Taiko Audio SGM Extreme : the Crème de la Crème

Blue58 here, aka Barry, who went thru a long and arduous process deciding btwn these two fuses.
 
I'm testing SGM Extreme.

I used some fuses to Extreme.

The result is interesting.



Best Fuse for SGM Extreme

1st. Beeswax Ultimate Fuse - 100%

2nd. Beeswax Super Fuse - 75%

3rd. Bundle Fuse - 60%

4th. Synergistic Research Blue Fuse - 55%



In case of EVO, Blue fuse is the best.

But in case of Extreme, Blue fuse makes Extreme lose dynamic and bass sound's power.

The best cost-effective fuse is bundle fuse.

It's very nice~!
Hi Esotar,
Interesting result and as spiritofmusic said, it was a long decision. Reason being the Blue needs over 400 hrs to show its strength whereas the Beeswax is pretty good from the start. I settled on the Blue for the SGM while using Beeswax elsewhere. It would be interesting to see if your impression changes with time.
And don’t forget to use a SOTA power cord preferably with Bocchino plugs;)
 
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Hi Esotar,
Interesting result and as spiritofmusic said, it was a long decision. Reason being the Blue needs over 400 hrs to show its strength whereas the Beeswax is pretty good from the start. I settled on the Blue for the SGM while using Beeswax elsewhere. It would be interesting to see if your impression changes with time.
And don’t forget to use a SOTA power cord preferably with Bocchino plugs;)

You're right~!

I need burn-in time.

If I feel difference, I'll post it.

You use GREAT DAC~!!

In case of Formula xHD, I'm recommending JPLAY Femto instead of HQPlayer.

Try it~!!
 
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The MSB doesn't upsample, it plays the native rate of the file

Dacs like the dCS Vivaldi/upsampler do upsample .....

What do you exactly mean by "plays the native rate of the file"? Does it mean that the DAC converter accepts the exact bits of the filer? Or just it does not upsample?
 
It is something I find difficult to understand. Just consider non upsampled redbook, to keep things simple. How can computer latency affect data streams at such low data rates?

Thank you for asking as I would like to propose there is no "problem" in the actual data stream. Data errors almost never occur, if at all. This is however a hard to explain topic. I'm trying to share some of the discoveries we made in this thread, but it's not easy to explain, so bear with me and please keep the questions coming if anything is unclear.

For the original SGM 2015 music server our primary focus was on reducing DAC filter quality influence by providing a higher quality (up sampled) data stream. Especially DSD DAC's could greatly benefit from this method and be provided with a means to convert every source format to DSD. Of course the end result would depend on how the DAC processes higher data rates, most DAC's use different filters for different sampling rates, or they can be user selected, so you would force the DAC to use, or be able to use a different filter this way. Some of the filters and up sampling algorithms provided by HQPlayer which we use for that purpose are so processor intensive you can most definitely not run it on for example a Roon Nucleus, let alone they will run in a DAC's FPGA. A good example it the Chord Dave which is all about high quality filters and boasts a very strong FPGA array but it cannot approach the filter algorithms quality HQPlayer can provide.

Fast forward to this day and age where DAC quality, and DAC filter quality technology has advanced significantly, especially in upper echelon R2R DAC's, we find ourselves in the situation that the benefit of pre-processing the data has either decreased or is gone altogether. Most of these simply sound best being fed native "bit perfect" data rates.

Here we get to the interesting part and your question: "How can computer latency affect data streams at such low data rates?" A natural companion to this question would be: "Why use so much processing power for such low data rates?"

"Low latency" is something computer audiophiles have been hunting since the early Logitech transporter/squeezebox days. I don't think anybody ever really knew why it sounded better, it's sought after in the studio recording scene, but obviously to avoid time related sync issues and audio stream interruptions, but not sound quality afaik. There are several tools available to measure a system's latency to this purpose (referred to as DPC latency, ISR routine execution time, interrupt to process latency etc). It's generally accepted that lower latency sounds better amongst computer audiophiles though.

As I attempted to explain in my previous post on the subject, lowering latencies reduce active processing times. You can view latency as a roadblock that you cannot pass until its removed. Or shifting your transmission into a gear before you can accelerate. During a latency "wait state", a processor, memory module or system bus data path is getting ready to accept data packets. It will be active though, drawing current, transmitting its unavoidable EMI and or RFI spectrum which any electrical component will do. With lower latencies we reduce overall system current draw, EMI, RFI and processing durations. Contrary to what you would expect, you can have lower current draw variations and net overall lower EMI / RFI emissions from higher processing power solutions being minimally loaded then from low processing power solutions being higher loaded. The general view that lower power servers generate less noise then higher power servers BECAUSE they consume less current is wrong in our experience. Of course there are a lot of variables in this equation as it's easier and cheaper to design a low noise power supply for lower current draw requirements so you may very well get better results from a low power solution, especially when using the same power supply. But this is not the design goal of the Extreme.

