ECdesigns

[matthias]

1 hour ago, hopkins said:

I believe the Khadas has an output of 2 vrms and 100 ohm. The powerDAC has 1.4 vrms and  31. 5 ohm.  So unfortunately you cannot have identical output level by lowering the output of the powerDAC. 

  

But the first clip (wav17) with the distortions seems to be louder than the Khadas. Can you elaborate?

Thanks

 

Matt

[Qhwoeprktiyns]

1 hour ago, matthias said:

  

But the first clip (wav17) with the distortions seems to be louder than the Khadas. Can you elaborate?

Thanks

 

Matt

 

The output impedance plays a role as well? Will try again. I do not think I had changed any settings on the recorder. 

[fas42]

Quick listen, without adjusting levels - 18.wav is the pick of the bunch; less digital glare, for want of a better term, 😉.

[matthias]

23 minutes ago, hopkins said:

 

The output impedance plays a role as well? Will try again. I do not think I had changed any settings on the recorder. 

 

When you use the PD with the highest possible output and you attenuate the volume with a preamp or passive VC downstream to the PD do you hear then also distorsions like in the first clip?

Thanks

 

Matt

[Qhwoeprktiyns]

49 minutes ago, matthias said:

 

When you use the PD with the highest possible output and you attenuate the volume with a preamp or passive VC downstream to the PD do you hear then also distorsions like in the first clip?

Thanks

 

Matt

 

The recorder (Tascam DR-100MK3) has an adjustable attenuation from 0 to -12db. I had initially set the attenuation to 0, and must have accidentally recorded the second with a higher attenuation (it is controlled by a wheel on the side of the recorder).

 

I checked all settings again, and recorded both sources with 0 attenuation (and no other settings that could significantly alter the sound).

 

The Khadas obviously has no volume adjustment, so it outputs 2vrms, and I recorded the PowerDAC on its highest gain setting, which corresponds to 1.4vrms.  I used Foobar on my PC to play the original file (with bit perfect mode in both cases).

 

So the PowerDAC will be a little bit lower than the Khadas, and you'll need to adjust for that on whatever system you are using. When I play back both files through the powerDAC, I think I get the same volume by adding 3db (1 level up). So here are the files (28 KD.wav = Khadas, 29 PD.wav = PowerDAC):

 

https://storage.googleapis.com/cloudplayer/Feuilles Mortes/28 KD.wav

 

https://storage.googleapis.com/cloudplayer/Feuilles Mortes/29 PD.wav

 

I only recorded a minute to avoid the louder passage later on.

 

Here is the original file: https://storage.googleapis.com/cloudplayer/Feuilles Mortes/24 mieko miyazaki %26 guo gan - les feuilles mortes.flac

 

 

[bodiebill]

3 hours ago, hopkins said:

When comparing DACs the output level has to be taken into account. The Musician Pegasus, that @bodiebill found more involving, has 2.2 vrms output versus 1.4 for the powerDAC. Don't know the output impedance of the Pegasus. 

 

The volume I just set by ear. And the comparison was mainly long term, in addition to A/B, switching DACs every other day with many different tracks. It feel it takes time to really get to know DACs at this level. 

[Varinder]

11 hours ago, Jacob said:

Hello To all.

I took a pause writing here since was waiting for my USB to Toslink converter.

The one i bought is from NobSound , after confiding with Gordon (from EC) - 
https://www.amazon.co.uk/Digital-Interface-Coaxial-Optical-Adapter/dp/B08GYWTY4Y/ref=sr_1_2?dchild=1&keywords=Nobsound+Mini+XMOS+XU208+USB&qid=1628539450&s=electronics&sr=1-2

 

It works perfectly, it can be powered but also PASIVE and i use it like this.

I recomend to order from Amazonm it arrive super fast.... my first unit from NobSound is still on the way after 1 month...

 

While i was waiting ... I got here The Memory Player NUC !!!
I want to thank someone here that mantiond them since inever knew that they excist,

 

As i wrote before  - once you step up with the DAC ... and the PD is one of the best if not THE BEST - everything start to matter.

