of your energy on getting them to produce timely sound – for SQ or SPL systems, it’s all about phase!

d4rin

dogbaker - Most people say the TWEETER must be as close to the MIDBASE as possible in a 2way or 3way Component sets.

As of right now, I have Morel speakers and the mid is in the door, While the tweeter is in the 'A' pillar. Everything seems to blend well together and there is no ear/hearing fatigue of any kind.

So my question is, If the tweeter was actually closer to the midbase would there be a soundquality improvement. Because it sounds fine even though the tweeter is up higher.

rogue13

[QUOTE=dogbaker;628855]Not equal but different is what I am saying...

Most people believe that at the outputs of an amp, preamp, speaker surface etc that the frequencies are being emitted all at the sample time – in other words, that an electro-mechanical or electronic device outputs: each and every frequency, at precisely the same time, which they don’t. They do so at deferent times, for each frequency, primarily because of impedance fluctuations caused by increases in frequency, within in purely electrical circuits (but not solely as we will learn – the digital domain also incurs delays). This in turn leads to a diminishment in stereo image reproductions, frequency and transient response - at the bare minimum!

So what devices are most troublesome?

No. 1: Speakers have the most errors up to 600ms at low frequencies - that’s over a half of a second
No. 2: Then preamps & amps are next in line with up to 10ms, at the bottom and top of the bandwidth
No. 3: Then Source units, when measured at their digital SPDIF stage or output – CD Players – iPods etc come in at up to 500 micro seconds in many entry level types (single bit), (up to 10Ms at analog outputs) and up to 500 nano seconds in most quality devices (multi-bit), (up to 3ms at analog outputs), some esoteric products are ultra fast, coming in with mere pico latencies (6K for one of these babies), (up to 3ms at analog outputs)...
No. 4: Then Cabling: Which is dependent on length and cable quality – in a car micro-nano latencies are common.

As a system, these sum and in fact, can create additional timing errors due to poor electrical compatibilities etc...

Digging in!

So let’s look at a source unit...I wish to keep this at a low level , so i will talk in broad terms and attempt to paint a picture of sorts...

I will attempt to reveal to you that there are in fact several time errors, which affect the quality and usability of the frequencies, outputted by electronic and electro-mechanical devices and in this case – so called digital devices...

This is a small part of the picture, but as time goes on – all aspects will be brought in.

Let's use a typical CD - Head-unit

It has a transport in which the CD is load into, from that point on, a laser assemble is guided across the CD emitting a measured wavelength of light. Diffractions from this light are collected as they reflect off the surface of the CD and are converted from light onto electrical pulse codes.

So let's pause here and review:

The disk is spinning, but not at a perfectly constant rate, light is pulsing at close to a perfect rate, but is traversing the gap at a constant speed. In that we have some mechanical elements integrated into our data acquisition, some very small but important time errors occur and they need to be compensated for, so a common approach is to incorporate 'buffers' which store/pause the data stream then release it to other stages (a little more on this in a bit).

These compensations and other processes take place after the photon pulses are converted to electron pulses. This conversation process introduces more time errors (as well as other distortions).

So at this juncture, we have converted photons pulses into electron pulses and we have a host of errors / distortions, but what’s primarily at the root of most of them (in the digital domain), is timing errors and of course low sample rate issues...

So let’s get back to our main flow...

Once conversion has taken place, buffering, oversampling and other digital process take place within and at various stages, but all in the digital domain, and all working towards one goal, the reproduction of a linear mathematical equivalent of an analog signal.

Now, this is where DAC’s topologies play a big part, as they each have pro’s and con’s, in terms of trades offs. Some have fewer stages which typically reduce distortions (Time and others), some have higher sample resolutions or faster thread poles, some require the implementation of analog filters post digital processing and the list goes on. As a rule though, Multi-bit topologies currently offer the best blend of compromise and overall value, in terms of dollars (yes – marketing has a say, unfortunately).

Regardless of DA topology, they all produce a myriad of distortions, with most of them rooted in sample resolution and processing time errors (but not all). The reason for this is that digital data – binary, is managed by transistors based processors, which can be viewed as electro -mechanical devices that are capable of turning on an off from a few times a second, to billions of times a second and onward (3.2 GHs CPU etc)... So in other words, it takes time to process or create digital data and that time leads to delays and in modern DAC’s they are processing many things at once and combining the results in a buffer, which in turns takes more time to organize the data, into packets (and makes errors while doing so, leading to more distortions). Once these packets are complete, they are ready to be converted into analog signals or to be transmitted as binary via a SPDIF or converted back to photon pulses and transmitted via a TOSlink output (which can add more errors).

