Print

Please login or register.
1 Hour 1 Day 1 Week 1 Month Forever
Login with username, password and session length

[AndrewFG]

Two things..

1. Some DSP effects change the sample rate, bit depth, or channel count to values different than what the renderer is expecting, and this causes playback not to work. Convolution is a quite complex DSP effect and so may indeed cause this problem. Does it play if you use no effect at all? And does it play if you use a simple DSP effect like volume leveling?

2. Not very many renderers support PCM 24 bit. So try it with PCM 16 bit No Header (L16) first.

[LHH]

Two things..

1. Some DSP effects change the sample rate, bit depth, or channel count to values different than what the renderer is expecting, and this causes playback not to work. Convolution is a quite complex DSP effect and so may indeed cause this problem. Does it play if you use no effect at all? And does it play if you use a simple DSP effect like volume leveling?

2. Not very many renderers support PCM 24 bit. So try it with PCM 16 bit No Header (L16) first.

1. It plays fine with no DSP Studio effects (and plays fine with them selected as well -- it's just that they fail to actually activate). Outside of the "DSP Studio" window, some processing works correctly -- for instance, if I select "Sample rate: 192000 hz" within the advanced section of the "DLNA Servers" window, the audio is upsampled and sent to my renderer at 192khz. However, if I set the sample rate in the "DLNA Servers" window to "Same as source", and resample using the "DSP Studio Output Format" window, no resampling actually occurs. Likewise, setting the output to PCM 24bit in the "DLNA Servers" window works fine (my renderer receives the audio files in 24bit), but activating the Equalizer or Convolution etc. within DSP Studio does nothing at all (the audio still plays, but the selected processing is not applied).

2. The Yamaha NP-S2000 is compatible with PCM up to 24/192, and you can set the display to show the sample rate and bit depth it is receiving. I've attached a photo. I tried setting the output in the "DLNA Servers" window to "PCM 16 bit No Header" to see if that helped. Playback works fine (as it does at 24bit), but just as with 24bit, none of the processing selected within the "DSP Studio" window is applied.

[AndrewFG]

As mentioned in my prior post, if DSP were to change sample rate, bit depth, or channel count, it would be sending a stream with different audio parameters than it had previously "advertised" to the renderer via the UPNP mechanisms, and this inconsistency would break the playback entirely.

Based on your symptoms, it sounds like this version of MC is detecting such potential inconsistencies and rather than letting the playback break entirely, it is instead suspending the DSP so that at least you hear something...

[LHH]

As mentioned in my prior post, if DSP were to change sample rate, bit depth, or channel count, it would be sending a stream with different audio parameters than it had previously "advertised" to the renderer via the UPNP mechanisms, and this inconsistency would break the playback entirely.

Based on your symptoms, it sounds like this version of MC is detecting such potential inconsistencies and rather than letting the playback break entirely, it is instead suspending the DSP so that at least you hear something...

I don't think that can be the problem, as it is not only the processing that has the potential to change the sample rate, bit depth, or channel count that fails to activate. Nothing within the DSP Studio window activates -- the equalizer, for instance, doesn't work. If MC has decided that there is a potential inconsistency when I am using the equalizer, then MC is being a little overzealous about maintaining consistency, no?

[AndrewFG]

Ok. You need to wait for a response from bob..

[bob]

Unfortunately I still cannot get any of the DSP studio settings to work over DLNA (equalizer included). I'm attaching a screenshot of my settings -- I *think* I have everything set correctly.

The settings look correct.
Would you verify that the equalizer isn't working? I just tested it with your settings and it worked fine (I sent all the bands to minimum except for the first two which makes a unmissable horrible sound).
Note that a converted file is cached. You should stop and restart MC when changing the settings to make sure you aren't using a cached file conversion (or select a different file to play).

[LHH]

The settings look correct.
Would you verify that the equalizer isn't working? I just tested it with your settings and it worked fine (I sent all the bands to minimum except for the first two which makes a unmissable horrible sound).
Note that a converted file is cached. You should stop and restart MC when changing the settings to make sure you aren't using a cached file conversion (or select a different file to play).

