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iFi audio Ships Pro iDSD DAC/Streamer

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22 minutes ago, audiobomber said:

Crysopeia FPGA. Check the Overview and Specifications tabs on the iFi site: https://ifi-audio.com/portfolio-view/pro-idsd/

 

thanks for link....

 

this sounds intereseting...

Studio Grade Remastering DSD1024
For the first time, Pro iDSD brings Studio Grade DSD1024 remastering to the mass. Now one can remaster all his/her music to a superlative DSD1024 format just like in a recording studio and enjoy the ultra-high resolution it provides.

 

For the DSD1024, if the MultiBit DAC is one half of the ‘heart’ of the Pro iDSD’s digital engine, then the other half is surely the Crysopeia FPGA. This is where we believe FPGA excels, by handling the remastering duties to attain the DSD1024 audio format. In fact the Pro iDSD can handle all audio formats to to DSD1024 or DSD512 or PCM 768kHz with user-selectable digital filters.

 

I didn't like TEAC hardware upsampling, but maybe this will be better?  Look forward to reviews.

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I wonder if they benefit from their own "ifi regens or ifi powers", or if they believe they incorporated these things in this streamer/dac...I am not buying another dac that needs toys...eventually someone will get it right with the design without the need of band-aid toys.

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  Read the power supply section. Looks like a filtered switcher. 


 

2012 Mac Mini, i5 - 2.5 GHz, 16 GB RAM. SSD,  PM/PV software, Focusrite Clarett 4Pre 4 channel interface. Daysequerra M4.0X Broadcast monitor., My_Ref Evolution rev a , Klipsch La Scala II, Blue Sky Sub 12

Clarett used as ADC for vinyl rips.

Corning Optical Thunderbolt cable used to connect computer to 4Pre. Dac fed by iFi iPower and Noise Trapper isolation transformer. 

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2 hours ago, Panelhead said:

  Read the power supply section. Looks like a filtered switcher. 

 

"Using classic tube design, brought up-to-date with 21st Century technology, all incoming DC is converted to a high-frequency waveform then rectified and filtered by a choke input capacitor filter. This produces a first-level DC bus from which all further voltages are derived."

 

Can anyone explain how they are linking switched power supplies to "classic tube design"?


Hey MQA, if it is not all $voodoo$, show us the math!

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from the specs:

 

"DSD filters:  fixed 3rd order analogue filter @ 80kHz with correction for DSD’s -6dB gain"

 

So 18db down at 160kHz right?  You would not want to send it DSD64, or even DSD128 probably...??

 

Is this why @mansryou mentioned one time its best to send DSD256 to the iDSD line (micro, etc.)?

 

 


Hey MQA, if it is not all $voodoo$, show us the math!

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I should note that I am still intrigued - what's the retail?


Hey MQA, if it is not all $voodoo$, show us the math!

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6 minutes ago, crenca said:

from the specs:

 

"DSD filters:  fixed 3rd order analogue filter @ 80kHz with correction for DSD’s -6dB gain"

 

So 18db down at 160kHz right?  You would not want to send it DSD64, or even DSD128 probably...??

 

Is this why @mansryou mentioned one time its best to send DSD256 to the iDSD line (micro, etc.)?

That sounds familiar.

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Perhaps @AMR/iFi audioor someone else can chime in.  If you source is DSD64, it was my understanding that you would want your analog filter to kick in around 25-30kHz where the HF noise kicks in due to the noise shaped nature of DSD.  With DSD128, it would be around 50kHz.  

 

Perhaps this is a limitation of DAC's able to accept a range of DSD rates - you can only have one analog filter, at least without significant cost/complexity?

 

It would not matter to me because I would upsample to DSD256 or higher in any case, but it appears it would matter to folks playing DSD64 source files directly to the DAC...


Hey MQA, if it is not all $voodoo$, show us the math!

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1 hour ago, crenca said:

 

"Using classic tube design, brought up-to-date with 21st Century technology, all incoming DC is converted to a high-frequency waveform then rectified and filtered by a choke input capacitor filter. This produces a first-level DC bus from which all further voltages are derived."

 

Can anyone explain how they are linking switched power supplies to "classic tube design"?

 

via a marketer's keyboard

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I have one here for review and I've been mostly using it with the Gibbs Transient Optimised filter, which is a 16k tap filter that takes the music up to 705/768k, without the DSD re-sampling. You can switch in our out the tube circuit. For me, it reaches just about Hugo 2/Qutest level of performance, though I need to focus on snippets of high-quality guitar or piano music to pick them apart going through a headphone amp and Focal Utopias.

 

It has other filter options, such as "Bit Perfect" which are predictably softer.

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10 hours ago, AMR/iFi audio said:

If you were to filter DSD at 25kHz you would downgrade the transient/supersonic audio performance to CD standard levels, which is pointless.

 

Hardly : -3dB @ 25kHz is rather a long way away from brickwall (say -96dB) @ 22kHz in both phase and amplitude.

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2 hours ago, Currawong said:

I have one here for review and I've been mostly using it with the Gibbs Transient Optimised filter, which is a 16k tap filter that takes the music up to 705/768k, without the DSD re-sampling. You can switch in our out the tube circuit. For me, it reaches just about Hugo 2/Qutest level of performance, though I need to focus on snippets of high-quality guitar or piano music to pick them apart going through a headphone amp and Focal Utopias.

 

It has other filter options, such as "Bit Perfect" which are predictably softer.

 

Hi, nice initial impressions video.

