INTERACT FORUM
More => Old Versions => JRiver Media Center 20 for Windows => Topic started by: AoXoMoXoA on October 15, 2014, 11:10:26 am
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[Split from WDM Feature (http://yabb.jriver.com/interact/index.php?topic=92593.0) thread. Please see that thread for instructions.
Please use this thread for problems: http://yabb.jriver.com/interact/index.php?topic=93720.msg646166#msg646166 ]
excitement, but no joy!
It does not work . . . set MC driver as default sound device, ran through Windows sound test and it worked.
Next tried to play music in Foobar through MC's driver and got only silence although Foobar shows it is playing.
Went back into Windows sound settings and tried to test the MC sound device and got nothing.
Tried playing music in MC but still nothing, even after reboot.
After re-setting my audio to the regular device sound was returned, but when I tried to set a zone in MC to use the MC driver I still had no audio even thought MC appeared to be playing . . . . then it crashed Windows forcing a restart.
Windows 8.1 / MC 20.0.25
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Just a note that this is a holy grail of audio. I so wanted a way to route audio through MC, and now it's here. It's a really elegant solution. Thanks to everyone that helped.
For so long, I said "Hey, driver programmer wanted" because I was afraid of drivers. I hoped, hoped, hoped, but finally bit the bullet and learned myself.
Matt, you are the king man! Developing a WDM audio driver on the Windows platform is about the hardest thing to program - source: I work at Microsoft as a programmer.
Kudos!
Best regards, Mitch
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Impressive work.
While I have no trouble with crashing, I do have audio/video sync problems. The audio is delayed by roughly 300 ms (personal estimate).
Am I the only one experiencing this?
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Impressive work.
While I have no trouble with crashing, I do have audio/video sync problems. The audio is delayed by roughly 300 ms (personal estimate).
Am I the only one experiencing this?
Set the buffering in Options > Audio > Device settings lower. Pick a really tight value.
Also, Options > Audio > Advanced > Live playback latency. Turn that down.
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Tried playing music in MC but still nothing, even after reboot.
After re-setting my audio to the regular device sound was returned, but when I tried to set a zone in MC to use the MC driver I still had no audio even thought MC appeared to be playing . . . . then it crashed Windows forcing a restart.
It sounds like you're trying to select the JRiver driver as an output in JRiver? If that's what you're doing, that won't work (and might cause an infinite loop). JRiver needs to be set to output to your normal soundcard, with the JRiver driver set as system default.
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That works!
Didn't before somehow. But I retried it with success.
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Fantastic! And it's not even christmas yet.. :)
It just works! I love it.. ;D
Thanks Matt!
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It sounds like you're trying to select the JRiver driver as an output in JRiver? If that's what you're doing, that won't work (and might cause an infinite loop). JRiver needs to be set to output to your normal soundcard, with the JRiver driver set as system default.
OK, tried it again after reading your post and that was my error . . . MC was set to output to the Default Device, which once set to JRiver Driver caused said loop.
Works now. Thanks
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I can't wait to try this out later! Excellent news!!! :)
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Good to see this out so early in the r20 cycle. Couple of questions;
Do you know what latency this adds in itself or is it specific to the PC it runs on?
Is it possible to zone switch on activation of the wdm input?
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Do you know what latency this adds in itself or is it specific to the PC it runs on?
The added latency is configurable based on the "live playback latency" setting; the actual total tip-to-tip software latency is pretty close to the sum of the live playback latency setting and the audio output device buffer setting (based on my measurements). It may be adding a handful of additional ms, but definitely in the single digits (I was seeing mostly onesies or twosies) on top of the configured latency.
The "minimums" will depend on what buffering settings your devices/computer will tolerate. On all the PCs I tested, I could achieve a configuration that was good enough for lipsync without convolution (although on one machine it was verging on unacceptable). By contrast with convolution some of my machines were still solid, but some were definitely not.
Is it possible to zone switch on activation of the wdm input?
Yes, the rule that will choose the WDM input is [Name]="IPC"
I personally recommend setting up a separate zone to receive WDM playback as that will make certain activities much easier/more seamless (e.g. running netflix in theater view).
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Finally! Thank you very much!
This and some improvements to the DSP Section and the audiopart is perfect (for me). If this was the only improvement in v20 I'd still pay the full upgrade price.
Also thanks to mwillems for the zone-switch tip.
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Gave it a quick try at lunch time using Spotify. Works great!
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This thread makes me happy for so many reasons :)
Thanks for all of your hard work.
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Here is a simple YouTube Clip (https://www.youtube.com/watch?v=r2GcmELJZKw) to test your AV Sync and it seems spot on for me with low latency settings for my DAC.
- Tools-> Options-> Audio-> Audio Device -> Device Settings -> Buffering = "Minimum hardware size"
- Tools-> Options-> Audio-> Advanced-> Live playback latency = "Minimum"
Note: You may have issues with "Minimum hardware size" pending your HW and connection methods. eg I found with my o2 DAC it works best on the USB3 port with these low latency settings.
Thanks
Nathan
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Thanks. Works perfectly with both airserver and shairport4w to AirPlay to mc.
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thanks @mwillems for the speedy reply
another question, what do these sliders do in the properties dialog for the device? max value says it is a signed 32 bit int value but what it's for is the question :)
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HAPPY! HAPPY! JOY! JOY!!!
Thank you!! Thank you!! Thank you!!
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Matt, nice feature! I have YouTube playing through "Player" with a locally connected USB device. Can you look into supporting this feature with DLNA renderers? When I try a DLNA renderer I get an error regardless of the specified output format (PCM, MP3) selected.
Jesus R
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@MC team, nice work! What "interfaces" does the driver support (MME, DirectSound, WASAPI, ASIO, KS)? Also, is the installation of the WDM driver optional or standard/mandatory?
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@MC team, nice work! What "interfaces" does the driver support (MME, DirectSound, WASAPI, ASIO, KS)? Also, is the installation of the WDM driver optional or standard/mandatory?
The driver install is standard.
For what interfaces are supported, it's any interface that talks to a WDM driver. That's WASAPI, KS, DirectSound, etc.
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Here is a simple YouTube Clip (https://www.youtube.com/watch?v=r2GcmELJZKw) to test your AV Sync and it seems spot on for me with low latency settings for my DAC.
I get a bit of drift on that YouTube version as the video progresses, and I don't think it is my hardware. The YouTube version is just a bit of a poor copy.
Try downloading the "BBC-Audio-Sync-Test" from http://clip.dj/bbc-hd-audio-sync-test-download-mp3-mp4-bCPEidaVzQU
That was the best version on the page I found.
I also found these clips useful:
Audio-Video-Sync-Test-Calibration-23.976fps
Audio-Video-Sync-Test-Calibration-29.97fps
BBC-Test-cards-on-BBC-HD-Shutdown-26th-March-2013
Especially the last clip. Just don't get caught up in the popup surveys and rubbish on that page.
Thanks for putting me on the path to find those jmone.
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This is amazing! Thank you very much.
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Yeah it was the BBC test clip I was looking for but in my haste got the other one. ;D
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Works great on my Windows 7 Professional desktop computer! This is a great feature.
What should the sample rate in the WDM drivers be set to? The sample rate of the source material? For example, for Spotify, set to 24-bit, 44100 Hz? For YouTube videos, set to 48000 Hz?
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Thanks so much JRiver, I can't wait to try this!
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Is this driver available for previous versions of jriver?
Or do i have to upgrade to jriver 20 to get this driver?
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Its a new feature in MC20, its not available for previous versions, sorry.
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Works fine for me, except when I select my Squeezebox Touch as a zone.
When I was fiddling with this, MC hung, was unkillable, I rebooted which happened to pick up a windows update, during which I got a "DRIVER_POWER_STATE_FAILURE" message which may or may not be relevant. Eventually re-reboot worked, and all is fine.
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Well, I'm not sure to understand everything (I'm french and my english is not so good). I've selected JRiver MC 20 as default sound device but it works only if the J River MC 20 software remains open, is it the standard behavior ?
Thank you for you rhelp.
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And what about using a DAC ? Maybe this is a stupid question, but I have a USB DAC plugged and I used to set it as default in windows sound options. Is it better to use JRiver by default, the DAC ?
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Well, I'm not sure to understand everything (I'm french and my english is not so good). I've selected JRiver MC 20 as default sound device but it works only if the J River MC 20 software remains open, is it the standard behavior ?
Thank you for you rhelp.
Yes. When you set the jrmc wdm driver as the default audio device all audio played on the pc will be routed to the jrmc app. I has to be open to receive the audio.
Then in jrmc you can send the received audio to your dacs sound device.
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I can't seem to get this to work - followed instructions as per the thread but I just get no sound... think it has to do with the Tools>Options>audio device - in JRiver....
could somebody please give a detailed instruction on what that setting should be...
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Just installed the latest version, using it with GeekOut ASIO.
Installation went fine, however during testing it seems I can only hear left channel. I tried
- changing speaker configuration to 5.1 and reverted back to stereo
- uninstalling the driver and deleting its files. Had to re-install MC20 to get back the WDM
OS is Windows 7 x64 Pro, using WindowBlinds 8 and Fences.
Any ideas?
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I can't seem to get this to work - followed instructions as per the thread but I just get no sound... think it has to do with the Tools>Options>audio device - in JRiver....
could somebody please give a detailed instruction on what that setting should be...
It should be set to whatever your speakers are actually attached to. For example mine is set to Sound Blaster Zx [WASAPI]
Device settings set the buffering to Minimum Hardware Size if that gives you any problems (clicks, stuttering, etc) raise it incrementally until it plays properly.
Also set Advanced/Live playback latency to minimum.
Right click the volume icon in MC select system volume and make sure it is not all the way down. The normal "System volume" (windows volume) no longer has any influence because you are bypassing the windows mixer. I was a bit puzzled at first because my volume keys on my keyboard and mouse were not lowering the volume and then the light in my head went off and I realized I had to use the application volume (youtube) and/or MC.
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Just installed the latest version, using it with GeekOut ASIO.
Installation went fine, however during testing it seems I can only hear left channel. I tried
- changing speaker configuration to 5.1 and reverted back to stereo
- uninstalling the driver and deleting its files. Had to re-install MC20 to get back the WDM
OS is Windows 7 x64 Pro, using WindowBlinds 8 and Fences.
Any ideas?
What are you using for hardware? (DAC, onboard audio, sound blaster, etc)
I am not familiar with GeekOut ASIO have you tried any other options.
I am using Win 7 x64 Home Premium and Creative Sound Blaster Zx [WASAPI] with out any issues
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What are you using for hardware? (DAC, onboard audio, sound blaster, etc)
I am not familiar with GeekOut ASIO have you tried any other options.
I am using Win 7 x64 Home Premium and Creative Sound Blaster Zx [WASAPI] with out any issues
Hi there,
I am using an external USB DAC (this http://lhlabs.com/marketplace/geek-out/overview.html). The DAC provider offers both ASIO and WDM drivers and I am using the former in JRiver for audio playback.
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The Windows / Control Panel / Sound / Driver / Configure / Test function does not really work if you select 5.1 output; from some channels you hear the test beep and for some not; I suppose it is something to do with latency..
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Well, I'm not sure to understand everything (I'm french and my english is not so good). I've selected JRiver MC 20 as default sound device but it works only if the J River MC 20 software remains open, is it the standard behavior ?
Thank you for you rhelp.
Yes. MC needs to be open to capture anything.
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MC needs to be open to capture anything.
It would be a nice feature if the driver could start MC.exe if a sound comes in and if MC is not already running.
However I don't know if a driver (in the OS kernel layer) is actually permitted to start an application (in the OS user layer) (??)
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That would be extremely problematic.