I've tried a few things and am getting random results. I can't seem to get the equalizer to work when the output is set to "same as source" in the "DLNA Servers" window. Setting the sample rate in the "DLNA Servers" window to any one sample rate (from 44.1 to 192k) results in the equalizer working sometimes (although I guess this could be a coincidence).

I have no idea what's going on:

-- Restarting MC or the Media Server seems to make no difference in whether the equalizer works or not.
-- When changes are made in the DSP Studio Equalizer window, those settings are sometimes saved and sometimes not. On reopening DSP Studio the settings may or may not be the same as how they were left.
-- Irrespective of whether the equalizer is working (and music is playing), the DSP Studio Equalizer will display "Playback stopped or current playback doesn't support processing".

In contrast, changing the sample rate or bit depth in the DLNA Servers window works every time, immediately upon playing a new file.

[bob]

You are accessing the DSP studio in the Advanced Audio settings in the DLNA server, correct?
I just tried all combinations of Volume leveling and sample rate (under advanced, not in DSP studio settings) with the EQ on and it worked fine.
Perhaps your renderer is caching files?

[ksalno]

Hi. I'm running MC 21.0.23 on Windows 10. Has there been any progress in getting the DSP Studio automated down sampling to work? Or is it still necessary to manually change the Advanced>Sample Rate setting to stream a DXD file to a network renderer that can only handle 192K?

[bob]

Hi. I'm running MC 21.0.23 on Windows 10. Has there been any progress in getting the DSP Studio automated down sampling to work? Or is it still necessary to manually change the Advanced>Sample Rate setting to stream a DXD file to a network renderer that can only handle 192K?

That change is still in the pipeline. You must use the Advanced>Sample Rate setting at this time.

[ksalno]

Thanks, Bob. Unfortunately, changing the sample rate that way only seems to work for PCM files. My DAC will accept only single-rate DSD files, so I typically use DSP Studio to down sample anything higher but when I set the Sample Rate under Advanced to 176K, for example, my DSD files won't play.

[preproman]

I'm just now getting my feet wet with the Media Network part of JRiver.

I'm using the Lumin A-1 Network Streamer with the Lumin app as the controller.  

While typing these questions I'm at work so I can't test different things out until I get home.

So while configuring the Media Network in JRiver, I see Generic DLNA and I see Audiophile 24 bit.  Is this saying by using the Generic DLNA my 24 bit files will not play and I have to use the Audiophile 24 bit? - and how about DSD files?  The Lumin A-1 can play up to DSD64 2.8MH with formats of DSF, DIFF and DoP If I'm not mistaken.

What are the configurations I should be using if I want the play "all files" and formats from Red Book and HiRes (AIFF, FLAC, WAV) to DSD64 (DSF and DiFF)?

I will try any and all suggestion when I get off work.

Thanks

[zappa]

I'm encountering the same problem reported earlier in this thread with convolution and DLNA.

I have a CAPS v4 Cortes running Win 7 Pro x64 and MC 21.0.15 configured as a DLNA server. This machine uses a Xeon CPU and has a JRMark of around 4600.
I'm using an AURALiC Aries as a DLNA renderer. They are connected over a wired gigabit network. MC is pushing music to the Aries.

I've been experimenting with DSP on the DLNA server. Gapless playback works fine, even with a dozen or more parametric EQ filters. When using convolution playback is not gapless; the gap between tracks is short - a small fraction of a second, but definitely noticeable on music that's continuous across multiple tracks.

Monitoring the performance of the server, the CPU isn't breaking a sweat (max 12%) during playback - so I don't think this is due to hardware limitations.

Has anyone found a solution or workaround to this? (I mean a workaround other than running convolution on a client PC or pre-convolving all of my music files.)

Thanks,
-Brent

Hi everybody.

Is this problem resolved (When using convolution the playback is not gapless) ?

Thanks in advance and nice day for everyone 

[zappa]

I'm encountering the same problem reported earlier in this thread with convolution and DLNA.