Every part in the audio chain become more important since the resolution and clarity is so high you just start to notice things that are simply were not visable before.
I also advice to use Intona USB isolator 7055C . Far better than 7054 models!!

 

So here is my setup:
1. Memory Player NUC
2. simple USB 
3. Intona 7055C
4. Neotech NEUB-1020 USB cable 
5. NobSound Mini XMOS converter
6. EC PD optical cable
7. EC Power DAC
8. Kennerton Rognir Headphone
or

* Neotech NEVD-2010 x2 as Interconnect
* AURIS Ueterpe
* Kennerton Rognir Headphone

Most of the time i use the Auris amp since it is more convinient to adjust the volume with different headphones - and YES direct connect to PD sounds BETTER !

The Memory Player NUC is simply amazing source  - as the PD came along it is  become very clear that the file source is super critical.
For Comparison i use QLS390 as a WAV player simulating HQ CD transport - it was the best sound .... until the MP NUC came and killed it :):)

A few days ago i had couple of friends came over, they are Pro guys that have HQ equipment from Music MIxing - they were FLOORED by the sound !! As I AM TOO.

We try to describe the sound here and this is not easy... how can you describe Analogue sound from Digital source? 
Well .... as always with audio  - you will have to try yourself :):)

Anyone wish to contact me - +972-54-9333389  Jacob. Please use WhastApp.

P.S - i just try to use PD with a Power Bank insted of the Power Suply - and it worked quite well.
I will do more test and compare.

Don’t your Memory player has toslink out ,., if so did you try toslink option vs Usb out with PD .. any difference?

[matthias]

57 minutes ago, hopkins said:

Here is the original file: https://storage.googleapis.com/cloudplayer/Feuilles Mortes/24 mieko miyazaki %26 guo gan - les feuilles mortes.flac

 

Is the Khadas or the PD playback more similar to the original file?

 

Matt

[tapatrick]

11 hours ago, Jacob said:

While i was waiting ... I got here The Memory Player NUC !!!

Great choice :) and incredible value. If you are up for it I can recommend a few modifications.

Put the NUC into an Akasa fanless case which removes the noisy internal fan and houses the NUC in a robust metal case. 

https://www.youtube.com/watch?v=vQcrydSYrtE

I also upgraded the Ram stick, did not reconnect the wifi and microphone and removed the SSD (if you are not going to use it for storing music). Then it can run on 12v - I am using a Paul Hynes SR4 rather than the supplied brick.

 

[Qhwoeprktiyns]

1 hour ago, matthias said:

 

Is the Khadas or the PD playback more similar to the original file?

 

Matt

 

Playback of the original file through the PD sounds better than the other two but I would give a slight edge to the PD recording over the Khadas recording. Khadas recording sounds a little more aggressive to my ears. 

 

I also took the opportunity to compare again playback of the UPL versus Foobar/UT96 and Foobar/U192 and I still cannot hear differences.  I really don't know what to say about all this... I am a little bit puzzled about the results of the use of the Memory Player + Intona Isolator reported earlier by Jacob, especially since this is all going through an XMOS USB -> optical converter which is probably introducing again noise/jitter... Taking a fancy player + isolator but putting it through a basic converter you have to wonder what the benefit would be.

 

It would be good to do some blind testing comparing this setup with the UPL, and then again with a basic source. 

 

I am sure the Memory Player is a good player to use when you don't have the PowerDAC, but I cannot help being spkeptical about its relevance here. I wish I could try one or better, once again, listen to all this with others in attendance, in the same system.

[Jacob]

3 hours ago, Varinder said:

Don’t your Memory player has toslink out ,., if so did you try toslink option vs Usb out with PD .. any difference?

No, this NUC model dont have optical out.
not sure but i think that optical out from a PC will not perform as optical out from a CD player. 
Im working now with MP guys to figure out a simple and more cheaper solution for the MP a way that it could be sold via distributors and wont need special instalation for each client  - this will reduce the cost segnificaly.
But choosing a hardware with Optical out will increase the total cost.
They also prefer the USB option, and it is better to have Intona on the line  - this unit will follow you everywhere.