Now if we can agree on the hypothesis that introducing any type of component into your system can alter your system sound, not restricted to the signal it puts out, it will become a lot easier to explain more differences. No matter if it's a server, a cd transport, an amplifier, a cable, a fuse, a grounding or ground modulating device. In fact why don't we extend the definition of your system to your phone charger, your imac, your WIFI router or your new refrigerator?
 
Thank you for asking as I would like to propose there is no "problem" in the actual data stream. Data errors almost never occur, if at all. This is however a hard to explain topic. I'm trying to share some of the discoveries we made in this thread, but it's not easy to explain, so bear with me and please keep the questions coming if anything is unclear.

Thanks.

For the original SGM 2015 music server our primary focus was on reducing DAC filter quality influence by providing a higher quality (up sampled) data stream. Especially DSD DAC's could greatly benefit from this method and be provided with a means to convert every source format to DSD. Of course the end result would depend on how the DAC processes higher data rates, most DAC's use different filters for different sampling rates, or they can be user selected, so you would force the DAC to use, or be able to use a different filter this way. Some of the filters and up sampling algorithms provided by HQPlayer which we use for that purpose are so processor intensive you can most definitely not run it on for example a Roon Nucleus, let alone they will run in a DAC's FPGA. A good example it the Chord Dave which is all about high quality filters and boasts a very strong FPGA array but it cannot approach the filter algorithms quality HQPlayer can provide.

Easy to understand - I have always considered the SGM2015 a custom product, not a general server.

Fast forward to this day and age where DAC quality, and DAC filter quality technology has advanced significantly, especially in upper echelon R2R DAC's, we find ourselves in the situation that the benefit of pre-processing the data has either decreased or is gone altogether. Most of these simply sound best being fed native "bit perfect" data rates.

Little is know in detail about most upper echelon R2R DACs - manufacturers keep this information confidential. R2R become fashionable.

Here we get to the interesting part and your question: "How can computer latency affect data streams at such low data rates?" A natural companion to this question would be: "Why use so much processing power for such low data rates?"

"Low latency" is something computer audiophiles have been hunting since the early Logitech transporter/squeezebox days. I don't think anybody ever really knew why it sounded better, it's sought after in the studio recording scene, but obviously to avoid time related sync issues and audio stream interruptions, but not sound quality afaik. There are several tools available to measure a system's latency to this purpose (referred to as DPC latency, ISR routine execution time, interrupt to process latency etc). It's generally accepted that lower latency sounds better amongst computer audiophiles though.

Being generally accepted without a proper explanation will just drive us going in circles ... :)

As I attempted to explain in my previous post on the subject, lowering latencies reduce active processing times. You can view latency as a roadblock that you cannot pass until its removed. Or shifting your transmission into a gear before you can accelerate. During a latency "wait state", a processor, memory module or system bus data path is getting ready to accept data packets. It will be active though, drawing current, transmitting its unavoidable EMI and or RFI spectrum which any electrical component will do. With lower latencies we reduce overall system current draw, EMI, RFI and processing durations. Contrary to what you would expect, you can have lower current draw variations and net overall lower EMI / RFI emissions from higher processing power solutions being minimally loaded then from low processing power solutions being higher loaded. The general view that lower power servers generate less noise then higher power servers BECAUSE they consume less current is wrong in our experience. Of course there are a lot of variables in this equation as it's easier and cheaper to design a low noise power supply for lower current draw requirements so you may very well get better results from a low power solution, especially when using the same power supply. But this is not the design goal of the Extreme.

Very true, but it seems all we have now is EMI / RFI emission ...

Now if we can agree on the hypothesis that introducing any type of component into your system can alter your system sound, not restricted to the signal it puts out, it will become a lot easier to explain more differences. No matter if it's a server, a cd transport, an amplifier, a cable, a fuse, a grounding or ground modulating device. In fact why don't we extend the definition of your system to your phone charger, your imac, your WIFI router or your new refrigerator?

Yes, we can agree that introducing some (sorry, not any ) type of component into our system can alter our system sound, not restricted to the signal it puts out .
 
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Thanks.

Easy to understand - I have always considered the SGM2015 a custom product, not a general server.

In it's initial design yes, however it can be easily reconfigured for "general" purpose use.

Very true, but it seems all we have now is EMI / RFI emission …

Can we agree to add powerline harmonics / distortion to the equation?