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Yes. When you set the jrmc wdm driver as the default audio device all audio played on the pc will be routed to the jrmc app. I has to be open to receive the audio.
Then in jrmc you can send the received audio to your dacs sound device.
Thank you !
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It would be a nice feature if the driver could start MC.exe if a sound comes in and if MC is not already running.
However I don't know if a driver (in the OS kernel layer) is actually permitted to start an application (in the OS user layer) (??)
I can't confirm this right now because I'm not in front of a box, but I think running the MC server widget in the tray will do this; my recollection is that it will spawn a JRiver window on incoming system sound if no JRiver window is open (I can't test right now, but that's my recollection of how it worked).
The Windows / Control Panel / Sound / Driver / Configure / Test function does not really work if you select 5.1 output; from some channels you hear the test beep and for some not; I suppose it is something to do with latency..
Very short sounds don't always come through, or come through truncated; a better test is to play known 5.1 content and selectively mute channels in JRiver.
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Holy grail, indeed! I honestly didn't expect you'd pull through when I renewed my license. This opens up so many possibilities. Forget "please integrate music service X in jriver", now you get nearly all the benefits of service integration without it being actually integrated!
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Fantastic!!!! Thanks a bunch!
Used it with Wimp streaming service today, works almost perfect. It seems to stutter just a bit every now and then. May not be the WDM driver's fault for all I know, though.
In Wimp, and contrary to Spotify, the user can choose audio output device. So there is no need to use default device (which would mess up my IP telephony setup at work). Combined with new 24bit rips and lossless flac streaming, this is a killer combo.
Looking forward to test it with video streaming services.
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Thank you very much. This is a really great new feature.
It was little interesting to get setup. Because you can't choose direct sound in JRiver. As it's now referring to itself. ;D
This may be useful to show in JRiver that you are using JRiver in the os setting.
I had to lower the latency to play video games but it seemed to work well after that.
The only real problem i had was linked zones did not work well.
I was trying to play to the sound card and my wireless headphones at the same time.
Has anyone else try this?
If JRiver is shut down can it tell windows it it disconnected?
So it can fall back to the other audio sources.
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Sorry to ask this. I would like to be sure to understand the benefit of this nex feature.
How this wdm driver solution is better than for example than the quobuz player or any player?
In my system, and for many of us, the usb dac makes any sound going out better (to say short).
If someone can explain me why it is so nice ;)
Thanks
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Sorry to ask this. I would like to be sure to understand the benefit of this nex feature.
How this wdm driver solution is better than for example than the quobuz player or any player?
In my system, and for many of us, the usb dac makes any sound going out better (to say short).
If someone can explain me why it is so nice ;)
Thanks
For example I prefer to use Spotify with the iPad app. Now I have a very stable solution to AirPlay from the app to Jrmc and take advantage of its dsp. In my case to apply a convolver for room correction. Also because the ability to setup zones and automatically switch between them, I can switch between this and playing my own jrmc library with jremote with ease. So in short anything that I play can take advantage of jrmc dsp/audio path.
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How this wdm driver solution is better than for example than the quobuz player or any player?
In my system, and for many of us, the usb dac makes any sound going out better (to say short).
If someone can explain me why it is so nice ;)
Media Center has the best audiophile playback chain available. It offers perfect modes like ASIO and WASAPI, it offers full control over sample rates and sets hardware sample rates, it offers a comprehensive set of DSP, etc. You can leverage that same chain from other players by using the WDM driver.
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Amazing feature! Does it work for remote zones too? I mean is it possible now to route sound from notebook's web browser window to htpc hdmi out?
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I can't seem to get this to work - followed instructions as per the thread but I just get no sound... think it has to do with the Tools>Options>audio device - in JRiver....
could somebody please give a detailed instruction on what that setting should be...
Did you manage to get it to work because I can't either??
In JR/Audio device set to my SC. In Windows sound devices set JRiver media center.
I'm running W7 64bit
I can see sound playing in the spectrum analayzer, but I'm getting no sound.. ?
When I 'Test' the JRiver media center in Windows sound devices 'Ipc opening' is displayed at the top of the screen.
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The only real problem i had was linked zones did not work well.
I was trying to play to the sound card and my wireless headphones at the same time.
Has anyone else try this?
I doubt it's going to work with linked zones. I asked a similar question in regard to the ASIO input a while back. Basically, that doesn't work and would require a major redesign. I'd guess the same architecture issues apply here.
-tm
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I've got Spotify working with it now. Great!
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Great news, thank you team !
Just in time as I was about to start digging info on the older audio loopback feature. I guess my job is now much simpler :-)
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Just installed the latest version, using it with [USB external DAC] ASIO.
Installation went fine, however during testing it seems I can only hear left channel.
Confirmed (using a different DAC than his), so it's a bug.
Do you want WDM bugs listed here or in the usual build bug thread ?
Details: If you click "Test" it lights up the left speaker icon and you hear the test tone, it then lights up the right speaker icon and you hear nothing. If you click on the right speaker icon, you hear the test tone, oddly enough.
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Split WDM Fine Points (http://yabb.jriver.com/interact/index.php?topic=92620.0)
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Nice job. While this has actually made me realize that I'm already playing almost everything inside Media Center now, it seems to be working well and there are definitely some good uses for it.
It's certainly going to make things easier for people who were using Loopback a lot, or people with power amplifiers who need volume protection enabled for everything running on the PC.
Yes, the rule that will choose the WDM input is [Name]="IPC"I personally recommend setting up a separate zone to receive WDM playback as that will make certain activities much easier/more seamless (e.g. running netflix in theater view).
As a tip for anyone else trying to get this working, I recommend that you move this rule to the very top of your Zone Switch list.
It did not seem to be working for me until I made that change.
Rather than switching based on the name containing "IPC" I prefer to use: [Filename]=[live:////ipc"
While it is unlikely, this will guarantee that there won't be any false positives.
For example I prefer to use Spotify with the iPad app. Now I have a very stable solution to AirPlay from the app to Jrmc and take advantage of its dsp. In my case to apply a convolver for room correction. Also because the ability to setup zones and automatically switch between them, I can switch between this and playing my own jrmc library with JRemote with ease. So in short anything that I play can take advantage of jrmc dsp/audio path.
AirPlay → Airfoil Speakers → JRiver WDM seems to work rather well.
As a free alternative, there is Shairport4w (http://sourceforge.net/projects/shairport4w/), though this didn't seem to play nicely with the hardware AirPort devices on my network and was preventing them from being listed on iOS devices. (Airfoil Speakers (http://www.rogueamoeba.com/airfoil/windows/) has no such issues)
This means that any app on iOS can now send audio into Media Center for processing and output (e.g. convolution/room correction, EQ, declipping, or anything else you want to do) so guests can now play music that is stored on their device or via apps like Spotify directly to Media Center with zero configuration.
The only real problem i had was linked zones did not work well.
I was trying to play to the sound card and my wireless headphones at the same time.
You're correct that it doesn't seem to work in linked zones.
However, once you have playback going into Media Center via the WDM driver to a single zone, you can route that to other zones which are linked together by using Loopback.
What you may want to do is route audio from the WDM driver into a new zone (using the Zone Switch rule listed above) which uses the Null Output device.
Then link your other zones together and enable Loopback.
Tip: If you start Loopback (File → Open Live → WASAPI Loopback) you can press and hold on one of the "car radio" buttons (1-12 in Playing Now) to create a button which immediately starts Loopback.
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As a tip for anyone else trying to get this working, I recommend that you move this rule to the very top of your Zone Switch list.
It did not seem to be working for me until I made that change.
You're right, I can confirm it does need to be at the top; it's because the IPC/WDM feed registers as audio content and so will get picked up by rules that play audio. Another (failsafe) way to do it that doesn't rely on placement of the rule is to affirmatively exclude IPC from other zones (which I did as a precaution).
[Filename]=[live:////ipc"
I haven't tried using the filename, but shouldn't the unclosed bracket above be a quotation mark (bolded for emphasis)?
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You're right, I can confirm it does need to be at the top, unless you affirmatively exclude it from other zones that play audio (which I also did as a precaution). It's because the IPC registers as audio content and so will get picked up by rules that play audio.
I haven't tried using the filename, but shouldn't the unclosed bracket above be a quotation mark (bolded for emphasis)?
No, the rule is "starts with" live://ipc
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No, the rule is "starts with" file://ipc
Got it, the closing quote was what was throwing me; had to brush up on my "search language"
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Used it with Wimp streaming service today, works almost perfect. It seems to stutter just a bit every now and then. May not be the WDM driver's fault for all I know, though.
I'm using WDM driver with Deezer and am noticing frequent crackling sounds, as often as once every 10-20 seconds. From what I can tell it's happening at random (replaying portions of a song that made crackling sounds may not result in crackling sounds at the same exact point). Maybe some buffers are set too low? I'm using it with a DAC.
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I'm using WDM driver with Deezer and am noticing frequent crackling sounds, as often as once every 10-20 seconds. From what I can tell it's happening at random (replaying portions of a song that made crackling sounds may not result in crackling sounds at the same exact point). Maybe some buffers are set too low? I'm using it with a DAC.
Increase buffering with Options > Audio > Advanced > Live playback latency. See if that squares you up.
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I've set both buffers to 250 ms and the crackling sounds remain frequent.
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I've set both buffers to 250 ms and the crackling sounds remain frequent.
Well just try a bigger value as a test.
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Well just try a bigger value as a test.
500 ms, still crackles.
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500 ms, still crackles.
What happens if you increase the buffering in Options > Audio > Device settings?
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What happens if you increase the buffering in Options > Audio > Device settings?
It's already at 500 ms. No difference :'(
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It does seem to be quite sensitive to interruption at times.
I assumed it was because I have a handful of VST plug-ins running for declipping, dithering, and headphone listening - I don't normally use the declipping, but it helps with streaming services.
I've not run into those problems when playing content inside Media Center directly, rather than an external app via the WDM driver though.
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I can't seem to get my WDM zone to automatically become the active zone. It will switch to it just fine using either the [Name] or [Filename] rule regardless of the order of the ZoneSwitch rule. However, the last zone active stays as the visible zone. This means that I have to manually click on the WDM zone in order for its volume control to work.
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I can't seem to get my WDM zone to automatically become the active zone. It will switch to it just fine using either the [Name] or [Filename] rule regardless of the order of the ZoneSwitch rule. However, the last zone active stays as the visible zone. This means that I have to manually click on the WDM zone in order for its volume control to work.
I get the same behavior, but from my perspective it's a feature; otherwise the integrated netflix would never work at all. With the two zone system, when netflix starts in a regular playback zone, the audio gets re-routed to the WDM zone. If the WDM zone stole focus the playing video would vanish, and I'd need to manually switch back to the main playback zone (and netflix may well have stopped playing by then because it tends to stop when it's not in focus).
I assumed that the WDM didn't steal focus on purpose for exactly that reason, and I much prefer the current functionality to the alternative which would break any kind of use of the internal browser for streaming video. It makes volume control a bit of a pain, but there are workarounds (gizmo, etc.).
It's already at 500 ms. No difference :'(
What happens if you try a very low number? I couldn't get good playback with one system until I lowered the buffer to 20ms.
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I can't seem to get my WDM zone to automatically become the active zone. It will switch to it just fine using either the [Name] or [Filename] rule regardless of the order of the ZoneSwitch rule. However, the last zone active stays as the visible zone. This means that I have to manually click on the WDM zone in order for its volume control to work.
That's how Zone Switch has always worked.
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What happens if you try a very low number? I couldn't get good playback with one system until I lowered the buffer to 20ms.
I think you may be on to something. I changed all the buffers back to the recommended values and then sorta forgot about it until I suddenly noticed: a significant decrease in crackling. I don't dare change the settings any further, seeing as it's working pretty much okay now.