[Jacob]

1 hour ago, hopkins said:

 

Playback of the original file through the PD sounds better than the other two but I would give a slight edge to the PD recording over the Khadas recording. Khadas recording sounds a little more aggressive to my ears. 

 

I also took the opportunity to compare again playback of the UPL versus Foobar/UT96 and Foobar/U192 and I still cannot hear differences.  I really don't know what to say about all this... I am a little bit puzzled about the results of the use of the Memory Player + Intona Isolator reported earlier by Jacob, especially since this is all going through an XMOS USB -> optical converter which is probably introducing again noise/jitter... Taking a fancy player + isolator but putting it through a basic converter you have to wonder what the benefit would be.

 

It would be good to do some blind testing comparing this setup with the UPL, and then again with a basic source. 

 

I am sure the Memory Player is a good player to use when you don't have the PowerDAC, but I cannot help being spkeptical about its relevance here. I wish I could try one or better, once again, listen to all this with others in attendance, in the same system.

MP - it IS the player to use with PD!!
Because PD is so good and revealing - the quality of the source become more important.

Wait for a month +/- - we will present a relativly cheaper version of MP  - i think it will be based on this PC:
https://www.pepper-jobs.com/collections/mini-pc/products/glk-uc2x-mini-pc

with this remote control:
https://www.pepper-jobs.com/products/w10-gyro-azerty-smart-remote

and this option too:
https://www.monect.com/  as a phone control

 

I will do more test compared to UPL vs MP and post here.

[Qhwoeprktiyns]

49 minutes ago, Jacob said:

Because PD is so good and revealing - the quality of the source become more important.

 

So far, my experience has been "Because PD is so good the quality of the source is no longer important". I am starting to feel very lonely supporting this claim :)

 

[bodiebill]

Time for a poll? Anyone know how to do a proper one with vote boxes? In the vein of:

 

  A. With the PD the quality of the source become more important.

  B. With the PD the quality of the source become less important.

 

A highlights the 'revealing' aspect, B the 'correcting' or 'repairing' aspects in the sense of noise and jitter. @hopkins please let us know if you have a different look on B.

 

I would only be interested in votes based on experience, not on theory (which can go either way).

[Qhwoeprktiyns]

47 minutes ago, bodiebill said:

A highlights the 'revealing' aspect, B the 'correcting' or 'repairing' aspects in the sense of noise and jitter. @hopkins please let us know if you have a different look on B.

 

Don't get me started 😅

B is not repairing or correcting:

- noise from the source is essentially blocked (with optical). There remains potentially a fraction (negligeable) of noise in the small bandwidth used for the actual signal (192khz). 

- timing information from the source is not used. It is not relevant. 

 

I think we already know what the poll results will be, no? 

[bodiebill]

24 minutes ago, hopkins said:

B is not repairing or correcting:

- noise from the source is essentially blocked (with optical). There remains potentially a fraction (negligeable) of noise in the small bandwidth used for the actual signal (192khz).

So 'mostly blocking' for noise?

 

Quote

- timing information from the source is not used. It is not relevant. 

Dou you mean the PD is 'fully reclocking' for jitter?

 

Could not it be argued that there is a 'correcting' aspect to both?

 

Quote

I think we already know what the poll results will be, no? 

Not so sure.

To be honest, I would choose B as that is what I experienced. But the question is whether that is an entirely good thing. If, for instance, I would throw a blanket over my speakers, I am sure this would increase source immunity. Not trying to be funny here, just to aid my thinking about the subject, which appears not as clear-cut as we once thought.

 

[Qhwoeprktiyns]

18 minutes ago, bodiebill said:

So 'mostly blocking' for noise?

 

Dou you mean the PD is fully 'reclocking' for jitter?

 

Could not it be argued that there is a 'correcting' aspect to both?

 

Not so sure.

To be honest, I would choose A as that is what I experienced. But the question is whether that is an entirely good thing. If, for instance, I would throw a blanket over my speakers, I am sure this would increase source immunity. Not trying to be funny here, just to aid thinking about the subject, which appears not as clear-cut as we once thought.

