Sonore microRendu

[m3lraaHnevetS]

Hi Jesus, I just spoke to T+A support and they state that to send DSD >128 via Linux to their DAC8 DSD you need to use the new 4.9 Linux kernel...any change we will be able to use the microRendu regarding this...?

[vortecjr]

Hi Jesus, I just spoke to T+A support and they state that to send DSD >128 via Linux to their DAC8 DSD you need to use the new 4.9 Linux kernel...any change we will be able to use the microRendu regarding this...?

 

It's not that simple. Email me the contact information of the person you talked to at T+A. My email is [email protected]

[m3lraaHnevetS]

The person I spoke to earlier had to go and ask someone else. I just called back and got a yet an other person on the phone, this one wanted the serial number of my DAC to sort out what it can or can not do. He will call me back tomorrow. to be continued...I will of course try and get a name and email address too.

[austinpop]

Hi @vortecjr ,

 

I'm new to the mRendu, but recently was lent a unit for several weeks to try out. I have started reporting my impressions on this thread: http://www.computeraudiophile.com/f22-networking-networked-audio-and-streaming/auralic-aries-mini-vs-sonore-microrendu-vs-soul-music-sms-200-listening-impressions-31499/

 

Kudos on a very nicely engineered product!

 

I did run into a couple of functional issues that I wanted to raise here. It's certainly possible that it is user error, but could also be an opportunity for improvement. I am running version 2.3, and am using MPD/DLNA mode. My music server is MinimServer, running both on a Synology NAS, and also on a W10 Enterprise box.

 

The functional issues:

 

 

1. My DAC, the Ayre Codex, requires DSD to be delivered as DoP. I believe I have the mR configured correctly, by unchecking the Enable Direct DSD Support checkbox. However, I several times encountered bursts of white noise when selecting a DSD64 track to play. I cannot reliable reproduce it, but the highest probability of it happening is when interrupting a stream of PCM by selecting a DSD track.
    
   I was able to fix it by enabling DoP encoding upstream in the MinimServer itself, by setting stream.transcode to "dsf:dopwav, dff:dopwav".
    
   Perhaps ther are opportunities to improve your handling of DoP?
    
    
   
2. I have also been evaluating HQPlayer, so frequently want to switch from MPD/DLNA to HQPlayer NAA mode. While the switch from MPD/DLNA to NAA is very quick and 100% reliable, the reverse is not the case.
    
   Invariably, when I am in NAA mode, and click on the MPD/DLNA button, the operation just hangs. Eventually I kill the page and refresh it. At this point, it shows a gray "Unknown" next to the NAA button. Finally, another click on the MPD/DLNA button succeeds, but the operation takes a long time - at least 20-30 seconds.
    
   Is this a known issue? Or is there a workaround?
   

Please let me know. Thanks.

[d_elm]

Hi @vortecjr ,

 

I'm new to the mRendu, but recently was lent a unit for several weeks to try out. I have started reporting my impressions on this thread: http://www.computeraudiophile.com/f22-networking-networked-audio-and-streaming/auralic-aries-mini-vs-sonore-microrendu-vs-soul-music-sms-200-listening-impressions-31499/

 

Kudos on a very nicely engineered product!

 

I did run into a couple of functional issues that I wanted to raise here. It's certainly possible that it is user error, but could also be an opportunity for improvement. I am running version 2.3, and am using MPD/DLNA mode. My music server is MinimServer, running both on a Synology NAS, and also on a W10 Enterprise box.

 

The functional issues:

 

 

1. My DAC, the Ayre Codex, requires DSD to be delivered as DoP. I believe I have the mR configured correctly, by unchecking the Enable Direct DSD Support checkbox. However, I several times encountered bursts of white noise when selecting a DSD64 track to play. I cannot reliable reproduce it, but the highest probability of it happening is when interrupting a stream of PCM by selecting a DSD track.
    
   I was able to fix it by enabling DoP encoding upstream in the MinimServer itself, by setting stream.transcode to "dsf:dopwav, dff:dopwav".
    
   Perhaps ther are opportunities to improve your handling of DoP?
    
    
   
2. I have also been evaluating HQPlayer, so frequently want to switch from MPD/DLNA to HQPlayer NAA mode. While the switch from MPD/DLNA to NAA is very quick and 100% reliable, the reverse is not the case.
    
   Invariably, when I am in NAA mode, and click on the MPD/DLNA button, the operation just hangs. Eventually I kill the page and refresh it. At this point, it shows a gray "Unknown" next to the NAA button. Finally, another click on the MPD/DLNA button succeeds, but the operation takes a long time - at least 20-30 seconds.
    