 

The manual states iFi recommend using the volume knob past 12 o'clock (same as their iDSD range) due to the analogue volume control.

 

How far along the volume knob do you need to go with your MrSpeakers AEON closed, with say modern louder music?

 

That was a problem I had with the iDSD BL - I could only use it in ECO and IEMatch 'High' mode to get past 12 o'clock and dynamics were too restricted with my AEON closed with these settings (probably current limited with these settings). For these cans, I find the xDSD much better in that regarding but interested in the Pro iDSD.

 

I kind of wished the Pro iDSD volume worked like the xDSD where the volume control itself is entirely analogue but the knob/adjustment is digital - so channel imbalance is a non issue. But when I read the Pro iDSD manual and it's recommendation to use > 12o'clock, it reminded me of my iDSD BL issues.

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15 hours ago, AMR/iFi audio said:

In the SACD 'Scarlet Book', Sony and Philips recommend the use of a 50kHz 3rd order low-pass filter with a mandated 100kHz filter cutoff. Our 80kHz filter allows a little more noise to pass than the 50khz recommendation.

Not a little. A lot.

 

sacd-noise.thumb.png.e14e7cc6f4a50cc5ddc0105db707aa18.png

 

This is the frequency spectrum of an actual SACD. There's a lot of noise energy in the 50-80 kHz band. If you ask me, even 50 kHz is too high a cut-off for this noise profile. 30 kHz would be more reasonable.

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20 hours ago, AMR/iFi audio said:

 

In the SACD 'Scarlet Book', Sony and Philips recommend the use of a 50kHz 3rd order low-pass filter with a mandated 100kHz filter cutoff. Our 80kHz filter allows a little more noise to pass than the 50khz recommendation.

 

If you were to filter DSD at 25kHz you would downgrade the transient/supersonic audio performance to CD standard levels, which is pointless.

 

4 hours ago, mansr said:

Not a little. A lot.

 

sacd-noise.thumb.png.e14e7cc6f4a50cc5ddc0105db707aa18.png

 

This is the frequency spectrum of an actual SACD. There's a lot of noise energy in the 50-80 kHz band. If you ask me, even 50 kHz is too high a cut-off for this noise profile. 30 kHz would be more reasonable.

 

This is part of the debate around DSD as I understand it (and thus I could be wrong):  the significance of this noise, where and how the the noise (and thus the design of the filter) effects "transient/supersonic audio performance", etc.

 

Personally, for me I lean towards mansr here that letting the noise through is worse than any theoretical transient performance which near as I can tell, is based on the very unproven and speculative idea(s) about ultrasonic "speed" affecting the waveform and thus lower audible frequencies that we are all interested in.  On the other hand, the noise is low in level - though approaching 80 dbfs down is usually otherwise notable...

 

 

 


Hey MQA, if it is not all $voodoo$, show us the math!

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7 hours ago, crenca said:

In the SACD 'Scarlet Book', Sony and Philips recommend the use of a 50kHz 3rd order low-pass filter

 

Hmmh, this is actually wrong. I don't know where people pulled this figure, but I've seen it circulating around every now and then.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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1 minute ago, mansr said:

Let's approach this from the other end. If too much ultrasonic noise reaches the tweeters, worst case, they burn out. If not, it will likely cause distortion. Even if the tweeters can handle the noise, it is a waste of amp power. Now most decent amps have a bandwidth of 100-200 kHz. Noise in the transition band is likely to result in IMD products appearing in the audible range. Not a great idea. The best solution is to remove the noise using a dedicated filter as early as possible, in the DAC.

 

DSD noise is quite little concern overall. It is uncorrelated with the music (when using good modulator), so the intermodulation products are also noise - only increase the audio band noise floor. While much bigger concern with many DAC chips when using PCM inputs are the left over images which are fully correlated with music, and thus intermodulation products are also correlated and cause clearly discernible distortion tones.

 

OTOH, class-D power amplifiers tend to leak much more this kind of uncorrelated ultrasonic noise than DSD DACs because implementing steep filters without increasing output impedance is quite challenging. So concerns of such modulation noise could be primarily put there.

 


Signalyst - Developer of HQPlayer

Pulse & Fidelity - Software Defined Amplifiers

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40 minutes ago, Miska said:

DSD noise is quite little concern overall. It is uncorrelated with the music (when using good modulator), so the intermodulation products are also noise - only increase the audio band noise floor.

It becomes a problem if that noise is audible.

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It becomes a problem if that noise is audible... or causes audible effects.


"The overwhelming majority [of audiophiles] have very little knowledge, if any, about the most basic principles and operating characteristics of audio equipment. They often base their purchasing decisions on hearsay, and the preaching of media sages. Unfortunately, because of commercial considerations, much information is rooted in increasing revenue, not in assisting the audiophile. It seems as if the only requirements for becoming an "authority" in the world of audio is a keyboard."

-- Bruce Rozenblit of Transcendent Sound

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1 hour ago, Ralf11 said:

or causes audible effects

 

Whilst not directly audible, it has strong potential to cause audible effects. No downstream circuitry will have zero IMD and IMD performance for all amplifiers falls with increasing frequency.

 

As to the question of whether the noise is correlated or not, its fairly easy to figure out by observation that its anti-correlated. Seeing as a DSD output stage only has two possible levels, its always outputting its full signal level, it has nowhere else to go. When your music is quiet it must follow that the noise is loud and vice-versa.

Edited by opus101
afterthought

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