 

 

Yes, blocking for noise from the source (with the exception of RF if the source is really close to the rest of the equipment). 

 

Whether "discarding" = "blocking", I'll let you decide. For the clock signal it is not reclocking in the traditional sense. There is no "DAC chip" which uses a clock signal in the powerDAC. I'll elaborate later (John send me an explanation). This is pretty technical.

 

The powerDAC is not throwing a blanket over the source. It is using the "good stuff" (the binary data representing the frequencies/dB).

This data is essentially always the same whether you use a phone or a Taiko Extreme as a source. The way it is send to the DAC can be done differently. 

 

 

 

 

[Henley]

I do not see any benefits of this experiment above given that the AD and used software probably is not tranparent enough to make any sort of conlusion.

 

Furthermore you should do 20-50 AD/DA loops in order to find out whether a certain DAC is more transparent than the other. If I’m correct there was a thread on Gearsluts years ago were they did such an extensive experiment with the Lynx Hilo and Prism Lyra.

[Qhwoeprktiyns]

2 minutes ago, Henley said:

I do not see any benefits of this experiment above given that the AD and used software probably is not tranparent enough to make any sort of conlusion.

 

Yes, that's essentially my conclusion as well. The ADC in the recorder levels the sound differences. 

[Khronos]

Hello. Due to monetary reasons I won't be able to meaningfully contribute to this thread with real experiences. But I've spent this last hour comparing samples of the KDTB and the ECDPB with the original song. At first, I thought the Khadas and the PD were closely matched with a slight nod to the PD due to the lack of "stepping" (a transition between sounds, as if the sound was climbing some stairs between transients) and the fact it wasn't as forward, less harsh. Though I wouldn't have blamed anybody for not hearing differences.

 

But there was something that kept at it in my brain. I can't tell what it was. So I jumped over to YouTube and mentally compared the tonality of the PowerDAC to several other dacs through Youtube. Call it insanity, but I can archive the way something sounds due to strange brain DSP (I am not sure what it is or whether I belong in a mental hospital). MSB Select II: Midrange Bump. Aries Cerat Kassandra Ref: One of the most tone Deaf DACs I've heard. PD Dream DAC: Stepping is like an upwards slide (Something I understand people would love) but lacks some Gestalt in its tone. I turned over to this shootout (

)

 

And checked my usual boxes. In my opinion the LAB12 actually sounds best, and was hereto my reference due to perfectly natural timbre coupled with spacious sound and non-existent stepping (it starts and stops!). I can easily hear how the amp actually holds back the rest of the system. I kept comparing. Turned back to the WAVs, turned back to YouTube, to the original. Something told me "This lacks palpability, flow" and some other words that don't have meaning outside of my brain. Then in a whim I turned over to recording 18.

 

There was absolutely no difference between recording and original.

 

Everything was there, just slightly quieter. I could wax lyrically about the sound but the best way I could describe it is "this breaks the dogma of digital versus analog". It unfettered the recording. How could this be possible? I am not an engineer, but I have a small understanding of ECD Technology  based on Mr. Brown's post. If memory serves, this has 6800uF or so of buffer capacity, similar to the famed Bakoon International AMP-13R. Yet to my knowledge the latter sounds effortless albeit lacking tone weight. At the beginning of writing this post I thought it could be an issue with the actual capacity of the buffer until I remembered the AMP-13R. Thus came in the second suspect: Slew Rate. How quickly can the caps inject juice into the sound? The Power Supply is at least as important as the signal path so chances are this buffer could be holding things back due to mismatched slew rate or something (mind that this runs at a maximum speed of 192khz) thus summoning weird electrical shenanigans that could not particularly be foreseen and creating this problem. I've seen film capacitors being used in parallel so perhaps this could be a fixture for the problem, something worth considering for the more heavy-weight Powerdac-S (the beast I am saving up for). Though for most, just running it one or two notches lower and adding a bit of extra gain at preamp would work. This sounds like a hassle, but let me explain. The Powerdac-R is €1500. People spend ten times as much in a single USB cable. If the DAC actually sounds like sample 18 I think the PD may have the rights to be called the best DAC in the world. Unmatched. Capable of going against the MSB Select II, the Chord DAVE, the Lampizator Pacific, and possibly better them. A true case of David versus Goliath. And even against the LAB12, i think they are both more complementary than anything. I truly hope ECD follows the steps of Soekris and begins OEMing the board. Even at the same cost of the PowerDAC with everything included it's relatively speaking almost highway robbery.