   Is this a known issue? Or is there a workaround?
   

Please let me know. Thanks.

 

I am also using Ayre Codex with uR and NAS. In the BubbleUPnP control app playlist I can switch between 352.8k and DSD64, either way, and there is only a one second pause, no noise. On the NAS I run minumServer and BubbleUPnP server.

 

 

Sent from my iPad using Computer Audiophile

[austinpop]

I am also using Ayre Codex with uR and NAS. In the BubbleUPnP control app playlist I can switch between 352.8k and DSD64, either way, and there is only a one second pause, no noise. On the NAS I run minumServer and BubbleUPnP server.

 

 

Sent from my iPad using Computer Audiophile

 

Good to know. Yeah, I can only report what I heard.

 

I didn't run Bubble UPnP, either on the mR, or on the NAS. As for control apps, that's another area I've been exploring, as with the Aries Mini, I've gotten used to the Lightning DS app.

 

I've been alternating between Linn Kazoo, Lumin, and Audionet iMM. I honestly don't remember which app I was using when this happened. And setting dopwav in Minim has completely solved the issue, as I guess the mR only sees a WAV stream and doesn't have to do the DoP encoding.

[barrows]

Switching between PCM and DSD on the fly can trip up some DACs. It is very hard for developers to "fix" these issues in the player code (µR), as every DAC can handle things a little bit differently.

My best advice is to not play mixed playlists of DSD and PCM files, to me this is just risky. Be aware of when you are switching between PCM and DSD playback and stop playback and than start playback with the new file.

 

I am not trying to suggest that this is a DAC issue, it is an everything issue, but taking some simple precautions in how you handle playback (user) can avoid all problems. If it were vinyl, no one would be noting that things do not sound "right" when I put on a 45 after playing a 33 1/3 record...

[austinpop]

Switching between PCM and DSD on the fly can trip up some DACs. It is very hard for developers to "fix" these issues in the player code (µR), as every DAC can handle things a little bit differently.

My best advice is to not play mixed playlists of DSD and PCM files, to me this is just risky. Be aware of when you are switching between PCM and DSD playback and stop playback and than start playback with the new file.

 

I am not trying to suggest that this is a DAC issue, it is an everything issue, but taking some simple precautions in how you handle playback (user) can avoid all problems. If it were vinyl, no one would be noting that things do not sound "right" when I put on a 45 after playing a 33 1/3 record...

Good points, but please keep in mind I am not a novice, and don't need "the talk."

 

If you read my review, I have very good things to say about the mR. However, I have not had this issue with the Aries Mini, which is why I thought this worth reporting.

 

Clearly, there are ways for endpoints to be more "defensively" coded to minimize these effects, perhaps by longer mutes. I don't know the code. Also, it's possible that the Aries platforms use of buffering reduces this as well.

 

Whatever the reason, I thought you guys wold be interested. I can't tell from your sig if you're affiliated with Sonore.

 

As I said, I found a workaround, and moved on. With the dopwav setting in Minim, I can happily switch between PCM and DSD in midstream.

[barrows]

Good points, but please keep in mind I am not a novice, and don't need "the talk."

 

If you read my review, I have very good things to say about the mR. However, I have not had this issue with the Aries Mini, which is why I thought this worth reporting.

 

Clearly, there are ways for endpoints to be more "defensively" coded to minimize these effects, perhaps by longer mutes. I don't know the code. Also, it's possible that the Aries platforms use of buffering reduces this as well.

 

Whatever the reason, I thought you guys wold be interested. I can't tell from your sig if you're affiliated with Sonore.

 

As I said, I found a workaround, and moved on. With the dopwav setting in Minim, I can happily switch between PCM and DSD in midstream.

 

Huh, please take a closer look at my signature, it appears clear to me? Anyway, my point was to suggest that every DAC is different in how it handles this switch, so no matter how you code the player, it may work perfectly with one DAC, and a little differently with another DAC.

For example, the Aries Mini might be able to handle this elegantly with the Ayre, but could still have problems with a different DAC, and the µR could handle this perfectly with my DAC, but exhibit a problem with another DAC.

No developer can test with every different DAC interface on the market, and even if one could, it might be impossible to make one code which works perfectly with every DAC-that is if one expects to play mixed PCM/DSD playlists gapless.

 

I menat no criticism to you, and my response was meant as a public service, so was intended generally to suggest that if people try to trip up sample rate switching with any device, they probably will succeed, given the huge number of variables involved (between DAC choice, interface choice, playback softwares, and settings).

[austinpop]

Huh, please take a closer look at my signature, it appears clear to me? Anyway, my point was to suggest that every DAC is different in how it handles this switch, so no matter how you code the player, it may work perfectly with one DAC, and a little differently with another DAC.