I have a CAPS v4 Cortes running Win 7 Pro x64 and MC 21.0.15 configured as a DLNA server. This machine uses a Xeon CPU and has a JRMark of around 4600.
I'm using an AURALiC Aries as a DLNA renderer. They are connected over a wired gigabit network. MC is pushing music to the Aries.

I've been experimenting with DSP on the DLNA server. Gapless playback works fine, even with a dozen or more parametric EQ filters. When using convolution playback is not gapless; the gap between tracks is short - a small fraction of a second, but definitely noticeable on music that's continuous across multiple tracks.

Monitoring the performance of the server, the CPU isn't breaking a sweat (max 12%) during playback - so I don't think this is due to hardware limitations.

Has anyone found a solution or workaround to this? (I mean a workaround other than running convolution on a client PC or pre-convolving all of my music files.)

Thanks,
-Brent

Hi everybody.

Is this problem resolved (When using convolution the playback is not gapless) ?

Thanks in advance and nice day for everyone  

Hi,

Nobody knows if this problem was resolved ?
I would like to know about it before purchasing the JRiver.

Thanks in advance and nice day for everybody here 

[Archimago]

Using JRiver 21 convolution filter currently and streaming to ODROID-C2 with Volumio via DLNA/UPnP.

Sadly, still not gapless with convolution turned on but gapless otherwise.

Is the gap a reflection of the latency induced by the FIR filter? Would it be possible to calculate the latency based on tap length and samplerate, and start playback of the next track past the expected latency period? Maybe add a checkbox to activate this compensation?

Unfortunate annoyance...

[AndrewFG]

Using JRiver 21 convolution filter currently and streaming to ODROID-C2 with Volumio via DLNA/UPnP.

Sadly, still not gapless with convolution turned on but gapless otherwise.

Is the gap a reflection of the latency induced by the FIR filter? Would it be possible to calculate the latency based on tap length and samplerate, and start playback of the next track past the expected latency period? Maybe add a checkbox to activate this compensation?

Unfortunate annoyance...

An interesting question..

The way gapless works, is that the Control Point provides the Renderer with two track URLs -- namely the URL of the current track, and the URL of the "next" track. This enables the Renderer to start downloading the next track into a buffer while it is playing the current track.

At least that is the theory. However many renderers do not really start downloading the next track until they have finished playing the current one. This rather defeats the point of "gapless" of course. Many renderer manufacturers (marketing departments thereof) like to pretend that supporting SetNextAVTransportURI means that they did enough for them to claim gapless support. Hey ho.

So, if your renderer is one of the ones alluded to above, then yes, the delay could be due to the DSP filter latency, and no, there is nothing that MC could do about that.

[zappa]

Thanks for your reply.

So, if I have well understood, you are saying that the problem is on the renderer's side.

Some renderers, already reported in this thread, cannot do it (Auralic ARIES, ODROID-C2+Volumio and even the MC20 configured as renderer).

Do you know, please, the renderers that can playback gapless using JRiver convolution filter on the server side ?

Thanks in advance and have a nice day.

[AndrewFG]

So, if I have well understood, you are saying that the problem is on the renderer's side.

Yes. I am afraid that is indeed the case.

Some renderers, already reported in this thread, cannot do it (Auralic ARIES, ODROID-C2+Volumio and even the MC20 configured as renderer).

Indeed. There is a basic test for a renderer's capability to do gapless playback -- namely whether the renderer supports the optional UPnP command method called "SetNextAVTransportURI". If a renderer does not support this method then it certainly cannot to do gapless playback. But unfortunately supporting the "SetNextAVTransportURI" method is essential, but not sufficient.

There are some renderers (like yours) that do support "SetNextAVTransportURI" but do not start downloading the next track into a buffer while the current track is playing. And since they do not have the next track pre- buffered, they are not actually capable of making a smooth gapless transition between one track and the next. Often the gap is shorter than you would get with regular gapped play, but it is not totally gapless..

Do you know, please, the renderers that can playback gapless using JRiver convolution filter on the server side ?