Yes, we can agree that introducing some (sorry, not any ) type of component into our system can alter our system sound, not restricted to the signal it puts out .

So let's move on to Ethernet networking.

Ethernet-EC.JPG

The attached picture shows how incredible ethernet error correction works. It's been designed to deal with distortion and noise associated with up to 100 meters of unshielded copper cable. (For general interest, shielding mainly affects NEXT). There's also additional error detection / handling on a different layer where unrecoverable data is being retransmitted. However incredible all this does significantly increase ethernet PHY power consumption. And arguably we don't need it's full capabilities for short connections but it is what it is.

Moving on to Ethernet enabled DAC's. You will actually have a complete computing environment running an operating system, cpu, memory and a hardware and software networking stack, endpoint software like a upnp renderer, or Roon endpoint software. This is quite a bit more then a USB or AES/EBU receiver. It will most definitely draw more power and have higher EMI/RFI emissions.

But, from a usability point of view, it's absolutely great as it allows you to stream directly to your DAC from any other networking connected streaming source. However getting it to sound optimal requires quite a bit of tinkering in your networking environment. The often used selling point it nullifies the influence of your source is definitely not true. And I'm as of yet unconvinced it's the way to go to obtain maximum sound quality.
 
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Moving on to Ethernet enabled DAC's. You will actually have a complete computing environment running an operating system, cpu, memory and a hardware and software networking stack, endpoint software like a upnp renderer, or Roon endpoint software. This is quite a bit more then a USB or AES/EBU receiver. It will most definitely draw more power and have higher EMI/RFI emissions.

But, from a usability point of view, it's absolutely great as it allows you to stream directly to your DAC from any other networking connected streaming source. However getting it to sound optimal requires quite a bit of tinkering in your networking environment. The often used selling point it nullifies the influence of your source is definitely not true. And I'm as of yet unconvinced it's the way to go to obtain maximum sound quality.

I agree with Emile totally on this point.

Ethernet DACs are convenient but the sonic performance depend on the implementation.

I have observed that several such DACs actually contain an internal smallish computer motherbroard.
Have the problem of EMI/RFI addition been fully addressed/solved?
I don't think so.
 
SGM Extreme and SGM 2015 EVO have different mechanism each other.

Digital components need to process quickly.

EVO is very fast.

Look at this picture :

EVO Speed.png

Processing speed is 9.3x and average 9.8x

But SGM Extreme is very strange.

Please look at this picture :


Extreme Speed.JPG


Extreme's processing speed is just 4.7x.

This speed is inner storage files playback speed.

If I play files from NAS, processing speed is 3.4x.

If I use Tidal service, processing speed is 3.0x.


BUT

Extreme sound is flawless.

I never request Extreme setting to Emile.

Nevertheless, Extreme quality is super fantastic~!!

Although I'm not engineer, I know digital products must be fast, because of noise and vibration.

Audio signal have to be sent as soon as possible.

But SGM Extreme processing speed is very slow.

Extreme has many secrets~!
 
SGM Extreme and SGM 2015 EVO have different mechanism each other.

Digital components need to process quickly.

EVO is very fast.

Look at this picture :

View attachment 48912

Processing speed is 9.3x and average 9.8x

But SGM Extreme is very strange.

Please look at this picture :


View attachment 48913


Extreme's processing speed is just 4.7x.

This speed is inner storage files playback speed.

If I play files from NAS, processing speed is 3.4x.

If I use Tidal service, processing speed is 3.0x.


BUT

Extreme sound is flawless.

I never request Extreme setting to Emile.

Nevertheless, Extreme quality is super fantastic~!!

Although I'm not engineer, I know digital products must be fast, because of noise and vibration.

Audio signal have to be sent as soon as possible.

But SGM Extreme processing speed is very slow.

Extreme has many secrets~!

Why is there upsampling? I need bit perfect.
 
Why is there upsampling? I need bit perfect.
I guess he likes how that sounds with his particular DAC. Just because the Extreme was primarily designed for maximum bit perfect playback performance does not mean it cannot up sample if so desired. There is a whole range of DAC's out there which still benefit from being served an up sampled signal.

See for example: http://archimago.blogspot.com/2019/02/measurements-look-at-hqplayer-325.html
And/or: http://archimago.blogspot.com/2019/02/musings-measurements-on-why-2496.html
 
I guess he likes how that sounds with his particular DAC. Just because the Extreme was primarily designed for maximum bit perfect playback performance does not mean it cannot up sample if so desired. There is a whole range of DAC's out there which still benefit from being served an up sampled signal.