For example, the Aries Mini might be able to handle this elegantly with the Ayre, but could still have problems with a different DAC, and the µR could handle this perfectly with my DAC, but exhibit a problem with another DAC.

No developer can test with every different DAC interface on the market, and even if one could, it might be impossible to make one code which works perfectly with every DAC-that is if one expects to play mixed PCM/DSD playlists gapless.

 

I menat no criticism to you, and my response was meant as a public service, so was intended generally to suggest that if people try to trip up sample rate switching with any device, they probably will succeed, given the huge number of variables involved (between DAC choice, interface choice, playback softwares, and settings).

 

Duly noted.

[vortecjr]

Hi @vortecjr ,

 

I'm new to the mRendu, but recently was lent a unit for several weeks to try out. I have started reporting my impressions on this thread: http://www.computeraudiophile.com/f22-networking-networked-audio-and-streaming/auralic-aries-mini-vs-sonore-microrendu-vs-soul-music-sms-200-listening-impressions-31499/

 

Kudos on a very nicely engineered product!

 

I did run into a couple of functional issues that I wanted to raise here. It's certainly possible that it is user error, but could also be an opportunity for improvement. I am running version 2.3, and am using MPD/DLNA mode. My music server is MinimServer, running both on a Synology NAS, and also on a W10 Enterprise box.

 

The functional issues:

 

 

1. My DAC, the Ayre Codex, requires DSD to be delivered as DoP. I believe I have the mR configured correctly, by unchecking the Enable Direct DSD Support checkbox. However, I several times encountered bursts of white noise when selecting a DSD64 track to play. I cannot reliable reproduce it, but the highest probability of it happening is when interrupting a stream of PCM by selecting a DSD track.
    
   I was able to fix it by enabling DoP encoding upstream in the MinimServer itself, by setting stream.transcode to "dsf:dopwav, dff:dopwav".
    
   Perhaps ther are opportunities to improve your handling of DoP?
    
    
   
2. I have also been evaluating HQPlayer, so frequently want to switch from MPD/DLNA to HQPlayer NAA mode. While the switch from MPD/DLNA to NAA is very quick and 100% reliable, the reverse is not the case.
    
   Invariably, when I am in NAA mode, and click on the MPD/DLNA button, the operation just hangs. Eventually I kill the page and refresh it. At this point, it shows a gray "Unknown" next to the NAA button. Finally, another click on the MPD/DLNA button succeeds, but the operation takes a long time - at least 20-30 seconds.
    
   Is this a known issue? Or is there a workaround?
   

Please let me know. Thanks.

 

I appreciate the compliment!

 

This is a complex project and sometimes our intentions are not well documented:)

 

1. SonicOrbiter can accept streams from DLNA/UPnP servers and play them via the MPD/DLNA output mode. The unit can also play local files via the MPD/DLNA output mode. As such, if your server supports DoP streaming you don't enable DoP in Settings and instead do so at the server. If you have local DSD files and want to use DoP you enable it in Settings. This approach is whats best overall.

 

2. When you switch from one output mode to another there are various apps being turned off and various apps being turned on. This was done so the unit behaves as if it was designed specifically for one output mode. The sequence is different between the selections and each takes it a certain amount of time. Some can have built in delays for specific reasons. If NAA is not initially properly, for you, then you have to select it again. Our intent here was not that people would switch back and forth for A/B testing. We figure that people would set and forget things so there is no optimization here.

[m3lraaHnevetS]

It's not that simple. Email me the contact information of the person you talked to at T+A. My email is [email protected]

 

Hi Jesus, spoke to T+A again and it turns out that there is nothing to be expected from them in terms of new firmware...

 

[vortecjr]

Hi Jesus, spoke to T+A again and it turns out that there is nothing to be expected from them in terms of new firmware...

 

[ATTACH=CONFIG]33010[/ATTACH]

 

Do you have microRendu? If so, we can explore some things.

[m3lraaHnevetS]

Do you have microRendu? If so, we can explore some things.

 

I do, sure tell me

[vortecjr]

I do, sure tell me

With the DAC connected post the output from Apps / DAC Diagnostics.

[m3lraaHnevetS]

With the DAC connected post the output from Apps / DAC Diagnostics.

 

[vortecjr]

[ATTACH=CONFIG]33011[/ATTACH]

 

I don't see why you couldn't stream DSD128 via DoP right now with the microRendu. If the device supports DSD128 via DoP you should be good to go. You can't stream native DSD because that device does not support the correct format needed for Linux.