 

It is 1:30 AM. I admit I just made an account to say this.

 

Therefore I think the inner PSU lacks some sort of slew rate. I can't hear the usual signs of the recorder clipping (death of DR coupled with a heaviness of midrange) so I can't think that was the problem. Unless the universe is pulling a Lejonklou (look up the Lejonklou SINgularity story) and there's two resistor values that surpass the rest I truly can't think of another reason.

 

Do you have any comments on this? Opinions? I am not a golden ears (more like insane teenager) but I thought this could be a worthy attempt to shed light on the conundrum. 

 

PostData: Hello, I'm new to audiophilestyle. I hope I don't annoy people too much.

[Qhwoeprktiyns]

Concerning "reclocking" here is a lengthy mail that John send me to explain how the powerDAC works. It is obviously technical but some here may be interested in getting a deeper understanding of this. 

 

"The PowerDAC-R has no reclocker. 

 

With re-clocker I refer to receiving a -jittery- clock (from source or XMOS processor for example) and feeding it into a circuit that re-times it using a low jitter master clock.

 

Isolators (in an external re-clocker for example) will be placed in series with the perfect insulation provided by Toslink. It's like extending the say 2 meter Toslink barrier by say 0.05mm (optical insulator thickness on a digital insulator chip often used for USB insulation). In other words, it's irrelevant in the case of the PowerDAC-R.

 

Most re-clockers have to use asynchronous re-clocking because there is no mechanism available to slave the source using feedback. When source and DAC clocks aren't synchronised, this results in pops, clicks and dropouts.

 

This is usually done using an asynchronous sample rate converter chip. The incoming signal is re-sampled using a fixed frequency master clock. The frequency of this master clock is not critical and it doesn't need to be an exact multiple of the incoming sample rate. All samples are re-calculated, so in and output streams stay in sync. 

By doing so, a new digital output signal is generated with different sample values, in such way that input and output streams are time aligned again. The new samples will also contain the phase noise (jitter) of the master clock. In other words, the master clock jitter is now fused to the new output sample values.

 

So synchronisation is obtained by changing the data. I am not 100% sure, but it seems the Mutec re-clockers -have- to use this or similar method in order to re-clock all kind of different input signals.

 

Only USB is an exception, here we can use an XMOS or Atmel (Amanero) chip that provides feedback to the source. This way a fixed master clock an be used and the source can be instructed (feedback) to adjust its data rate so it stays in perfect sync with the fixed master clock. 

The master clock needs to be an -exact- multiple of the incoming sample rate (usually 22.5792 Mhz for the 44.1 KHz group and 24.596 MHz for the 48KHz group).

 

These XMOS & Atmel based re-clockers do not change the data but maintain sync using feedback.

 

XMOS and Atmel processors will output -massive- jitter because of how these processors work. These jitter loaded I2S, S/PDIF or DSD output signals are then synchronously re-clocked with the master (audio) clock. The re-clocked signal then appears to have low jitter.

 

-> By doing so we create a powerful radio transmitter (master clock = carrier frequency, jitter modulates it) that transmits the jitter spectrum wirelessly across all circuits (because circuits are almost never shielded inside a DAC). What should work in theory doesn't work very well in practice. This is why most DACs based on such re-clockers and stellar master clocks are still source sensitive.

 

Asynchronous sample rate converter as its name implies can also perform sample rate conversion (up/downsampling) and other DSP actions like filtering out pops and clicks. 

The downside is that the data will have to be changed in the process. So playback is no longer bit-perfect, the DAC receives different data and obviously will sound different, even if a theoretical DAC would provide 100% source immunity, as this has -nothing- to do with noise or jitter.