Over the last years, people have been using my Digital media Renderer Analyser (DMRA) to test their renderers. If you look at the DMRA web page ( http://www.whitebear.ch/dmra ) under the heading "Database of Renderer Test Reports" there is a link where you can download the log reports of all the renderers so far submitted. The DMRA tests if the "SetNextAVTransportURI" method is supported, but admittedly it does not test if the renderer does the also essential pre- buffering of the next track...

[Archimago]

Andrew:
Given than Volumio does actually handle gapless without convolution DSP active, I assume there is potentially pre-buffering of the next track happening, right?

I'm thinking the gap is just the latency of the FIR filter (typically 0.125-0.75sec for 64k and 128k tap length filters depending on samplerate)... Perhaps if JRiver presented the audio for the next track where it actually starts after the latency based on FIR filter length and expected latency just like how it synchronizes audio/video playback, perhaps that's what's needed???

[AndrewFG]

Andrew:
Given than Volumio does actually handle gapless without convolution DSP active, I assume there is potentially pre-buffering of the next track happening, right?

I'm thinking the gap is just the latency of the FIR filter (typically 0.125-0.75sec for 64k and 128k tap length filters depending on samplerate)... Perhaps if JRiver presented the audio for the next track where it actually starts after the latency based on FIR filter length and expected latency just like how it synchronizes audio/video playback, perhaps that's what's needed???

Yes, you may be right. You could test this by downloading the DMR Analyser from my sig, and running the test yourself. The regular renderer report is not sufficient to tell exactly what is going on, so you have to check the box "Enable HTTP logging" on the bottom of the screen, then run the tests again, and save & post the HTTP log file here..

EDIT: PS there is no way that MC could compensate for its own latency. Well, .. the only way that I could possibly imagine doing something that would be to start serving the track without DSP, and start processing the DSP in parallel at the same time, and somehow doing a soft cross fade between the non DSP version stream and the DSP version stream. But frankly I think that would sound totally dreadful..

[zappa]

So, sadly, the conclusion could be :

- there is no renderer able to play gapless using JRiver convolution filter on the server side ?

I have looked at your database and found there two of reported here renderers (AURALIC ARIES, JRMC20), according to your database both support gapless playback, but sadly it's not enough to do it with convolution activated.

[JonFo]

I can't wait to try DSP -> DLNA, but first I had an issue pushing content to my new renderer, a NAD D7050.

Thanks to AndrewFG (Whitebear) excellent Media Renderer Analyzer, I discover that even though this render is quite capable with direct-connected sources, its DLNA support has a bug with Gapless support.
But thanks to the handy dandy settings recommendations this app provides, I can configure MC to generate the correct outputs to drive it.

Incredibly useful app, and nicely implemented, donation sent!

[AndrewFG]

^

Many thanks. 

[JonFo]

Following on to the discussion about renders that claim to support "SetNextAVTransportURI" and then have issues, my new renderer, the NAD D 7050, indicates it supports that, but then totally screws up the playback of subsequent tracks (when using PCM).

The first track will play just fine, then the second track will sound garbled and 'choppy', then the third track just emits clicks as if the sample rate or bit depth were totally off. The unit then is stuck in some bad state and it requires a full power cycle to reboot.

The problem is that once the ControlPoint (MC) sends a SetNext, the NAD renderer must drop info about bit depth or sample rate and screws up thereafter. If it never receives a SetNext, then it works super reliably.

So if in doubt about whether the issue is DSP studio or some other variable, you can reduce your chance of issues by changing the DLNA Controller Options for your Rendering device in the 'Now Playing' view and select the ‘Disable SetNext support’ option.

And since this is a thread about DSP studio I can happily report it works just fine to provide EQ on the tracks being served up to the NAD via L16/44 PCM over DLNA now that I disable SetNext.

[AndrewFG]

Following on to the discussion about renders that claim to support "SetNextAVTransportURI" and then have issues, my new renderer, the NAD D 7050, indicates it supports that, but then totally screws up the playback of subsequent tracks (when using PCM).