So as you can see, there is a lot of room for error - All of the signal goodness is based primarily on sample rate – then processor rate and software/algorithm quality – and lastly circuit and board design.

I realize that I have left sample rate and circuit board element out of this overview until this point, but I am quite sure that I will be ask about them specifically, as time progress and I will address them at that time.

This is a general overview of the key contributing elements that have the most significant effect on phase response at the digital output stages, as such; the other elements are less important and have been left out off of focus within this discussion.

So let’s pause again and review:

DAC’s manage binary signals via transistors based CPU’s, which are in essences micro relays. These devices open and close controlling the flow of electrons etc. All this creates a bottle neck and takes time to process (hence the name processor). DAC’s process many layers of data at the same time and combine the results in buffers, which sort, compile and create the data packets for streaming to the final stage – digital to analog conversion or other. Most processors are running in the MHz range, which holds them down to at best, low-micro– high-nano scale frequency generation, some esoteric products use faster processors and more advances software algorithms, resulting in faster and more accurate frequency production.

Regardless of speed in the digital domain, they all must enter into a final stage of conversion from digital to analog and it is in this stage where in terms of phase they all incur shifts to a very similar degree, which is to say that most implement very similar analog circuits. Depending of what type of DA topology that the incorporated, they may also be required to additional analog filters to diminish some harsh effects of poor digital processing, which in turn can and does lead to significant phase shifts, centered around the filter frequency, increasing delays into the high micro – low milli second range, in some very entry level products.

In closing,

My point was and is that it has offend been argued that all CD Players/DAC’s should and do sound the same – but, that argument began in 1980 and has most been mostly abandon, as we have all experienced different tonal qualities (sonic signatures) from different DAC’s, as evidence by the fact that, year after year, as a myriad of audio companies release new and improved variants, we continue to trade-up. If you have the opportunity, listen to some older DAC’s within your existing system for a few hours - then put yours back in – I am highly confident that you will hear a difference – who cares if it’s an improvement or not – that would be subjective – but you should perceive a difference and I am sure you will, unless the audio system is very poor, as a whole.

If you’re unable or unwilling to attempt the aforementioned then at least ask yourself this: why if most of us have not perceived such, why have we all been gear junkies for the last 5-10-15-20-25-30 plus years... the answer, simple – if technology hasn’t in fact improved year after year, as objectively verifiable advancements in science and mathematics have claimed, then it can before no other reason than, wait for it – we must be able to hear the difference...

So now ask yourself, where did that difference come from?

Now this is for a few readers specifically: the answer is NOT FROM OUTER SPACE, but from the group effort of Scientists and Engineers!

So what was once argued has passed because the mass major of people have in fact, been able to hear the improvements within each new variant of DAC’s (and other devices) and also that we have been able to start objectively measuring and therefore map more of the contributing: electrical, electro-mechanical and digital factors, which govern the transfer function of the data that is requisite to high quality audio reproduction.

What has been discovered is that a correlation exists that allows for the classification of
DAC performance, based on their frequency phase responses. DAC’s which have mere pico latencies in the delay of frequency production (pre-conversion into analog signal) tend to produce sonic qualities closer to analog (post conversion). Furthermore, it has been discovered that delays, which occur and are greater in the mid range, but within the same scale, (pico, nano etc...) produce a sound that is laid back to most listeners. Most DAC’s produce latencies, (pre-conversion) in the micro ranges and have a poor grouping in latencies scores. In the land of electrons, micros seconds are miles and pico seconds are feet (i am not being literal)! Regardless, of the aforementioned digital qualities, the design and build quality of the analog stages are very important, and heavily influence the final outcome of the analog signal, which will next be present to the preamp – and the story continues!

While we still don’t have the complete picture and may make a few mistakes, while we are still forced to permit some assumption(s), it has become clear, that pico scale latencies are in fact, important to ultra high fidelity audio reproduction – not low!
As to the question, can we actual hear such delays – if we couldn’t they wouldn’t be becoming an important consideration in ultra high fidelity audio design.

May I ask your opinion; on the rather new to north america DEX-P99RS as a source unit. Quality of components, bang for buck $1400 msrp. Scale 1-10.

AAAAAAA

iTrader: (7)

Ok so you are referring to jitter.

I believe some of your conclusions are flawed. I suppose I will get back to this later.

dogbaker

Hi AAAAAAA

I was talking about much more than Jitter, which is a generic term by the way (as in not specific)...

Just as a point of fact - what I have written is not my opinion, it is published fact.

No need to get back to me, as I just outputted 2000 words for your benifit and your reply made it apparent that I wasted my time...

Perhaps others will perceive / receive a benfit from the question(s) that you possed, but that I in fact answered.