"The PowerDAC-R -only- collects the data from the incoming signal and places it in RAM (Random Access Memory).

 

Next, the incoming sample rate is measured with a software-based frequency meter. So now the data is available and the playback rate is determined.

 

Low jitter master clock (Vectron, Pletronics, -85 ... -90dB phase noise @ 10Hz) is used to generate a clean latch signal for the Fractal D/A converter. So we are -not- re-clocking a jittery clock signal and creating said radio transmitter.  We are generating a new, clean clock from scratch that will be used to latch the data (we stored in RAM) into the Fractal D/A converter.

The mentioned synchronisation problem was fixed using a proprietary time domain fusing protocol.

 

The Fractal D/A converter is completely different from existing D/A converters as it chops up the most significant bits (the most critical one's for obtaining low bit errors) into tiny fragments. This is what the fractal circuit does. The fragmenting reduces bit errors and glitches. The downside is that we have to use more bits that represent the same bit depth (circuit gets a bit more complicated).

 

The Fractal D/A converter does not work with I2S or other serial data interface and requires no clocks to clock in serial data because there is no serial data. All problems related to high frequency clocks and high frequency serial data are eliminated this way.

 

All bits are presented to the Power D/A converter simultaneously (parallel). Then we wait until all bits are stable and switching noise is completely gone (because the bits no longer change). Now one single latch pulse (derived form the low jitter master clock) will write the new sample to the D/A converter output at the moment there is minimum electrical interference. This helps to minimise trigger uncertainty and provides a clean, low jitter latch signal.

 

In other words we only have to load new data and latch this data once every sample. The highest frequency we need (data and latch) equals the sample rate (44.1 ... 192 KHz). With conventional DACs we need much higher clock and data rates, typically up to 24.576 MHz. These higher data and clock rates produce much more switching noise and because there is no radio silence during latching, jitter at the D/A conversion circuit will be rather high regardless of master clock phase noise (jitter).

 

The bandwidth we need for getting data into the PowerDAC-R D/A converter is therefore only 192 KHz. This helps to keep most of the source noise that still has to enter through Toslink out of the D/A converter output signal.

 

So what have we achieved with this?

 

1) The single, low frequency latch signal is -completely- independent from the source and because we use professional clocks, phase noise and related jitter is very low (data sheet specifies -85 ... -90 dB phase noise @ 10Hz offset). And that is pretty good. 

Because the latch frequency is much lower than this master clock frequency, the impact of this phase noise is further reduced, this is not the case with standard DACs that need the native master clock frequency for serial data clocking.

 

2) The large bandwidth noise from the source (bandwidth up to 1 GHz is required to get the data through) is band limited by the optical Toslink interface to approx. 25MHz. This minimises the noise injection into the D/A converter. Because we can't fully block all noise, marginal source dependency remains but in most cases this is so little that it's a non issue."

[Qhwoeprktiyns]

In addition to all this keep in mind that any time you change your system configuration, plugging in different equipment, you may get different results simply due to ground loops/interference which have nothing to do with the above explanations.

[yogibear]

4 hours ago, Khronos said:

Hello. Due to monetary reasons I won't be able to meaningfully contribute to this thread with real experiences. But I've spent this last hour comparing samples of the KDTB and the ECDPB with the original song. At first, I thought the Khadas and the PD were closely matched with a slight nod to the PD due to the lack of "stepping" (a transition between sounds, as if the sound was climbing some stairs between transients) and the fact it wasn't as forward, less harsh. Though I wouldn't have blamed anybody for not hearing differences.

 

But there was something that kept at it in my brain. I can't tell what it was. So I jumped over to YouTube and mentally compared the tonality of the PowerDAC to several other dacs through Youtube. Call it insanity, but I can archive the way something sounds due to strange brain DSP (I am not sure what it is or whether I belong in a mental hospital). MSB Select II: Midrange Bump. Aries Cerat Kassandra Ref: One of the most tone Deaf DACs I've heard. PD Dream DAC: Stepping is like an upwards slide (Something I understand people would love) but lacks some Gestalt in its tone. I turned over to this shootout (

)

 

And checked my usual boxes. In my opinion the LAB12 actually sounds best, and was hereto my reference due to perfectly natural timbre coupled with spacious sound and non-existent stepping (it starts and stops!). I can easily hear how the amp actually holds back the rest of the system. I kept comparing. Turned back to the WAVs, turned back to YouTube, to the original. Something told me "This lacks palpability, flow" and some other words that don't have meaning outside of my brain. Then in a whim I turned over to recording 18.