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Right click the volume icon in MC select system volume and make sure it is not all the way down. The normal "System volume" (windows volume) no longer has any influence because you are bypassing the windows mixer. I was a bit puzzled at first because my volume keys on my keyboard and mouse were not lowering the volume and then the light in my head went off and I realized I had to use the application volume (youtube) and/or MC.
Any chance this might change? I use a bluetooth remote (it's far more responsive than bothering with an app like Gizmo) and it changes the system volume. This seems nice, but there's no way I'd use it as it stands because my DAC doesn't have a remote and the preamp I route my audio through has a remote that eats AAA batteries like no tomorrow.
Thanks,
--Ryan
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I can't seem to get my WDM zone to automatically become the active zone. It will switch to it just fine using either the [Name] or [Filename] rule regardless of the order of the ZoneSwitch rule. However, the last zone active stays as the visible zone. This means that I have to manually click on the WDM zone in order for its volume control to work.
What would help several different problems, would be:
* Another Selector in Zoneswitch - just exactly like the "stop zone" selector - but instead a "Switch Active Zone to this Zone When the Zone is Started".
Thanks !
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Matt.. Well done. What a milestone!!
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It should be set to whatever your speakers are actually attached to. For example mine is set to Sound Blaster Zx [WASAPI]
Device settings set the buffering to Minimum Hardware Size if that gives you any problems (clicks, stuttering, etc) raise it incrementally until it plays properly.
Also set Advanced/Live playback latency to minimum.
Right click the volume icon in MC select system volume and make sure it is not all the way down. The normal "System volume" (windows volume) no longer has any influence because you are bypassing the windows mixer. I was a bit puzzled at first because my volume keys on my keyboard and mouse were not lowering the volume and then the light in my head went off and I realized I had to use the application volume (youtube) and/or MC.
THank you for your reply - but still can't get anything out of any option....I'm using JBL Pebbles and that is a USB connection to the back of the PC... the soundcard outputs seem to be my optical and digital outs. still no sound with any of the options presented(http://)
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The Windows / Control Panel / Sound / Driver / Configure / Test function does not really work if you select 5.1 output; from some channels you hear the test beep and for some not; I suppose it is something to do with latency..
Very short sounds don't always come through, or come through truncated; a better test is to play known 5.1 content and selectively mute channels in JRiver.
I tested this thoroughly with my setup, as below in my signature, and using the motherboard audio S/PDIF optical out, sending DD AC-3.
The speaker test function only works for the Left, Center, and Side Left when using 5.1 channels, when I press the "test" button. Windows always delays the output in these tests, so it is necessary to run them at least twice to get sound output on all speakers, but running them multiple times still only plays those three speakers.
However, when I click on the individual speaker images, test sounds are played from all speakers. The response is quite slow though, and if I click the speaker more than once, even with a reasonable gap between clicks, I only get one sound output. This is the Windows delay being introduced perhaps, or the receiver taking some time to respond. I can clearly see MC play each sound in the Action Window, and until that window clears, and for a little while after, a new click won't produce a sound. I observed that this sometimes meant I missed Windows Event sounds, if two or more events followed in quick succession.
This is true on my main PC/client system which only has stereo speakers as well. Only the Left speaker plays on clicking the "test" button, but both play on clicking the speaker images. I do get clipping or broken sound output for the test sounds on this PC as well, but not on the HTPC.
However 5.1 media plays normally on the HTPC system, and stereo plays normally on my main PC/client. While the "test button" feature does appear to be broken, this certainly isn't a critical issue.
One other feature doesn't quite match my system requirements. I have Bose satellites for my front Left and Right speakers, and Bose monitor speakers for my Side Left and Side Right. The satellite work best when they are not set as full range, or at least I think they do, and I'd like to try them that way. The Side monitor speaker are full range, so should be set that way. However I can't set them this way. If I select the fronts as full range, I can set the sides as full range. If I deselect the fronts as full range, the sides are automatically deselected and can't be manually selected as full range. Again, not a big issue, but a little strange. I can leave my fronts set at full range for now though.
Finally, on the HTPC I was able to reduce the Buffering setting to "Minimum Hardware Size", but I couldn't reduce the Live Playback Latency to anything less than the default "50 Milliseconds (recommended)" setting, or I would get clipping or similar effects.
Oh, also Windows also reported that the 20.0.25.0 driver was not digitally signed.
None of this outweighs the brilliance of having a WDM sound driver for any and all Windows applications to use on the HTPC.
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I am testing using this on a JRiver in client mode. It's a bit add to always have JRiver open/minimized.
Is there anything we can do to deal with that?
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I am testing using this on a JRiver in client mode. It's a bit add to always have JRiver open/minimized.
Is there anything we can do to deal with that?
If you run mc as a server then when you close the app it will still receive the wdm routed sound and play it without a window open or minimized.
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Yes but at work I run it as a client.
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Yes but at work I run it as a client.
just disable the library and Dlna servers in options > media network. It will then act like a windowless client in the background when used with the wdm driver to play sound trough it.
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Used it with Wimp streaming service today, works almost perfect. It seems to stutter just a bit every now and then. May not be the WDM driver's fault for all I know, though.
Options > Audio > Advanced > Live playback latency = 50ms: Some skipes and stutters
Options > Audio > Advanced > Live playback latency = 500ms: static noise comes and goes in right channel
Options > Audio > Advanced > Live playback latency = 250ms: All good (streaming from Wimp)!
With Live playback latency = 50ms, ASIO buffer sizes of 50 or 500ms did not make any difference.
EDIT: Only DSP in MC is "Headphones".
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Yes but at work I run it as a client.
He's talking about running the server widget, not the media server options (they're separate). The media server widget is just a stripped down MC instance that lives in the tray; it doesn't necessarily have to serve anything.
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I can't seem to get my WDM zone to automatically become the active zone. It will switch to it just fine using either the [Name] or [Filename] rule regardless of the order of the ZoneSwitch rule. However, the last zone active stays as the visible zone. This means that I have to manually click on the WDM zone in order for its volume control to work.
What would help several different problems, would be:
* Another Selector in Zoneswitch - just exactly like the "stop zone" selector - but instead a "Switch Active Zone to this Zone When the Zone is Started".
Thanks !
Requested in MC20 Feature Requests here! http://yabb.jriver.com/interact/index.php?topic=90662.msg634413;topicseen#msg634413 (http://yabb.jriver.com/interact/index.php?topic=90662.msg634413;topicseen#msg634413)
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What would help several different problems, would be:
* Another Selector in Zoneswitch - just exactly like the "stop zone" selector - but instead a "Switch Active Zone to this Zone When the Zone is Started".
Thanks !
As long as it's an option and/or configurable I'm on board (+1). I just don't want to lose the ability to not switch to the active zone for reasons described above.
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Does MC20 and the WDM driver support relaying a web audio stream to another device?
I have a Windows PC running MC 20.0.25 with the Windows sound set to JRMC20 using the WDM driver. I can open a web browser and play music from the web through MC20 – it works just fine. Let’s call this machine PC1.
I also have a dedicated music server running Windows and MC 20.0.25. Let’s call this machine PC2. I can open a browser on PC2 and stream music from the web through MC20 using the WDM driver.
On PC1 in MC20, I can select PC2 as a zone and play music from my library on PC2. That works fine.
What doesn’t work for me is this scenario:
PC1: Windows sound is set to MC20 using WDM driver. MC20 current zone is set to PC2.
PC2: Current zone is set to Player.
On PC1, I stream music from the web.
On PC2, I see the Ipc file in the Player Playing Now content pane.
However, no audio is output from PC2.
Have I missed something in my setup? Or is this not supported?
Thanks,
-Brent
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Forwarding the audio over the network is not supported. Although that might be an interesting idea...
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I noticed a quick update that got rid of some redundancy, but in the device options is "exclusive access" also redundant (or should i still use it) or have i missed something as it seems to me setting this up this way makes MC 100% exclusive but muchly improved as i don't have problems elsewhere. TIA.
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That's how Zone Switch has always worked.
Zone Switch always switches to the active zone for me except for the WDM zone. I tried to figure out what kstuart's issue was in this thread (http://yabb.jriver.com/interact/index.php?topic=90242.msg624285#msg624285) with no success.
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Zone Switch always switches to the active zone for me except for the WDM zone. I tried to figure out what kstuart's issue was in this thread (http://yabb.jriver.com/interact/index.php?topic=90242.msg624285#msg624285) with no success.
As noted above (http://yabb.jriver.com/interact/index.php?topic=92593.msg637763#msg637763) I think it's by design, otherwise it would be very hard to effectively use the WDM for audio from video streaming through the integrated browser/netflix.
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)p( thanks
I didn't think it would be a good idea to run server and client pointing to another server on the same system.
But you're right it works.
I wonder if we can add zones to the tray icons for media center and media server.
So we can switch input without opening media center.
It also takes a long time to switch from media center client to the server playing the audio.
Not long be enough time to wonder if it's going to work.
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Great feature!
Another bonus -
Just tested and it enables measurements using REW through JRiver and Dirac Live!
Finally a simple way to validate Dirac's results and correct XO settings.
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Hi Matt - I just wanted to say thank you for all your work on this. This feature has been worth the wait (and the price of the upgrade this year). It works perfectly on my system so far (using both Netflix and Spotify).
cheers,
Stuart
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Did you manage to get it to work because I can't either??
In JR/Audio device set to my SC. In Windows sound devices set JRiver media center.
I'm running W7 64bit
I can see sound playing in the spectrum analayzer, but I'm getting no sound.. ?
When I 'Test' the JRiver media center in Windows sound devices 'Ipc opening' is displayed at the top of the screen.
Got it working. I had to enable just about all my sound devices in Windows playback devices before I could set jriver media center to the default device..
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When I enable WDM JRiver in windows control panel I obtain a distorted sound, the same that I obtain when I enable PCM to DSD conversion as I explain this thread http://yabb.jriver.com/interact/index.php?topic=91987.msg633009#msg633009; I have already played with buffer size but nothing changed.
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This new feature brings on a whole new world of applications. I was hoping, just hoping, that it would work flawlessly with multichannel audio convolution (I use a 15 path 64k taps Audiolense filter). The old loopback was limited to fewer (two) channels, multichannel long filters caused stutter.
IT WORKS!!!!
Thanks again to the JRiver team, maybe Matt in particular for learning how to code drivers.
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I'm finding that the default ~5 seconds of silence before stopping playback is too short. Increasing this to at least a minute would be preferable.
An option to simply enter how many seconds we want the connection to remain open would be good.
Since I'm routing the sound to separate devices, and Loopback seems to work well, I might even prefer that Media Center always keep the connection open.
I'm also finding that no matter how long I set the "play silence before for hardware synchronization" option, I'm still getting glitches for the first couple of seconds of playback.
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I'm finding that the default ~5 seconds of silence before stopping playback is too short. Increasing this to at least a minute would be preferable.
An option to simply enter how many seconds we want the connection to remain open would be good.
Since I'm routing the sound to separate devices, and Loopback seems to work well, I might even prefer that Media Center always keep the connection open.
I actually like the 5 second behavior quite a bit, and would prefer it, if anything, to be slightly shorter; but I definitely would support a configurable solution.
My audio device will sometimes error if I try to playback something else while the WDM input is open in another zone (even with appropriate zoneswitch stop playback rules). As is, it's easy for me to just wait a couple seconds for it to "latch off" rather than having to manually stop it every time. If the default were a minute, I would constantly be forgetting that it's still on, and throwing errors which disrupt theater view, etc.
I'm also finding that no matter how long I set the "play silence before for hardware synchronization" option, I'm still getting glitches for the first couple of seconds of playback.