The first track will play just fine, then the second track will sound garbled and 'choppy', then the third track just emits clicks as if the sample rate or bit depth were totally off. The unit then is stuck in some bad state and it requires a full power cycle to reboot.

The problem is that once the ControlPoint (MC) sends a SetNext, the NAD renderer must drop info about bit depth or sample rate and screws up thereafter. If it never receives a SetNext, then it works super reliably.

So if in doubt about whether the issue is DSP studio or some other variable, you can reduce your chance of issues by changing the DLNA Controller Options for your Rendering device in the 'Now Playing' view and select the ‘Disable SetNext support’ option.

And since this is a thread about DSP studio I can happily report it works just fine to provide EQ on the tracks being served up to the NAD via L16/44 PCM over DLNA now that I disable SetNext.

^

Indeed. I think this is a NAD specific issue. (I seem to have read about it before somewhere). I am guessing that the NAD is trying to download both the now playing track and the next track concurrently. And it presumably gives the now playing track download a higher priority, and perhaps the next track download is not getting enough resources (CPU cycles, memory buffers, IP network resources, etc.) for it to work properly so that it ends up dropping samples, to enable it to (pretend to) keep up. Or something like that (frankly I am speculating)..

Anyway, in addition to the suggestion to Disable SetNext in MC, I do suggest you should report this issue to the manufacturer; it is quite likely that they may want to fix this..

[jroyer]

HI, I am just starting to play with the network capabilities of JRiver and wonder how to leverage DSD and DLNA to leverage the following.  I want to setup a two computer setup.  Both running JRiver.  I would like my office PC to do all the computer and upsample all my album to DSD or DSD128 through the DSP tool, and then send the upsampled signal to the second computer (or stream to it) already in DSD format.  That second computer sits in my audio system and is connected to a USB DAC.

With two JRiver (one on each computer) I was able to leverage accessing the library of the "office" PC from my audio system PC.  However even if I choose the DSD option in the network menu to upsample to DSD on the main computer, the second computer only get the 44k signal.  Am I going at this the wrong way (with accessing the library of the second computer)?  I basically just want to stream upsampled DSD to my audio system music server which does not have the CPU power to do that work...  Any insight is appreciated.  Hope this is the right thread for this discussion!  Jacques

[tim.horel]

I'm now having the same issue, using a DLNA server I can't get the stream to process through the DSP.  It says it's not running.  It says that even with only the equalizer option checked.  The settings are as described to work, "Specified output format" with PCM 24b, which does play on the renderer (Marantz SR 6010). 

I'd really like to get volume leveling working...

Thanks.

[bob]

I'm now having the same issue, using a DLNA server I can't get the stream to process through the DSP.  It says it's not running.  It says that even with only the equalizer option checked.  The settings are as described to work, "Specified output format" with PCM 24b, which does play on the renderer (Marantz SR 6010). 

I'd really like to get volume leveling working...

Thanks.

You did the dsp studio settings IN the DLNA server configuration under Advanced?
If you have more than one DLNA server configured make sure the correct one is selected to output to the zone that represents your DLNA renderer (right-click on the zone and look at the DLNA option "Associate with DLNA Server"

[Taggert]

Almost everything is fine here with MC 22 as DLNA Server and active DSP. Apart from the Headphone DSP Setting. If I set "subtle", after leaving settings panel, this is always reset to "standard". Is there a way to avoid this?

[bob]

Almost everything is fine here with MC 22 as DLNA Server and active DSP. Apart from the Headphone DSP Setting. If I set "subtle", after leaving settings panel, this is always reset to "standard". Is there a way to avoid this?

Verified, checking into it.

[Taggert]

Thanks Bob, latest update 22.0.59 solved the problem!

[JJoe]

Resurrecting this topic from the dead   

Hello after reading both pages of this thread I'm not clear if all the previous DSP/DLNA/Convolution/Gapless...etc issues mentioned here have been resolved with the latest version of JRiver?

I'm intending to switch to JRiver from ROON so I can start using DSP convolution, Audiolense 5.x FIR filters but am hesitant to do so if all the previous issues have not been resolved yet since some of them are pretty fatal flaws if they still exist.