 

There was absolutely no difference between recording and original.

 

Everything was there, just slightly quieter. I could wax lyrically about the sound but the best way I could describe it is "this breaks the dogma of digital versus analog". It unfettered the recording. How could this be possible? I am not an engineer, but I have a small understanding of ECD Technology  based on Mr. Brown's post. If memory serves, this has 6800uF or so of buffer capacity, similar to the famed Bakoon International AMP-13R. Yet to my knowledge the latter sounds effortless albeit lacking tone weight. At the beginning of writing this post I thought it could be an issue with the actual capacity of the buffer until I remembered the AMP-13R. Thus came in the second suspect: Slew Rate. How quickly can the caps inject juice into the sound? The Power Supply is at least as important as the signal path so chances are this buffer could be holding things back due to mismatched slew rate or something (mind that this runs at a maximum speed of 192khz) thus summoning weird electrical shenanigans that could not particularly be foreseen and creating this problem. I've seen film capacitors being used in parallel so perhaps this could be a fixture for the problem, something worth considering for the more heavy-weight Powerdac-S (the beast I am saving up for). Though for most, just running it one or two notches lower and adding a bit of extra gain at preamp would work. This sounds like a hassle, but let me explain. The Powerdac-R is €1500. People spend ten times as much in a single USB cable. If the DAC actually sounds like sample 18 I think the PD may have the rights to be called the best DAC in the world. Unmatched. Capable of going against the MSB Select II, the Chord DAVE, the Lampizator Pacific, and possibly better them. A true case of David versus Goliath. And even against the LAB12, i think they are both more complementary than anything. I truly hope ECD follows the steps of Soekris and begins OEMing the board. Even at the same cost of the PowerDAC with everything included it's relatively speaking almost highway robbery.

 

It is 1:30 AM. I admit I just made an account to say this.

 

Therefore I think the inner PSU lacks some sort of slew rate. I can't hear the usual signs of the recorder clipping (death of DR coupled with a heaviness of midrange) so I can't think that was the problem. Unless the universe is pulling a Lejonklou (look up the Lejonklou SINgularity story) and there's two resistor values that surpass the rest I truly can't think of another reason.

 

Do you have any comments on this? Opinions? I am not a golden ears (more like insane teenager) but I thought this could be a worthy attempt to shed light on the conundrum. 

 

PostData: Hello, I'm new to audiophilestyle. I hope I don't annoy people too much.

I can echo your impressions with mine. I have been a full range driver fan playing in different open baffle configurations. Majority of DIYers are full range drivers haters. I am just lucky to have some rare drivers, vintage nos which have respectable frequency response from 17k down to under 75hz. (Measured and confirmed on REW). Playing them with a short simple chain enables me to do very fair comparison of amps and DAC and even the drivers themselves too.
 

I have been fiddling with PD in my chain for long and have found it sounds and shows it’s prowess when it’s clicked above 7 in my setup. The details and HF extension along with the natural vocals and effortless play back is exemplary.

 

Though I cannot still confirm the source independence but with great recordings, it’s beyond belief and gives non stop, unlimited joy. 

[Qhwoeprktiyns]

17 minutes ago, yogibear said:

I am just lucky to have some rare drivers, vintage nos which have respectable frequency response from 17k down to under 75hz

 

I have always fantasized about vintage full range drivers. I got a hold of a pair of Zenith, these: 

 

http://www.glowinthedarkaudio.com/zenith-49cz852.html

 

I tested them in an open baffle. Very easy to drive, impressive dynamics and detail, but it was not satisfying for every day listening, just a really interesting "experiment". Can you share more information about your setup? 

 

Concerning the volume setting of the PD, which amplifier do you use it with? 