I can confirm that I'm seeing the same thing, the first few seconds are always kind of odd. I think the driver ignores that setting entirely.
I think it does that to maintain AV sync: when I start a Netflix movie using WDM, the video starts, but it takes a few beats for the driver to start playback in JRiver, so the audio starts up a little late and lags for a second or so, but then it suddenly "catches up" to the video and after that the sync is fine.
So in order to avoid that glitch, the driver would either need to start faster (which it probably will in time) or the "catch up" would need to be more gradual (which would mean poor sync for longer, which wouldn't be ideal).
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I actually like the 5 second behavior quite a bit, and would prefer it, if anything, to be slightly shorter; but I definitely would support a configurable solution.
What I'm finding is that if I'm watching a YouTube video and then it has to buffer for a few seconds before the next video in the playlist starts, or listening to a song which has silence at the end, the connection is dropped and restarts, which at best means a couple of seconds of audio glitching, and at worst I get the ASIO "format not supported error" and have to completely restart MC.
I think it does that to maintain AV sync: when I start a Netflix movie using WDM, the video starts, but it takes a few beats for the driver to start playback in JRiver, so the audio starts up a little late and lags for a second or so, but then it suddenly "catches up" to the video and after that the sync is fine.
So in order to avoid that glitch, the driver would either need to start faster (which it probably will in time) or the "catch up" would need to be more gradual (which would mean poor sync for longer, which wouldn't be ideal).
You could be right. I think it might be better to simply mute the first couple of seconds to hide that glitch though.
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What I'm finding is that if I'm watching a YouTube video and then it has to buffer for a few seconds before the next video in the playlist starts, or listening to a song which has silence at the end, the connection is dropped and restarts, which at best means a couple of seconds of audio glitching, and at worst I get the ASIO "format not supported error" and have to completely restart MC.
I can see why you would find that irritating; a configurable "latch off" may be the best answer.
You could be right. I think it might be better to simply mute the first couple of seconds to hide that glitch though.
Yeah although that would basically bork system sounds or any other quick sounds that only last for a second or so. I'm sure some folks wouldn't be sad to see system sounds go entirely, but that's another conversation...
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KILLER! Thanks guys.
I don't have much more to ask for ;)
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Who is doing the sample rate conversion that can be set in Windows sound card settings? MC or Windows?
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Who is doing the sample rate conversion that can be set in Windows sound card settings? MC or Windows?
Windows.
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OK, thanks.
When I select JRiver WDM in WASAPI exclusive (from Qobuz player) I have clicks&pops. But, in non-exclusive mode it works without problems. I have tried the maximum buffer in MC, without success. I have tried to play with clocks frequency, too.
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Try a smaller buffer.
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What is this setting, in the screenshot below, for? When should this option be selected?
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Try a smaller buffer.
The problem is the same with each buffer value (in MC, Audio, Advanced).
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Adjust the buffer size down in Options > Audio > Device settings. I use 10ms.
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Unfortunately, this doesn't solve the problem.
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What is this setting, in the screenshot below, for? When should this option be selected?
Don't mess with that ;)
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OK, thanks.
When I select JRiver WDM in WASAPI exclusive (from Qobuz player) I have clicks&pops. But, in non-exclusive mode it works without problems. I have tried the maximum buffer in MC, without success. I have tried to play with clocks frequency, too.
Unfortunately, this doesn't solve the problem.
I got Qobuz streaming in Flac pretty well --
this is what I recommend you do in this order (note every set-up is going to be different) -- even though we are talking about audio, I would get this to work for video streaming then you know the final adjustment will be in Qobuz -- get that BBC synchro video posted earlier in the thread. This is based on having set the JR driver in windows as default beforehand. Then launching JRiver (or closing and restarting it including the server if you are using it)
- create a new zone (IPC or WDM driver, streaming or whatever you want to call it :D
- in audio options for this zone; set the audio device at 50ms
- in the advanced options, start at 50ms and go downwards and sync to the video ... if you are 10ms and that still doesn't work, go back to the audio device and reduce the 50ms to a lower value [note I have found that "minimal" hardware setting is dodgy; maybe that is just on my set-up.
- once you have synced video to the audio, open Qobuz - go to Preferences (Options maybe? sorry my interface for Qobuz is in French^^); choose Flac; put their buffer to the lowest setting 64kb. If you still have cracks raise it progressively, but not too much. If there are still issues reduce or eliminate their crossfading setting.
That should do the trick provided you have a half way decent internet connection. Hope that helps some
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The "minimums" will depend on what buffering settings your devices/computer will tolerate. On all the PCs I tested, I could achieve a configuration that was good enough for lipsync without convolution (although on one machine it was verging on unacceptable). By contrast with convolution some of my machines were still solid, but some were definitely not.
I had a quick play with this earlier and it was definitely a fail with convolution on (using a minimum phase version of my usual filters) but seemed ok with convolution off. Do you have a good way to measure the actual latency in the chain?
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I had a quick play with this earlier and it was definitely a fail with convolution on (using a minimum phase version of my usual filters) but seemed ok with convolution off. Do you have a good way to measure the actual latency in the chain?
Try increasing the buffering in Options > Audio > Advanced > Live playback letency
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That should do the trick provided you have a half way decent internet connection. Hope that helps some
Merci, but no luck :(
I got Qobuz streaming in Flac pretty well --
Do you use WASAPI exclusive? I can't even use JRiver ASIO, because it seems there is an old bug in Qobuz player that force to use the first driver available in the list (and JRiver isn't the first ASIO).
I'm still in trial period with Qobuz, but I'm starting to have doubts with the player... shame for such interesting catalog
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Do you use WASAPI exclusive? I can't even use JRiver ASIO, because it seems there is an old bug in Qobuz player that force to use the first driver available in the list (and JRiver isn't the first ASIO).
I'm still in trial period with Qobuz, but I'm starting to have doubts with the player... shame for such interesting catalog
Yes, I use WASAPI exclusive for my Audio device in JRiver. NOT in Qobuz
In the Qobuz preferences, you choose the the default driver (WDM) you set in windows, which should read Default: Speakers (JRiver Media Center 20) . Don't use Wasapi exclusive that happens to come up under their menu way down the list . That shouldn't be an option even, thats not the WDM driver.
Maybe you are confusing the audio device that JRiver is using and the driver that external programs/apps would use? Haven't tried this yet with the new driver, but if you want to use ASIO, you should be able to but set from within JRiver not Qobuz.
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Yes, I use WASAPI exclusive for my Audio device in JRiver. NOT in Qobuz
Maybe, I'm missing something, here is my understanding:
1. We have one virtual sound card (JRiver) with two drivers. One is WDM (including WASAPI) and the other is ASIO.
2. And I have one physical soundcard (RME HDSP AES) with two drivers, one is WDM (including WASAPI) and the other is ASIO.
The external application sends audio to 1. (which is MC and its DSP).
And MC sends audio to 2. (which is the physical sound card).
Maybe you are confusing the audio device that JRiver is using and the driver that external programs/apps would use? Haven't tried this yet with the new driver, but if you want to use ASIO, you should be able to but set from within JRiver not Qobuz.
No single soudcard from above is addressed at the same time (or at any time) by both applications.
In the Qobuz preferences, you choose the the default driver (WDM) you set in windows, which should read Default: Speakers (JRiver Media Center 20) . Don't use Wasapi exclusive that happens to come up under their menu way down the list . That shouldn't be an option even, thats not the WDM driver.
WASAPI non-exclusive works. The advantage of ASIO and WASAPI exclusive is that it bypasses Windows mixer (re-sampling, etc...).
Here is what Matt says (http://yabb.jriver.com/interact/index.php?topic=92593.msg637547#msg637547)
For what interfaces are supported, it's any interface that talks to a WDM driver. That's WASAPI, KS, DirectSound, etc.
From what I hear, it's like a sync issue between cards (virtual and physical). For example, I had this problem in ASIO4ALL when aggregating multiple soundcards. So, anyone can confirm that WASAPI exclusive is supported through MC? Or in others words, is it possible to go bit perfect from application via MC to physical sound card?
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Next build:
Changed: The WDM driver enters silence mode after 10 seconds of silence instead of 5 seconds.
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Split: Measuring Latency of JRiver WDM (http://yabb.jriver.com/interact/index.php?topic=92734.0)
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I've been getting some odd freezes when I try to change audio tracks sometimes. I have one zone dedicated to my desktop application audio, then another for my main JRiver audio play back. Sometimes if I stop a track in my JRiver zone, Media Center hangs and I have to restart the program.
I'm using WASAPI non-exclusive mode so I can hear music and whatever application I'm using in normal use.
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A few questions about this. I'm just now playing with it a bit.
* Does it handle multichannel audio?
Does it make sense to run through the Windows Sound Control Panel's Configuration speaker test thing? I can tell you that it isn't working well for me on my server box (using just Realtek onboard audio in WASAPI mode, generally well behaved with MC).
When running the test: In the Stereo mode, the Left channels play, but the right ones are silent. In the other various modes I can get the rear left speakers to play as well, but none of them on the right do at all. Plus, my Left main speaker also plays (in addition to when Left main is supposed to play) when the Center is supposed to be, and the same happens for one of my rear surrounds.
So, I don't know if that works at all, or is supposed to... But, even if the speaker test in Windows doesn't work, does it take multichannel surround (like from Games, and other things like that that do provide stereo sources) but then automatically JRSS up to surround for actual stereo sources? Do I need to set Windows to the proper number of channels, or keep it on stereo?
Because that's awesome if I can set Windows to the "right number" and have Stereo sources still upmixed by JRSS. Right now, I have to keep Windows set at two channel, which lets my Receiver upmix those sounds, but then that breaks it when an application can actually play multichannel output (like games, or whatever).
* It has to be running, but, can you confirm that it works in Library Server mode, and not just full GUI mode? I seemed to have different results from the Windows configuration speaker test when MC was in Library Server mode, but it was acting so weird it is hard to tell. EDIT: I think this part seems to work fine in Library Server mode.
* Also, should we alter the Default Format setting in Windows for the MC WDM Driver? Does it make sense to just crank this up? Or maybe it is best to assume most other audio will be natively in some crap mode, so why have Windows resample it first? Not sure there.
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* Does it handle multichannel audio?
Yes it does. Like with any other WDM/DirectSound audio device, you will need to set the number of channels though. Only if you set it to 5.1 in the Windows options, it'll actually receive 5.1 - and it will always receive 5.1 then.
MC will then apply JRSS and any other processing.
Thats just how the whole Windows sound architecture works, unless you use WASAPI exclusive mode, Windows will always mix to whatever format is configured here before the driver sees it.
* Also, should we alter the Default Format setting in Windows for the MC WDM Driver? Does it make sense to just crank this up? Or maybe it is best to assume most other audio will be natively in some crap mode, so why have Windows resample it first? Not sure there.
If you listen to a lot of online music, 44.1k/24 is probably the best choice, or if its more gaming and general purpose audio, 48/24, unless you know that your audio is primarily of another sample rate. You would want to avoid too much resampling in Windows (even if their resampler isn't the worst in the world anymore).
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I want to play videos in MPC-HC and get the audio processed via JRiver MC (the primary reason being the absence of Pan & Scan controls in MC). So I have selected JRiver MC as the output device (Wasapi exclusive mode) in Reclock's settings. Now all I hear in MC is the static. Sometimes after seeking multiple times, I can hear the audio though but with lipsync issues. What settings should I change to get it working properly?
Also, MC's WDM driver doesn't accept greater than 24 bit audio. It would be ideal if it could accept 32 bit integer (or maybe floating point if possible) audio in Wasapi exclusive mode.