See for example: http://archimago.blogspot.com/2019/02/measurements-look-at-hqplayer-325.html
And/or: http://archimago.blogspot.com/2019/02/musings-measurements-on-why-2496.html

Maybe for less able DACs. I don’t see any top end DACs like MSB, dcs or TotalDAC benefits from upsampling. BTW, I have been using HQPlayer on and off. I don’t think highly of HQPlayer upsampling capabilities.
 
Maybe for less able DACs. I don’t see any top end DACs like MSB, dcs or TotalDAC benefits from upsampling. BTW, I have been using HQPlayer on and off. I don’t think highly of HQPlayer upsampling capabilities.

Yes I'd say for modern DAC's, once you cross the ~10-15K retail barrier, up sampling tends to not improve performance anymore. It sure doesn't for our resident Totaldac D1-12mk2.
 
There have been three interesting posts on Hiendy.com in Hong Kong from audiophiles visiting Ben Lau's showroom where the Extreme has been playing to the Totaldac D1-12 Mk 2

[CAS] 2019 Chinese New Year Play Test Report 2: SGM Extreme Server

Post # 48

After the completion of C K Keung’s report, I finally had the opportunity to meet the Extreme in Ben Sir’s showroom.

The internal structure of the chassis and the details of the components can only be described as shocking. This is an unprecedented product. The chassis is also a lot better than the first generation.

With the Totaldac D1 using the direct connection, using clean power supply configuration. We listened to Tidal, and Qobuz streaming and local file, very transparent stereo, very good sound density.

The Rachmaninov Music for Two pianos performed by Argerich was really shocking in its flow. The black church chorus is alive and well in all levels, and the rise and fall is harmonious. Even Chen Yixun's concerts are very involving vocal performances.

Absolutely a CAS equipment in a class of its own.

Post # 53

On Saturday to Ben Sir, seeing the Extreme chassis, appreciate the appearance first, and ... the large size is good.

Then Ben Sir says the heat radiator is made of pure copper, and is the reason it to becomes 43kg heavy, but, the chassis never becomes hot.

Starting to listen, and discovering that the sound is unprecedented, the background is very dark, the treble is so transparent, and bass is to die for

The most amazing is the soundstage width and very good depth, like a frog sitting in the bottom of a well, I believed that only the traditional top-level turntable could achieve first class depth of the soundstage.


Post # 56

I haven't been to Ben Sir's showroom for a while. Two days ago, I finally couldn't help but go to listen to the new “underground king” the SGMS extreme. As expected, this is the most “acoustics” I have ever heard in Ben Sir’s showroom. SGMS extreme is indeed a breakthrough product, completely reborn.

I listened to the multi-cut Manager with Roon and Jplay. I heard the sound of no digital high-frequency noise. The sound field was stable like a mountain and the density was very high, and the timbre and image of various instruments were very coherently distributed over the soundfield, filling Ben Sir's entire showroom, there was no cloud of false harmony. A variety of musical instruments exist together but each has its own power. It should be lively and lively. It should be authoritative and authoritative, fully showing the living music.
Ben Sir instantly converts different profiles to suit the tastes of different listeners. The development of digital music is close to the end game.

Congratulations to Emile and Edward for their efforts to solve a number of digital audio problems. Achieve low latency with separate CPU for Roon and OS, fanless operation, no more SSD, use PCIe storage and not SATA (noise source ?) and so on. Countless problems have come up with solutions. Bravo.

It is recommended that music audiophiles who are interested in CAS take time to go up and listen to it, and understand the realm of CAS development.
 
There have been three interesting posts on Hiendy.com in Hong Kong from audiophiles visiting Ben Lau's showroom where the Extreme has been playing to the Totaldac D1-12 Mk 2

Hello Edward,

The dac paired with the Extreme in Ben's showroom (Volent HK) has all along been the Totaldac Direct+Totaldac Reclocker combo.

I can imagine if it's a Totaldac Twelve Mk2 instead, the performance shall be at least a league higher! haha
 
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I ordered Taiko Audio [[DAIZA]] and I'm waiting for it.

BTW, my HI FI STAY audio rack [Concert Master] has no bottom plates.

For testing DAIZA, I bought normal bottom plate by HI FI STAY.

P1000374.JPG

P1000375.JPG

Looks good~!

Taiko Audio doesn't have many products, but all is fantastic~!!

Emile is Great Engineer and Super Kind business man.

I believe DAIZA will be also great~!!

I look forward to test DAIZA~!!!
 
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Congratulations to Emile and Edward for their efforts to solve a number of digital audio problems. Achieve low latency with separate CPU for Roon and OS, fanless operation, no more SSD, use PCIe storage and not SATA (noise source ?) and so on. Countless problems have come up with solutions. Bravo.

.
This part I do not understand. If they do not have SSD, how is music stored internally?
 

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