I will be using Win10 x64, ASIO USB Driver, Lynx Hilo and Audiolense 5.x initially which of course doesn't necessarily need DNLA but once I get my feet wet with that setup I am planning to then switch to an ETH connected Bricasti M5 DLNA Renderer. This is where the rubber will meet the road and where I will try to use the FIR Filters created with Audiolense with JRiver and serve them up via DLNA. The expectation is that everything will work, including gapless playback 

Can anyone confirm the status of the previous issues in this thread and if they feel the above proposed configuration may be problematic for what I am trying to achieve?

Thanks

[Scobie]

I have a best practice question on this topic.

if using a combination of DSP/DLNA and a hardware renderer that can perform some local equalisation, is there a preferred methodology of applying EQ?

Should I configure it via the JRiver DSP and flatten out the EQ on the receiver or is it just a matter of finding a balance? Bearing in mind the receiver talks to more than MC.

Thanks

Scobie

[bob]

I have a best practice question on this topic.

if using a combination of DSP/DLNA and a hardware renderer that can perform some local equalisation, is there a preferred methodology of applying EQ?

Should I configure it via the JRiver DSP and flatten out the EQ on the receiver or is it just a matter of finding a balance? Bearing in mind the receiver talks to more than MC.

Thanks

Scobie

I tend to do the EQ on the renderer, in my case mostly Id's
If I have a non-Id device that can't do EQ I create a second DLNA server for it and use DSP studio in that and associate that server with the non-Id renderer.

[B17NNS]

Audio
Specified output format
PCM 24 bits

Advanced
Sample rate same as source
DSP Studio

From here, on every option on the left column, I get "Playback stopped or current playback doesn't support processing".
And no effect on sound if I force equalization for example.

Having the exact same issue. Mac running JR23.

Every option says "Playback stopped or current playback doesn't support processing"

Did you find a solution? Like you am keen to address volume levelling.

[bob]

Having the exact same issue. Mac running JR23.

Every option says "Playback stopped or current playback doesn't support processing"

Did you find a solution? Like you am keen to address volume levelling.

Need more details.
What are you trying to play to?

[B17NNS]

Hi Bob.

Files are on a Synology NAS. Mixture of FLAC, MP3 and WAV.

MC is on an iMac, also using JRemote. Latest versions of everything.

Trying to play to a Denon AVR-X2400H. It's playing (I am having issues with fast scrubbing) but I can't activate any of the DSP settings.

Cheers, Matt

[bob]

Hi Bob.

Files are on a Synology NAS. Mixture of FLAC, MP3 and WAV.

MC is on an iMac, also using JRemote. Latest versions of everything.

Trying to play to a Denon AVR-X2400H. It's playing (I am having issues with fast scrubbing) but I can't activate any of the DSP settings.

Cheers, Matt

Since you've been configuring the DLNA server I assumed you went into the DSP studio under the Advanced part of the DLNA server audio settings to make the settings for your DLNA device?
That's where the setting for Volume leveling and output format are.
The DSP studio button on MC's main page is only for playback though local zones.

Also, I assume you are accessing the files on your NAS as files, not as a DLNA server under Playing Now->Playing from, correct?
The reason I ask is if you connect to the NAS as a DLNA server, when you push the files to the renderer from MC they will be coming from the NAS, not MC so the DSP studio effects can't be applied.

[B17NNS]

Since you've been configuring the DLNA server I assumed you went into the DSP studio under the Advanced part of the DLNA server audio settings to make the settings for your DLNA device?
That's where the setting for Volume leveling and output format are.
The DSP studio button on MC's main page is only for playback though local zones.

Yes, that's what I've been doing.

None of the options are available to me when configuring the DNLA server.

Every option says "Playback stopped or current playback doesn't support processing"

Also, I assume you are accessing the files on your NAS as files, not as a DLNA server under Playing Now->Playing from, correct?

No, I am accessing from "Main Library".

The reason I ask is if you connect to the NAS as a DLNA server, when you push the files to the renderer from MC they will be coming from the NAS, not MC so the DSP studio effects can't be applied.