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It should never set itself as the default device, never has done so for me.
it will also only ask the first time you install it, once you give permission it only needs to update the driver and not install a new device
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It should never set itself as the default device, never has done so for me.
it will also only ask the first time you install it, once you give permission it only needs to update the driver and not install a new device
It "unset" the driver as my default device though? Is that normal?
And, unless there are changes to the driver is it necessary to have it reinstalled each time there is an update?
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It "unset" the driver as my default device though? Is that normal?
Same here, everytime I update it seems to un-default the driver.
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From what I hear, it's like a sync issue between cards (virtual and physical). For example, I had this problem in ASIO4ALL when aggregating multiple soundcards. So, anyone can confirm that WASAPI exclusive is supported through MC? Or in others words, is it possible to go bit perfect from application via MC to physical sound card?
Also interested in this. Thanks for any input!
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From what I hear, it's like a sync issue between cards (virtual and physical). For example, I had this problem in ASIO4ALL when aggregating multiple soundcards. So, anyone can confirm that WASAPI exclusive is supported through MC? Or in others words, is it possible to go bit perfect from application via MC to physical sound card?
WASAPI exclusive works fine. I just tested MC19 in exclusive mode playing to MC20 in the driver. It works without hiccups.
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For all your fine work (and the post), thanks, again!
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Only if you set it to 5.1 in the Windows options, it'll actually receive 5.1 - and it will always receive 5.1 then.
MC will then apply JRSS and any other processing.
Any way you can "detect" empty channels?
Because that would be a killer feature. The way Windows does it is super-annoying now.
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WASAPI exclusive works fine. I just tested MC19 in exclusive mode playing to MC20 in the driver. It works without hiccups.
Great! You are right. I'm testing it with foobar WASAPI plugin (push style, because event style gives BSOD) and the sound is perfect.
If anyone can test JRiver WDM WASAPI exclusive in Qobuz player, I will be interested to know if it's only me who has a problem.
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Is this driver compatible with XP? I just updated and received a warning at the point of the driver install....out of caution, I opted to not install the driver.
Thanks...
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The driver will not work on XP, sorry.
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I want to play videos in MPC-HC and get the audio processed via JRiver MC (the primary reason being the absence of Pan & Scan controls in MC). So I have selected JRiver MC as the output device (Wasapi exclusive mode) in Reclock's settings. Now all I hear in MC is the static. Sometimes after seeking multiple times, I can hear the audio though but with lipsync issues. What settings should I change to get it working properly?
Also, MC's WDM driver doesn't accept greater than 24 bit audio. It would be ideal if it could accept 32 bit integer (or maybe floating point if possible) audio in Wasapi exclusive mode.
Can anyone please tell me the settings to fix the static (distortion/ noise) that I get on my laptop or is that a bug?
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Can anyone please tell me the settings to fix the static (distortion/ noise) that I get on my laptop or is that a bug?
Increase the buffer size/latency.
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After increasing the buffer size and live latency, the audio breaks and still distortion comes intermittently. The audio breaks for direct sound too. I have tried many combinations of buffer size and latency, none of the combinations play smooth audio.
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Try decreasing the size.
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:D ;D :D Matt, you guys keep doing things like this - you've got me for life!!!!!! Good work, Matt!!!!!!! Bless you!!!!!
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Try decreasing the size.
Already tried that. It doesn't help. Also tried on an old desktop system with Pentium D, the audio breaks, skips and distortion comes on that system too. In both these systems, before this WDM driver came, I was using wasapi loopback and it worked perfectly except for the lipsync issue which was really terrible.
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Can anyone please tell me the settings to fix the static (distortion/ noise) that I get on my laptop or is that a bug?
what are you streaming from? Is it Qobuz or another app that gives you output choices??
If so make sure you choose the driver that is set in default in windows -- > Default : (Speakers) JRiver media Center 20
don't choose the WASAPI Exclusive or the ASIO mode or whatever else they offer - this will cause static as the renderer, JRiver, is already running in WASAPI exclusive or ASIO or whatever. This will not detract from "bit perfectness", depending of course on the input received.
(devs please correct me if I'm wrong about this, but I'm pretty sure the default driver must be used and it shunts the signal to the proper output driver (or device rather), either I'm misunderstanding or others are)
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I use JREMOTE to conrol JRiver on my headless computer.
Up to now I have not done any streaming of Spotify, Wimp, etc. I have just played music that is stored on my computer.
With the WDM driver can I set up JRiver so that I can easily switch back and forth, using JREMOTE, between SPOTIFY and music stored on my computer?
Thanks!
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I use JREMOTE to conrol JRiver on my headless computer.
Up to now I have not done any streaming of Spotify, Wimp, etc. I have just played music that is stored on my computer.
With the WDM driver can I set up JRiver so that I can easily switch back and forth, using JREMOTE, between SPOTIFY and music stored on my computer?
Install Airfoil (http://www.rogueamoeba.com/airfoil/windows/) ($25) and run the "Airfoil Speakers" app.
Or Shairport4w (http://sourceforge.net/projects/shairport4w/). (free, but may be less reliable)
Configure it to play to the JRiver audio device.
Now you will be able to use the Spotify app on your phone/tablet to send audio to the Airfoil/Shairport audio device, which then plays through Media Center.
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Install Airfoil (http://www.rogueamoeba.com/airfoil/windows/) ($25) and run the "Airfoil Speakers" app.
Or Shairport4w (http://sourceforge.net/projects/shairport4w/). (free, but may be less reliable)
Configure it to play to the JRiver audio device.
Now you will be able to use the Spotify app on your phone/tablet to send audio to the Airfoil/Shairport audio device, which then plays through Media Center.
Thanks, but I do not think that is what I want. I do not want to use any wifi for steaming music between my devices.
I want to install Spotify on my headless computer. I want to be able to "remote" control spotify using my iPad. Can I configure JRiver using the new WDM Driver so that Spotify output goes through JRiver?
I want to be able to go back and forth between Spotify to playing music that is on my computer hard drive using just an iPad.
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Thanks, but I do not think that is what I want. I do not want to use any wifi for steaming music between my devices.
I want to install Spotify on my headless computer. I want to be able to "remote" control spotify using my iPad. Can I configure JRiver using the new WDM Driver so that Spotify output goes through JRiver?
I want to be able to go back and forth between Spotify to playing music that is on my computer hard drive using just an iPad.
You need the latest version of the spotify desktop app. It now supports being controlled by the spotify app on your ipad.
Set the jriver wdm driver as your default audio device and your set. For convenience assign a specific zone to the wdm driver. Use "[Name]="IPC"" to setup the zone.
Personally I prefer airplay because I can control spotify from the lock screen and control center on the ipad. This does not work when you control the desktop app from the spotify app on the ipad.
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Congrats to the team for this accomplishment!!
In the opening post of this thread (http://yabb.jriver.com/interact/index.php?topic=92593.0), it says, "It gets you everything MC can do with the sound. DSP, zones, volume leveling..."
I have grown very attached to volume leveling, but I thought that it only worked by using volume information stored in the tags of the files you're playing.
Is it possible to use volume leveling on audio from streamed sources like Spotify & YouTube?!
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Is it possible to use volume leveling on audio from streamed sources like Spotify & YouTube?!
No. You could use adaptive volume to compress everything to a "normalized" level, but it wouldn't sound very good.
Volume leveling needs pre-analysis to work correctly. Real-time analysis is not going to give good results.
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what are you streaming from? Is it Qobuz or another app that gives you output choices??
If so make sure you choose the driver that is set in default in windows -- > Default : (Speakers) JRiver media Center 20
don't choose the WASAPI Exclusive or the ASIO mode or whatever else they offer - this will cause static as the renderer, JRiver, is already running in WASAPI exclusive or ASIO or whatever. This will not detract from "bit perfectness", depending of course on the input received.
(devs please correct me if I'm wrong about this, but I'm pretty sure the default driver must be used and it shunts the signal to the proper output driver (or device rather), either I'm misunderstanding or others are)
I am not using Qobuz and not any streaming services either. The output choices are offered by Reclock renderer used in MPC-HC.
Yes, JRiver Media Center 20 has been selected as the default device in Windows.
Matt has posted earlier in this thread that he was able to output audio from MC 19 in Wasapi exclusive mode to MC 20 via WDM driver. So it works in his setup and probably in the setups of other users too. In both my systems, audio breaks, skips and distorts even for Direct Sound when using this WDM driver. Besides this, I have never had such audio problems on my systems.
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Adjust the buffer size. Smaller may be better.
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I am not using Qobuz and not any streaming services either. The output choices are offered by Reclock renderer used in MPC-HC.
Yes, JRiver Media Center 20 has been selected as the default device in Windows.
Matt has posted earlier in this thread that he was able to output audio from MC 19 in Wasapi exclusive mode to MC 20 via WDM driver. So it works in his setup and probably in the setups of other users too. In both my systems, audio breaks, skips and distorts even for Direct Sound when using this WDM driver. Besides this, I have never had such audio problems on my systems.
Thats why I asked confirmation of that! (Unless he was playing to JR 20 like a DLNA device which I don't know how to do). I cannot get MC 19 to play through JR WASAPI, or JR ASIO without clicks and basic crap sound. It will work with JR20 (Direct Sound) which is not "direct sound" I think it is their WDM driver -- and that plays back fine. So if Matt could confirm ......
I think the driver, if set to default correctly, shunts over the input without going through the windows mixer ... I'm showing even though it says JRiver Media Center 20 [Direct Sound] as the output device in JR 19, it is a direct connection (bit-perfect) to the playback (rendering) handled by JR 20 using the WASAPI exclusive output (or ASIO or whatever). It is not using Window's direct sound.
Simply, I think that the external source must be "played" or outputted using the WDM driver (if there is a choice) and not another output option =>by passing the window mixer using a direct connection audio path, then it gets rendered in JR 20.
when loading the Noire skin I even get the blue button showing.
If I am wrong, then I don't know how Matt set the output on JR 19 to WASAPI or ASIO and didn't get noise over the IPC input to JR 20.
Someone could confirm this? If I'm wrong I'd like to know. If I'm right maybe instead of saying JRiver Direct Sound maybe it should say JRiver direct input or something to that effect?
PS @Aproc -- I'm not familiar with Reclock and MPC-HC. But if you are using JR's driver and JRiver as a render why use another renderer in the chain? thanks
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Sorry to be negative but I'm very disappointed in this and I've disabled the JRiver WDM device.
On my HTPC I currently use the system volume control for WMC, Cyberlink PDVD and Netflix through a browser. When using JRiver I use it's Internal Volume.
I was hoping to use the new WDM driver with WMC, PDVD, Netlfix (web browser) but how am I suppose to control volume (with an IR remote) and why does the open JRiver application take focus after I start playing something from another application?
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I have an update regarding crackling sounds with WDM enabled and playing music through browser. It seems no matter what latency settings I use, I will experience crackling sounds if I have other programs open (like ms word, adobe reader, task manager) besides chrome and jriver MC.
If any one else experience crackling sounds, try shutting down all other programs and see if this improves the situation.
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Processing sound on the CPU like this can result in crackling audio if the CPU is either not fast enough, or too busy.
Increased buffer sizes (which result in increased latency) usually help to avoid this.
However some systems just won't be fast enough, or have other issues like high DPC latency causing interruptions in playback. (crackling)
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Processing sound on the CPU like this can result in crackling audio if the CPU is either not fast enough, or too busy.
Increased buffer sizes (which result in increased latency) usually help to avoid this.
However some systems just won't be fast enough, or have other issues like high DPC latency causing interruptions in playback. (crackling)
If the cpu is not fast enough or the system is having problems with DPC latency, shouldn't this likely produce crackling sounds when playing audio only through MC (not WDM)?