[bodiebill]

5 hours ago, Khronos said:

Hello. Due to monetary reasons I won't be able to meaningfully contribute to this thread with real experiences. But I've spent this last hour comparing samples of the KDTB and the ECDPB with the original song. At first, I thought the Khadas and the PD were closely matched with a slight nod to the PD due to the lack of "stepping" (a transition between sounds, as if the sound was climbing some stairs between transients) and the fact it wasn't as forward, less harsh. Though I wouldn't have blamed anybody for not hearing differences.

 

But there was something that kept at it in my brain. I can't tell what it was. So I jumped over to YouTube and mentally compared the tonality of the PowerDAC to several other dacs through Youtube. Call it insanity, but I can archive the way something sounds due to strange brain DSP (I am not sure what it is or whether I belong in a mental hospital). MSB Select II: Midrange Bump. Aries Cerat Kassandra Ref: One of the most tone Deaf DACs I've heard. PD Dream DAC: Stepping is like an upwards slide (Something I understand people would love) but lacks some Gestalt in its tone. I turned over to this shootout (

)

 

And checked my usual boxes. In my opinion the LAB12 actually sounds best, and was hereto my reference due to perfectly natural timbre coupled with spacious sound and non-existent stepping (it starts and stops!). I can easily hear how the amp actually holds back the rest of the system. I kept comparing. Turned back to the WAVs, turned back to YouTube, to the original. Something told me "This lacks palpability, flow" and some other words that don't have meaning outside of my brain. Then in a whim I turned over to recording 18.

 

There was absolutely no difference between recording and original.

 

Everything was there, just slightly quieter. I could wax lyrically about the sound but the best way I could describe it is "this breaks the dogma of digital versus analog". It unfettered the recording. How could this be possible? I am not an engineer, but I have a small understanding of ECD Technology  based on Mr. Brown's post. If memory serves, this has 6800uF or so of buffer capacity, similar to the famed Bakoon International AMP-13R. Yet to my knowledge the latter sounds effortless albeit lacking tone weight. At the beginning of writing this post I thought it could be an issue with the actual capacity of the buffer until I remembered the AMP-13R. Thus came in the second suspect: Slew Rate. How quickly can the caps inject juice into the sound? The Power Supply is at least as important as the signal path so chances are this buffer could be holding things back due to mismatched slew rate or something (mind that this runs at a maximum speed of 192khz) thus summoning weird electrical shenanigans that could not particularly be foreseen and creating this problem. I've seen film capacitors being used in parallel so perhaps this could be a fixture for the problem, something worth considering for the more heavy-weight Powerdac-S (the beast I am saving up for). Though for most, just running it one or two notches lower and adding a bit of extra gain at preamp would work. This sounds like a hassle, but let me explain. The Powerdac-R is €1500. People spend ten times as much in a single USB cable. If the DAC actually sounds like sample 18 I think the PD may have the rights to be called the best DAC in the world. Unmatched. Capable of going against the MSB Select II, the Chord DAVE, the Lampizator Pacific, and possibly better them. A true case of David versus Goliath. And even against the LAB12, i think they are both more complementary than anything. I truly hope ECD follows the steps of Soekris and begins OEMing the board. Even at the same cost of the PowerDAC with everything included it's relatively speaking almost highway robbery.

 

It is 1:30 AM. I admit I just made an account to say this.

 

Therefore I think the inner PSU lacks some sort of slew rate. I can't hear the usual signs of the recorder clipping (death of DR coupled with a heaviness of midrange) so I can't think that was the problem. Unless the universe is pulling a Lejonklou (look up the Lejonklou SINgularity story) and there's two resistor values that surpass the rest I truly can't think of another reason.

 

Do you have any comments on this? Opinions? I am not a golden ears (more like insane teenager) but I thought this could be a worthy attempt to shed light on the conundrum. 

 

PostData: Hello, I'm new to audiophilestyle. I hope I don't annoy people too much.

 

Welcome Khronos, inspiring post. Good to add some keen young ears.

When back home (now traveling) I will try your volume setting suggestion.