I don't think I am accessing them from the NAS, see above.

[B17NNS]

https://imgur.com/cIree5t

https://imgur.com/OHoH2xW

[bob]

That configuration looks right.

You won't see the DSP studio settings while playing to the renderer, that is only for local zones as I said.

You can prove it's working by doing something extreme like enabling the equalizer and turning the bass all the way up and everything else all the way down.

You might want to check your files to see if MC has analyzed them for volume leveling. Looks at tags or show that field in the detailed files display.

[B17NNS]

No, nothing

https://imgur.com/wAO2bY2

[B17NNS]

Oh, hang on. When the next track started to play the pitch and tempo was off.

All audio has been analysed.

Might have cracked it thanks

[petrossa]

i would love to try out these features, but my ISP sends a different IP adress for my router via cellular than they send via the inhouse router  which Mediacenter obviously uses. So whilst inhouse everything works perfectly no way i can connect other than direct wifi. Is there anyway i can tell Mediacenter to use the IP address my ISP sends via cellular?

[bob]

i would love to try out these features, but my ISP sends a different IP adress for my router via cellular than they send via the inhouse router  which Mediacenter obviously uses. So whilst inhouse everything works perfectly no way i can connect other than direct wifi. Is there anyway i can tell Mediacenter to use the IP address my ISP sends via cellular?

What are you trying to connect to your MC with?
JRemote,
Gizmo,
?

[bonedoc]

I currently run JRiver on a Dell Deskop with Windows 10 and use it to play music at my desk while working.
I would like to connect it to my home audio system which includes a Marantz Receiver in the basement and a Denon in the den.  Both are connected to my home network via wireless. Both are DLNA capable.
I have used Sonos in the past but sold that stuff in a recent move and would like to avoid using it again.
What information there is with the receivers is spotty abpout networking.  I'm pretty well versed with computers in general but know very little about sound other than the file formats for music.
Is there a straightforward way to accomplish this so I can turn on my stero system and tell it to import music off my computer using JRiver?

Thanks.

[JimH]

Welcome.  The wiki has topics on Media Network and DLNA.  You could start there and then ask questions here if you get stuck.

[Davidhe19]

It's a go ! 
I have deleted my previous messages as they are irrelevant.
I tried the same Linux device, this time with a different software package called Volumio (which is a Linux player based on MPD), with UPNP enabled.
It will play all formats with DSP enabled, including (surprise !!) M4A files at 192/24 rates.
I'm impressed !
Yair

Would you or somebody else share the actual settings to get it work.  I tried many different settings and it has not worked for me.  Thank you.  David

[bob]

Would you or somebody else share the actual settings to get it work.  I tried many different settings and it has not worked for me.  Thank you.  David

Can you give us some more information about what device you are trying to play to?

[Davidhe19]

I somehow missed this message.  As a shared experience, I reported that I got DSP partially worked under DLNA server mode.  I used DLNA/network approach.  Under DLNA server window, under Audio, chose specified output format only when necessary.  Under advanced setting, checked all the format except the high res format such did and SACD.  Under DSP studio window, check the output format box.  In sample rate selection, I have chosen the rate 4 times of the original rate. I can see all the files have upsampled properly.  I can identified the rate by sending streaming signal to a Mutec USB 3+ reclocker, which also has a decoding capability of DSD up to 256.  I really wish to make DSD in DoP mode to convert all the nine high-res format.  Unfortunately it does not work.  I am using SOtM sMS 200 as a rendrrer.  (May from SOtM is always very helpful).  I have been using a switch/WiFi router to connect hard disk, MacBook with MC, and sMS 200, by various combination of USB, Ethernet or WiFi connection.  To my ear, the best sound comes from the complete WiFi approach, followed by the so-called “the bridged” approach.  All other approach always give a sense of glare at tremble, more or less.  “The bridged” still bears some glare but also provide the excitement.  But I like more realistic sound coming from WiFi.  I thought “the bridged” approach make make the DSD over DoP work because I felt MacBook is in the loop in this configuration.  Unfortunately it does not.