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If the cpu is not fast enough or the system is having problems with DPC latency, shouldn't this likely produce crackling sounds when playing audio only through MC (not WDM)?
Playing directly to a hardware device is going to be far more reliable than playing to virtual sound device, being processed by playback software, and then played through a hardware device.
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No crackling issues or the like - but this is unfortunately unusable. No matter the buffer selections, there is a huge (.25 second or more) delay. Hope these issues get sorted out.....this was the feature that made me hit the 'buy' button.
If it matters:
Windows 8.1 Pro x64
2600K
MSI P67A-G45
ASUS STRIX GTX 970
Asus Xonar DX
Didn't test the soundcard as output device though, only HDMI to a reciever.
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Please read the instructions at the top again. You may be adjusting the wrong buffer.
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Adjust the buffer size. Smaller may be better.
5 ms buffer size fixed the audio problems in Direct Sound mode. :) It didn't fix the problems in Wasapi exclusive mode though. Any suggestion for that?
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I'm not familiar with Reclock and MPC-HC. But if you are using JR's driver and JRiver as a render why use another renderer in the chain?
When using Reclock in MPC-HC, I use JRiver for its DSP only. There are some videos that I have which have shifted frames and/ or screwed up aspect ratio. I can fix them in MPC-HC by using the Pan & Scan controls which are not available in JRiver.
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I have tried everything suggested and still have no sound using the Tidal service. I set my default device in Control Panel-Sound to Speakers - JRiver Media Center 20 - I can click on the left/right speakers and get sound. In MC I set up a new zone - set options - audio device to Default Audio Device [Direct Sound] and nothing plays - I'm using the recommended latency etc. In Control Panel-sound I can see that music is playing but no sound. I do have the volume turned up. I'm using MC 20.0.27 and Win 7 64. Any ideas?
thanks
tony
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In MC I set up a new zone - set options - audio device to Default Audio Device [Direct Sound] and nothing plays - I'm using the recommended latency etc.
thanks
tony
that's your problem -- everything pushed to the windows driver (direct sound) is being shunted to the WDM (IPC) INPUT tha you set as default. You should not be using direct sound as an output (which no longer will work because the new driver has taken over as default) Use your normal output device .. then you will get sound. If it was WASAPI or ASIO exclusive use that or a special output for your DAC, but not Direct Sound. Note that you should have jriver running before launching an external program (or at least media server20.exe)
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In MC I set up a new zone - set options - audio device to Default Audio Device [Direct Sound] and nothing plays - I'm using the recommended latency etc.
The above quoted issue is the problem; set your audio output device in JRiver to target your actual physical soundcard, not the default device. JRiver's WDM driver is the default device, so by trying to output to in JRiver you're just creating a closed loop.
EDIT: whoops ninja'd by Arindelle ;D
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AHA! That worked - thanks very much :)
t
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Blemming brilliant this - thank you :)
Incremental value could be added by allowing multiple instances of this WDM driver, e.g. (1+2), (3+4), (5+6) etc.
This way it would be possible to run multiple instance of Shairport4w each outputting audio to a different zone in MC, which would be way cool!
At the moment, I can only do this to two zones:
(1) Shairport4w => JRiver WDM => MC zone A
(2) Shairport4q => ASIOBridge => JRiver ASIO => MC zone B
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Quick question: Is the output bitdepth/sample rate dependent on the Windows settings in the Advanced settings in playback devices, or is it dependent on the settings in MC's DSP? Also, do we need to configure the multichannel output in Windows or is this also handled by the MC DSP?
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Any chance of getting MC "line out" to appear under Windows Recording Tab? Would be ideal for standalone applications that accept "What U Hear"/"Stereo Mix" as input i.e. Standalone visualizer.
Alternatively is there a way in MC to bridge output to the "Microphone - Jriver20" which does appear under Recording tab?
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Any chance of getting MC "line out" to appear under Windows Recording Tab? Would be ideal for standalone applications that accept "What U Hear"/"Stereo Mix" as input i.e. Standalone visualizer.
Alternatively is there a way in MC to bridge output to the "Microphone - Jriver20" which does appear under Recording tab?
Halfway to answering my own question with Hi-Fi CABLE & ASIO Bridge. :-)
http://vb-audio.pagesperso-orange.fr/Cable/index.htm
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Quick question: Is the output bitdepth/sample rate dependent on the Windows settings in the Advanced settings in playback devices, or is it dependent on the settings in MC's DSP? Also, do we need to configure the multichannel output in Windows or is this also handled by the MC DSP?
Yes it will be.
Everything in WDM is resampled unless the application is using WASAPI Exclusive Mode.
I don't know whether the MC driver supports WASAPI Exclusive or if it will switch though.
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I don't know whether the MC driver supports WASAPI Exclusive or if it will switch though.
Yes it does. As a test, I played in MC19 exclusive to MC20's driver. It worked fine.
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Yes it does. As a test, I played in MC19 exclusive to MC20's driver. It worked fine.
Excellent. I assumed it would work since Media Center handles all sorts of input & output configurations, but I didn't have the time to test it.
What I would say is that this update has reset all the device properties which isn't great.
I was using a custom name ("JRiver Media Center" instead of "Speakers") and icon (the app icon) with the device configured as a full-range 24/44.1 output, all of which was reset.
A number of other programs that I run use the device name to select the default audio device and had to be reconfigured as a result.
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Yes it will be.
Everything in WDM is resampled unless the application is using WASAPI Exclusive Mode.
I don't know whether the MC driver supports WASAPI Exclusive or if it will switch though.
I love the idea of the WDM driver, and I'm using it. But this extra resampling concerns me, and I'm not sure how it fits into the audio data flow when using this new driver.
I know that all Windows applications now send audio data to the MC WDM driver. I know that the MC WDM driver sends the audio data to MC. Then of course MC processes the data according to settings and sends it out to the real audio out device.
But when in the data flow does Windows resample the audio?
I'm also hearing a much larger delay before sound is played when using the WDM driver, which is of some concern, especially with short Windows event sounds. Is or can the WDM audio driver be optimised to improve response, or is this just a fact of life with Windows sound? I'm guessing that if the WDM driver was capable of WASAPI exclusive mode, thereby avoiding resampling (?), the delayed response would be reduced.
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I love the idea of the WDM driver, and I'm using it. But this extra resampling concerns me, and I'm not sure how it fits into the audio data flow when using this new driver.
Windows does high quality resampling if you're using a patched-up Windows 7, or Windows 8/8.1
Set it to 44.1kHz (or whatever your source uses) to avoid it.
But when in the data flow does Windows resample the audio?
Before it gets to MC.
As long as you aren't resampling it a second time (Windows, then MC) you probably don't have to be concerned about it.
I'm also hearing a much larger delay before sound is played when using the WDM driver, which is of some concern, especially with short Windows event sounds. Is or can the WDM audio driver be optimised to improve response, or is this just a fact of life with Windows sound? I'm guessing that if the WDM driver was capable of WASAPI exclusive mode, thereby avoiding resampling (?), the delayed response would be reduced.
The delay is because you're now playing through the driver & MC's buffer, and your sound device's buffer, instead of just the sound device.
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This rules now I can stream di.fm and use MilkDrop2 along with the rest of the MC20 awesomeness!!
Thanks for this.
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As long as you aren't resampling it a second time (Windows, then MC) you probably don't have to be concerned about it.
That's the problem. There is no setting in Windows to say "Do not resample". As I have multiple sources with different sample rates, and the WDM driver/Windows only allows me to set one sample rate, it is going to resample some sources. Also, as my receiver has limited capability, and I am resampling to the best it can do in many cases, I am likely to resample some audio twice.
I'm no audiophile, so it isn't killing me. But it is of concern. I would like to be able to tell Windows just to send the audio to MC, just as it receives it. No changes.
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That's the problem. There is no setting in Windows to say "Do not resample".
WASAPI Exclusive mode allows the application to do this, but the source application has to support it.
As I have multiple sources with different sample rates, and the WDM driver/Windows only allows me to set one sample rate, it is going to resample some sources. Also, as my receiver has limited capability, and I am resampling to the best it can do in many cases, I am likely to resample some audio twice.
Use the highest sample rate that your receiver supports for everything then. That is likely to be the least destructive option.
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It would be really great if JRiver's WDM driver can accept 32 bit integer (or floating point audio if possible) in Wasapi exclusive mode. Presently, it accepts up to 24 bit audio only.
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WASAPI Exclusive mode allows the application to do this, but the source application has to support it.
Okay. I didn't think I could combine WASAPI Exclusive mode with the MC WDM driver. I will look into that.
Unfortunately for things like Vimeo, movie trailer sites that don't use YouTube (I can run YouTube from within MC) and so on, that use a browser, the browser just plays to the default Windows audio device, so there is no opportunity to use WASAPI. I'll have to look into some of the other apps I use.
Use the highest sample rate that your receiver supports for everything then. That is likely to be the least destructive option.
That's what I am doing now. I guess I made the right choice.
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This feature alone is worth the upgrade price! Getting Internet radio stations thru the MC Audio/Connected Media menu was always hit and miss -sometimes the audio would play on the computer speakers, other times thru my DAC and hi-fi speakers. By setting up a new zone and designating MC Speakers as the default instead of my DAC device everything goes thru the DAC and hi-fi system. Is there any difference in performance when I am playing music stored on my computer or should I switch back to my original zone?
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I don't know what changed in 20.0.30 but I'm now getting stuttering every time the track changes in the Tidal app.
I think someone else mentioned similar things happening with system sounds. (I have them disabled)
I'm not sure if it's due to the connection being dropped after 10s of silence or something else like the Tidal app stopping and starting playback on track changes.
Edit: I'm also having this problem in the middle of tracks when there is a long silence. Please make the IPC cut-off a user preference rather than fixed at 5/10s.
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Could you provide an option NOT to install it during JRiver Media Center installation/update process? I do not use 99% of JRiver's functions, which is awesome because JRiver allows its users to disable almost all of them through its options. It would be a great idea if this driver could also be prevented from installation. I end up removing it from Device Manager, which is fine, but doing after each update is just extra clicking that doesn't have to be...
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Could you provide an option NOT to install it during JRiver Media Center installation/update process? I do not use 99% of JRiver's functions, which is awesome because JRiver allows its users to disable almost all of them through its options. It would be a great idea if this driver could also be prevented from installation. I end up removing it from Device Manager, which is fine, but doing after each update is just extra clicking that doesn't have to be...
It's harmless to just leave it.
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No it's not, I have a script that specifically picks the audio device called "Speakers". Of course JRiver also decided to install a device called "Speakers" and now my script picks the JRiver one instead of the other real first one that I wanted. I had to rename the JRiver "Speakers" to "JRiver Speakers" but of course a JRiver update re-installed a new version of the driver renaming it back.
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No it's not, I have a script that specifically picks the audio device called "Speakers". Of course JRiver also decided to install a device called "Speakers" and now my script picks the JRiver one instead of the other real first one that I wanted. I had to rename the JRiver "Speakers" to "JRiver Speakers" but of course a JRiver update re-installed a new version of the driver renaming it back.
I had a similar issue.
It would be nice if it could be named something like "JRiver Virtual Audio Device" by default, rather than "Speakers".
I didn't check last time, but it should also not configure itself as the default sound device when installed - which I believe is up to the driver, since some USB devices take over the default sound device when connected and others do not, for example.
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I didn't check last time, but it should also not configure itself as the default sound device when installed - which I believe is up to the driver, since some USB devices take over the default sound device when connected and others do not, for example.
It is not. Windows doesn't allow the driver to control this. It has its own logic to just randomly do this when it feels like it.
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I don't know what changed in 20.0.30 but I'm now getting stuttering every time the track changes in the Tidal app.
I'm having the same issues with the WDM driver and Tidal.