[m3lraaHnevetS]

I don't see why you couldn't stream DSD128 via DoP right now with the microRendu. If the device supports DSD128 via DoP you should be good to go. You can't stream native DSD because that devices does not support the correct format needed for Linux.

 

DSD128 via DoP works fine, I am still hoping to do DSD512 one day.. will it ever happen with the T+A DAC8 DSD via the mR?

[vortecjr]

DSD128 via DoP works fine, I am still hoping too do DSD512 one day.. will it ever happen with the T+A DAC8 DSD via the mR?

 

This is not a question for me to answer. Right now you are limited by the constraints of the T+A firmware. The microRendu can stream those rates just fine. Elektroakustik needs to update their firmware to support it.

[m3lraaHnevetS]

This is not a question for me to answer. Right now you are limited by the constraints of the T+A firmware. The microRendu can stream those rates just fine. Elektroakustik needs to update their firmware to support it.

 

I understand. can't keep from wondering what it will sound like. I bet it will better my simple noisy HTPC...maybe one day...guess I will sell the mR now that it is still hot

[vortecjr]

Jay Luong of Audio Bacon posted a review called 'Sonore microRendu & UpTone Audio UltraCap LPS-1 Review'

 

You can find the review here:

https://audiobacon.net/2017/02/03/sonore-microrendu-review/

[marksas3]

Jay Luong of Audio Bacon posted a review called 'Sonore microRendu & UpTone Audio UltraCap LPS-1 Review'

 

You can find the review here:

https://audiobacon.net/2017/02/03/sonore-microrendu-review/

 

I just replaced the TeraDak with the LPS-1 and honestly I would not describe any 'night and day' differences. The TeraDak unit is a decent upgrade in its own right. Granted I need more time with the LPS-1 to pass proper judgement, but I can already attest that the combo sounds really really good....love my mR

[Johnseye]

The Sonore microRendu project is in essence two projects in one.

The first project is the Sonicorbiter operating system. Sonicorbiter is a collaboration between Sonore by Simple Design and Small Green Computer. The Sonicorbiter operating system is a preconfigured renderer / endpoint / player with selectable output modes. We have spent countless hours behind the scenes working with the various developers of its Linux applications to assure they work well and deliver the kind of performance you demand.

 

The second project is the hardware platform (code named toaster). The toaster is a collaboration between Sonore by simple Design, Small Green Computer and John Swenson. As part of our research and development, we have tested various hardware solutions and quite frankly we were not impressed. In order to meet our goals and push our interests in a new direction we needed to start fresh. This hardware has taken John over a year to design and incorporates several key features. For example: There are regulated zones on the Sonore microRendu providing unprecedented clean power to all components. There are very low jitter oscillators - one for the CPU and one for the USB hub that generate a completely new USB data signal to feed your device. Make no mistake, this product is intended to give you goosebumps.

 

Jesus R

 

Hey there Jesus, I've just taken the plunge and purchased an microRendu from you. I'm trying to understand what I'm inserting into the chain and what it does. I've read quite a bit about the device and thought it worth giving it a try without fully understanding what it is doing with the data packets coming from the media server. I'm an IT Systems Engineer by trade, but now run the IT department for my company so please be technical in your explanation. I'd really like to understand if the packets are manipulated in any way, and if so how. If they are not, is the microRendu merely filtering any upstream noise that's come across the copper?

 

Thanks

[bunno77]

Hey there Jesus, I've just taken the plunge and purchased an microRendu from you. I'm trying to understand what I'm inserting into the chain and what it does. I've read quite a bit about the device and thought it worth giving it a try without fully understanding what it is doing with the data packets coming from the media server. I'm an IT Systems Engineer by trade, but now run the IT department for my company so please be technical in your explanation. I'd really like to understand if the packets are manipulated in any way, and if so how. If they are not, is the microRendu merely filtering any upstream noise that's come across the copper?

 

Thanks

Sounds like you're asking for intellectual property. It's a data renderer that connects to a dac in a hifi system. If you believe it sounds good and you can play media from where you wish then its a good investment right?

 

Not many manufacturers explain exactly how every little piece of their product functions. These guys have explain that which they wish on their website.

[Johnseye]

Sounds like you're asking for intellectual property. It's a data renderer that connects to a dac in a hifi system. If you believe it sounds good and you can play media from where you wish then its a good investment right?

 

Not many manufacturers explain exactly how every little piece of their product functions. These guys have explain that which they wish on their website.

Leave the intellectual property out of it. I'm not looking for secrets. I want to know what it actually does. Asking whether it changes the data packet in any way doesn't give away intellectual property. The physical components of the device speak for themselves. I'd like to know what the software is doing to the packets if anything at all.