If someone figures this out I'd be appreciative.
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I'm having the same issues with the WDM driver and Tidal.
If someone figures this out I'd be appreciative.
Please read the first post in this thread (http://yabb.jriver.com/interact/index.php?topic=92593.0).
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I had already tried everything suggested in the first post (selection in sound control panel, new zone, various buffer settings).
Changing the audio device buffer settings got rid of the same type of stuttering and skips with music streamed from Spotify.
But it didn't help with music streamed from Tidal.
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Could we get the option to set how long it takes to drop the stream? I know this was talked about earlier. I build speakers, and in process of doing so I take measurements......lots and lots of measurements. I use MC for eq/xo so it needs to be in the loop. It's a pain having the signal drop after 5 seconds. Then I have to take a "junk" measurement or run a tone generator to pick it back up. When I take a measurement then change eq I've lost the stream. Or if I hesitate a bit too long when taking polar data I lose it then too.
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I made an unconfirmed as yet observation about using the WDM driver, which may assist others in solving stuttering problems.
I wanted to use Airplay to mirror an iPad onto my HTPC and use MC's WDM and DSP including Room Correction to play the audio. So I installed "Air Server" on the HTPC to enable it as an Airplay receiver. The iPad was being used to play Plants vs Zombies 2, so the stream had lots of sound effects rather than a continuous stream like music.
The audio worked, but it was stuttering, or maybe "broken up" is a more correct description. Anyway, after lots of fiddling around, changing back and forth between the real device driver and the JRiver WDM I discovered that if I set Air Server to output to the "Speakers, JRiver Media Centre 20" I got bad audio, but if I set Air Server to output to the Default Windows audio driver the audio was fine. No issues at all. Of course the MC WDM was set as the default audio driver in Windows.
So for those having trouble with the WDM driver, if you are selecting it in your application, try setting the WDM Driver as the Windows default (as advised above) and selecting the Windows Default device in your application instead, if possible. This worked for me.
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So for those having trouble with the WDM driver, if you are selecting it in your application, try setting the WDM Driver as the Windows default (as advised above) and selecting the Windows Default device in your application instead, if possible. This worked for me.
Thanks Roderick.
I wish that were possible to do in the Tidal app, but it's not (at least for me).
My sound output choices (in the Tidal app) don't include a Windows default device.
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Changing the audio device buffer settings got rid of the same type of stuttering and skips with music streamed from Spotify.
But it didn't help with music streamed from Tidal.
Tidal requires more bandwidth..
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Fantastic guys, great option. I was waiting for this for a long time (used AsioLink until now)! Great!! For the people with latency problems, turn off UAC (really off) in Windows. This solved it for me. One question: is it possible to autostart MC with Windows minimized?
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Tidal requires more bandwidth..
It's nothing to do with bandwidth. If I play directly to any of the other sound devices on my PC (including virtual ones) it sounds fine.
Going through the JRiver WDM driver results in this skipping/stuttering for the first few seconds of every track.
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Confirm that you have set the WDM driver as the default in Windows.
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Could we get the option to set how long it takes to drop the stream? I know this was talked about earlier. I build speakers, and in process of doing so I take measurements......lots and lots of measurements. I use MC for eq/xo so it needs to be in the loop. It's a pain having the signal drop after 5 seconds. Then I have to take a "junk" measurement or run a tone generator to pick it back up. When I take a measurement then change eq I've lost the stream. Or if I hesitate a bit too long when taking polar data I lose it then too.
We changed to 10 seconds of silence instead of 5 seconds.
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Sorry, but it will be for my lack of knowledge of English, and Google does not translate well from English to Italian, but I did not understand one thing: how do I tell the WDM to use a separate sound card? By default it uses the integrated card.
I guess you have to create a new zone, but that rule must be set?
thanks
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We changed to 10 seconds of silence instead of 5 seconds.
Ahh, cool. Any reason not to let the user define the value in the advanced options?
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Sorry, but it will be for my lack of knowledge of English, and Google does not translate well from English to Italian, but I did not understand one thing: how do I tell the WDM to use a separate sound card? By default it uses the integrated card.
I guess you have to create a new zone, but that rule must be set?
thanks
It uses whatever soundcard MC is configured to use. You can configure MC in Options > Audio.
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Well I finally gave up and renamed my actual speakers from "Speakers" to something else because the WDM driver kept taking it's place, thanks JRiver ;D
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It uses whatever soundcard MC is configured to use. You can configure MC in Options > Audio.
I know, the problem is that if I go to Options> Audio and choose the driver JRiver media center 20, the sound will come out from the jack on the motherboard while I would like to come out from the sound card Asus Essence.
If I go to Window in the properties of the playback device, it says clearly that it is connected to the motherboard, how do I change this link?
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I know, the problem is that if I go to Options> Audio and choose the driver JRiver media center 20, the sound will come out from the jack on the motherboard while I would like to come out from the sound card Asus Essence.
If I go to Window in the properties of the playback device, it says clearly that it is connected to the motherboard, how do I change this link?
You have it backwards; you want the windows sound device to be the JRiver media center 20 device; in JRiver's Options > Audio you want to select the Asus Essence
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This is a great addition to MC and it works perfectly for me in 3 of 4 scenarios.
MC to Devialet via Ethernet, works.
Spotify to MC to Devialet via Ethernet, works.
MC to ADL Esprit via USB, works.
Spotify to MC to ADL Esprit, fails.
When I try to play Spotify to MC to ADL Esprit, I get constant stutters/dropouts. I have tried changing buffer values and latency with no change at higher or lower values. If I change the default playback device from JR to the ADL Esprit, the problem goes away. It would be nice if all 4 options worked the same for me. Any suggestions?
Thanks!
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Any suggestions?
If you are currently pointing Spotify to the MC WDM driver, try pointing it to the Default Windows sound driver instead. Of course, have the Default Windows sound driver set to the MC WDM driver.
So;
Spotify to Windows Default audio driver = MC WDM driver to ADL Esprit
instead of;
Spotify to MC WDM driver to ADL Esprit
This simple change, which should make no difference at all, worked for me in one scenario I have.
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How do I point Spotify to anything other than the default Windows driver? I cannot find any option within Spotify to make your suggested change. Can you expand on your details?
Thanks,
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Please read the first post in this thread (http://yabb.jriver.com/interact/index.php?topic=92593.0).
You need the latest version of MC20 (build 41 or greater). You then set Windows to use its new WDM driver.
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How do I point Spotify to anything other than the default Windows driver? I cannot find any option within Spotify to make your suggested change. Can you expand on your details?
Thanks,
Sorry I'm not a Spotify user, so I can't help with settings and so on. However, if there is no setting in Spotify for the audio output device, then it must be pointing to the Default Windows sound device anyway, so my suggestion wouldn't change anything.
I suspect that you have already read the first post as Jim suggested. If you wish to find out if Spotify always points to the Default Windows sound device or can be pointed directly to another sound device, you will need to contact Spotify support.
This may not be the cause of your stuttering though, so keep asking here for more Spotify user to chime in. Perhaps start a specific thread on the topic.
BTW, I assume you are using the Windows version of Spotify, and not running it on a iDevice and sending it to your HTPC?
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Read all the posts in this thread including the first. Everything appears to be correct and is completely operational with this exception. Spotify is coming through the WDM-USB connection with poor results. Since it works for all other connections without any problem, I keep thinking that it is a setting somewhere??? I'll keep trying but can always change the default player when I am listening to Spotify at my desk.
Thanks for your help! Maybe someone will have had this issue before and I just need to wait until they get back.
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I've been having some issues, where if I leave my PC idle (like leaving my work PC idle over the weekend), the JRiver audio device in Windows seems to disappear. The only way I can get it to come back appears to be restarting my PC. Is there any other workaround to get the device to reappear (manually reinstalling/loading the driver)?
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How do I point Spotify to anything other than the default Windows driver?
You can't.
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I have been running Tidal for the last 5 days, and I was getting stuttering using WDM driver through MC20.27. The stuttering was in the first 5 seconds of every song. I increased buffer size incrementally to test, from 100, to 250 and eventually 500, this lost the stuttering in the first 5 seconds, but I still had random blips throughout the track. I had Tidal set to the MC20 WDM driver. When I set Tidal to use the IFI HD USB ASIO driver, the playback is smooth throughout the track. I have experimented with the for the last two days, and have resigned myself to using Tidal without any of the sound improvements that may be offered using MC20 w/WDM drivers. Problem seems to have something to do with buffering through MC20.
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This is a great addition to MC and it works perfectly for me in 3 of 4 scenarios.
MC to Devialet via Ethernet, works.
Spotify to MC to Devialet via Ethernet, works.
MC to ADL Esprit via USB, works.
Spotify to MC to ADL Esprit, fails.
When I try to play Spotify to MC to ADL Esprit, I get constant stutters/dropouts. I have tried changing buffer values and latency with no change at higher or lower values. If I change the default playback device from JR to the ADL Esprit, the problem goes away. It would be nice if all 4 options worked the same for me. Any suggestions?
Thanks!
Try "Fidelizer"
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I have been running Tidal for the last 5 days, and I was getting stuttering using WDM driver through MC20.27. The stuttering was in the first 5 seconds of every song. I increased buffer size incrementally to test, from 100, to 250 and eventually 500, this lost the stuttering in the first 5 seconds, but I still had random blips throughout the track. I had Tidal set to the MC20 WDM driver. When I set Tidal to use the IFI HD USB ASIO driver, the playback is smooth throughout the track. I have experimented with the for the last two days, and have resigned myself to using Tidal without any of the sound improvements that may be offered using MC20 w/WDM drivers. Problem seems to have something to do with buffering through MC20.
Did you try a very small buffer?
From the first post:
You may need to adjust the buffer size down in Options > Audio > Device settings. I used 10ms.
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I mentioned on the Tidal thread I can't test it where I am, but Qobuz which I believe to be similiar (lossless streaming) has in its config panel a buffer (tampon memoire if any frenchies are reading this). I have set this to a very low setting (64kb). If Tidal has the same setting try setting it to the lowest possible buffer if your internet connection can handle it.
Otherwise, what Jim is saying is what you want to do, even if it seems like you should go the other way around. I have set up three machines on one
All but one machine has both the audio device settings buffer and (under advanced) the latency are set below 50 milliseconds. I'd avoid the minimum hardware setting though. Remember to juggle both.
First I create a zone, call it streaming or whatever. Then I start first to set the device setting at 50 milliseconds and move the latency setting down gradually starting at 50ms. If it gets to 10, go to the buffer setting and decrease that gradually. Then do an audio/video sync to fine tune between the two parameters.
Also uncheck memory playback option in JRiver (can't help, could hurt) as these streams are stored in a cache anyways
If you are using multiple machines, remember that each macine will be different
EDIT -- hmm, maybe this is not the only problem ... see this post http://yabb.jriver.com/interact/index.php?topic=92913.0 (http://yabb.jriver.com/interact/index.php?topic=92913.0)
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given this WDM disaster i strongly recommend that JRiver staff take the time and properly rewrite the correct instructions on how to use this thing. after a lot of frustratingly wasted time i did the smart thing and keep the thing disabled.
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Did you try a very small buffer?
From the first post:
I tried that the last couple of days, 50, 25 and 10, that causes stuttering/dropouts in the first 10 seconds of every track. Then I started going the other way, 100. 250 and 500, but then I got stuttering/dropouts randomly throughout the track, this is with the MC20 WDM driver. If I use the IFI HD Audio driver, I get smooth play, no stuttering/dropouts, so it looks like I can't use the MC20 WDM driver combination. I've been all over the settings, restarting Tidal and MC20 each time settings were changed, to no avail.
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given this WDM disaster i strongly recommend that JRiver staff take the time and properly rewrite the correct instructions on how to use this thing. after a lot of frustratingly wasted time i did the smart thing and keep the thing disabled.
I certainly wouldn't call it a disaster.
I would say that it has the same problem as most new features which is that they are not fully documented when released, may not be in a "final release" state on their initial release (though I'm not sure if it's on the "stable" branch yet?) and the program itself doesn't explain anything about them - it seems to be assumed that you're following the forums.
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Sorry, I misspoke. I just toggled the "disable event style" setting as you recommended and had those results. So it seems there are a lot of us that are having similar problems with the WDM when using a streaming service to a local USB device? I have gone up and down the buffer/latency scale with no improvement. For now, I will just disable the WMD to listen to Spotify. Maybe this problem will resolve itself with time and updates. Three out of four isn't bad . . . .
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Works with Amazon Prime Music (http://yabb.jriver.com/interact/index.php?topic=93279.0).
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Build 37 may fix the problem with the driver being set to default by Windows.
http://yabb.jriver.com/interact/index.php?topic=93304.0
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Could we get the option to set how long it takes to drop the stream? I know this was talked about earlier. I build speakers, and in process of doing so I take measurements......lots and lots of measurements. I use MC for eq/xo so it needs to be in the loop. It's a pain having the signal drop after 5 seconds. Then I have to take a "junk" measurement or run a tone generator to pick it back up. When I take a measurement then change eq I've lost the stream. Or if I hesitate a bit too long when taking polar data I lose it then too.
I second this request for exactly the same reasons. Either adjustable, or a check-box to turn it on and off would be dandy.
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I second this request for exactly the same reasons. Either adjustable, or a check-box to turn it on and off would be dandy.
Just in case you missed it earlier:
We changed to 10 seconds of silence instead of 5 seconds.
It's a little better for you, I hope.
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Yes, I did see that. Adjustable or a switch would even better. (don't want to fight my tools)
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10 seconds still isn't enough for speaker design and many measurements. I tried it again last night and after dropping the stream many times switched back to HiFi Cable.
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+1 for making it a preference. Since it stutters every time it initializes, I'd want it to remain open for several minutes.
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+1 for making it a preference. Since it stutters every time it initializes, I'd want it to remain open for several minutes.
Same here! Would be really nice.
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I have no plans on using the WMD driver. Is there any harm in letting it install anyway? Is there a way to turn off the install?
The sound card I use for MC is not my default card and I don't want windows noises to ever come out of my audio system speakers.
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I have no plans on using the WMD driver. Is there any harm in letting it install anyway? Is there a way to turn off the install?
The sound card I use for MC is not my default card and I don't want windows noises to ever come out of my audio system speakers.
From build 29:
The installer can be run from the command line with the /NoDriver switch to skip the driver install, but it does no harm to have it installed.
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Split WASAPI Modes (http://yabb.jriver.com/interact/index.php?topic=93400.msg642730#msg642730)
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Not sure why you split my posts into the other thread. The issue I have is unrelated to wasapi event style. Toggling wasapi event style on and off has no affect on my problem. The wdm driver works great when inputting directsound. It crackles when inputting wasapi. Not on all sources however. Local files play fine. Some streaming files crackle. It is hit and miss what works and what doesn't. Most files I've played from soundcloud crackle. About half the youtube videos I've watched do not. I'm done posting on the issue however. If Matt eventually figures it out great, if not directsound works fine and is a good enough solution.
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In audio options, did you try small buffer sizes? 10, 50, 100, for example.
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If you reread my posts I've tried every combination of buffer size. Nothing changes the crackle problem with wasapi input into jriver. Again I can use directsound. If I feel the resampling is an issue for my local audio files I'll set jriver up as my audio player again. Just trying to avoid it, as the biggest bonuses to this for me would be eq without windows resampling (can already use equalizer apo with directsound out) and of course convolution, all while using xbmc.
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If you reread my posts I've tried every combination of buffer size.
I'm sorry, but I don't re-read every post before I offer a suggestion. I hope you understand why.
I can't imagine why you would get crackling on WASAPI but not DirectSound.
The most common reason for any sound problem with WASAPI is having the "Event Style" setting wrong for the DAC.
I don't think you're seeing a JRiver bug.
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I'm sorry, but I don't re-read every post before I offer a suggestion. I hope you understand why.
I can't imagine why you would get crackling on WASAPI but not DirectSound.
The most common reason for any sound problem with WASAPI is having the "Event Style" setting wrong for the DAC.
I don't think you're seeing a JRiver bug.
What I'm seeing is 100% a bug with the JRiver audio driver. I installed the new JRiver on my media server in the bedroom, and installed the latest version of Kodi. Same exact issue with completely different hardware, only much worse. Using wasapi in to JRiver caused my computer to blue screen. For what it's worth, changing the latency and buffer settings to hardware minimum caused my other pc to blue screen even when using directsound in to JRiver. This is a great feature, but it's essential in beta right now.
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I don't have any help to lend but, for what it's worth, berstuck is not alone in having trouble inputting WASAPI to the WDM driver. I posted a thread about noise and blue screen issues here: http://yabb.jriver.com/interact/index.php?topic=93044.0
A couple people chimed in with similar issues in that thread, and I have seen comments about it in other threads as well.
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Are you using 20.0.37?
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No, 20.0.27. I'm on the stable update path.
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Like berstuck, I suffer from crackling sounds no matter what buffer size I set or whether event style is disabled or not. Using 20.0.37.
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No, 20.0.27. I'm on the stable update path.
Try the build from the top of this board.
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Try the build from the top of this board.
Jim,
I tried 20.0.37 now but I'm afraid the problems with white noise and BSOD when inputting WASAPI into the WDM driver are exactly the same. I can replicate the problems 100% of the time.
On a separate but also unfortunate note, I cannot configure the .37 WDM driver to more than 2 channels (the configure option is greyed out in the settings for playback devices), so I guess I have to revert to .27... The icon for the JRiver playback device is now two RCA plugs instead of a speaker, if it helps.
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I tried 20.0.37 now but I'm afraid the problems with white noise and BSOD when inputting WASAPI into the WDM driver are exactly the same.
Please explain exactly what you mean by "inputting WASAPI...".
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Please explain exactly what you mean by "inputting WASAPI...".
I mean playing media in some Windows application (other than JRiver), using WASAPI as opposed to Directsound for the audio, and using the new JRiver WDM driver as the audio device in Windows system (or the media application) settings.
I got the impression in some threads that the issue is confused with routing audio FROM JRiver to a sound card or DAC using WASAPI, which is not the case.
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MC did it again . . . WDM set itself as default driver upon install of the update in spite of it having been set as DISABLED in my system.
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The installer can be run from the command line with the /NoDriver switch to skip the driver install, but it does no harm to have it installed.
When you do this, do you have to use /NoDriver every time you update MC20? Or does it remember that the driver was not installed and thus won't attempt installing it when updating?
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Had some really weird behavior today. Was in the middle of playing a game using the WDM driver, but then my IPC auido completely died and deafening static starting going full blast. I redirected my Windows audio output directly to my soundcard instead of JRiver, and audio was fine. Any idea what nuked the WDM driver here?
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Had some really weird behavior today. Was in the middle of playing a game using the WDM driver, but then my IPC auido completely died and deafening static starting going full blast. I redirected my Windows audio output directly to my soundcard instead of JRiver, and audio was fine. Any idea what nuked the WDM driver here?
That happened to me once a couple builds ago.
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That happened to me once a couple builds ago.
I ended up rebooting and it seems to work now. I'm on the Latest branch if that helps.
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Ahh, cool. Any reason not to let the user define the value in the advanced options?
Any thoughts from the MC team on this, wrt keeping the wdm stream open?
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Is this restricted to 2 channel stereo? Just installed the latest version of MC 20 to play with this and watching LEGO Marvel Super Heroes on Amazon Instant my AVR is showing Stereo.
But that might be because the stream from Amazon only contains 2 channel stereo. Will try a film and see what's what.
Just tried Riddick as well and that's showing Stereo too...
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Hmmm. Watching Brave DVD which I know is 5.1 but it's showing as Stereo.....
EDIT: Ha ha, what an idiot. Brave default soundtrack is 2.0!!
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It changed a few versions ago. The driver is stereo-only right now.
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Ah OK. That's cool, just wanted to make sure it wasn't something wrong my end.
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The new driver does not work on my system. No sound whatsoever unless I disable the driver and go back to original Realtek drivers.
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The new driver does not work on my system. No sound whatsoever unless I disable the driver and go back to original Realtek drivers.
Are you running MC or Media Server? It's required if you're using WDM as the Windows default driver.
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WDM routed spotify and chrome audio through MC fine. My problem is that it deleted everything I had in my "playing now" list everytime I play sounds from another program. Is there an easy way to route other audio traffic through another zone or something to stop me from losing my play list each time?
thanks
dbdog
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WDM routed spotify and chrome audio through MC fine. My problem is that it deleted everything I had in my "playing now" list everytime I play sounds from another program. Is there an easy way to route other audio traffic through another zone or something to stop me from losing my play list each time?
thanks
dbdog
Yes. You need to setup a second zone for the WDM driver input and setup a zoneswitch rule to route the audio to that zone. Check out the instructions in this thread: http://yabb.jriver.com/interact/index.php?topic=93043.0
The method described over there is designed to deal with issues related to streaming video, but the zone setup described should solve your problem too.
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Waiting for the Record button ... :P
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On my system the WDM works very well for some sample rates but not with outhers. Let me know if I can help debug this.
System: Player is JRMC17, DAC is www.henryaudio.com in USB Audio Class 2 ASIO mode, OS is Win7-64. Source: wav files converted to the 6 classical sample rates (44.1 up through 192) by JRMC17 or 18.
The DAC and source material work very well with MC in normal operation. With 20.0.41's WDM I'm OK with 44.1, 88.2 and 176.4ksps, but get silence with 48, 96 and 192.
Setting the protocol to WASAPI event style in the player, the DAC (operating through WDM and ASIO driver) is correctly set up to use those three sample rates. With WASAPI (not event style) I get an unreliable playback. With Direct sound I also get audio on 44.1, 88.2 and 176.4ksps material, but now everything is resampled to 44.1. With DS I also get silence on 48, 96 and 192.
I kind of expected DS to resample all 6 rates to the default sample rate set by the OS, and both kinds of WASAPI to forwared the source material 1:1 without sample rate conversion.
Best,
Børge
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this is my problem:
i've always used xbmc like frontend for movie and jriver like external player for view the movie selected on xbmc (xbmc not support ASIO).
with new wdm driver yesterday i've try to tell to xbmc to output on audio driver WASAPI JRiver WDM driver, actually the jriver wdm driver isn't default windows driver because i want to output audio of windows on tv (hdmi).
Whit this setup when i launch a movie inside xbmc the video starting and the audio output correctly through jriver on my ASIO card (Ipc play) but the audio glitch and if i try to change an options during play i take a good WINDOWS BLUE SCREEN!!!
practically i want to use a jriver wdm driver like bridge between wdm and asio (for app like xbmc that not support asio) and apply all dsp to output asio device.
Why?
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Yes. You need to setup a second zone for the WDM driver input and setup a zoneswitch rule to route the audio to that zone. Check out the instructions in this thread: http://yabb.jriver.com/interact/index.php?topic=93043.0
The method described over there is designed to deal with issues related to streaming video, but the zone setup described should solve your problem too.
Thanks for trying to help. As usual when I fiddle with technical stuff, it didn't work for me. I don't have time or skills to trouble shoot this.
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If you don't want WDM, turn it off in Windows Control Panel Sounds. Then install from the command line, adding /NODRIVER after the name of the install file.
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Please use this thread for problems: http://yabb.jriver.com/interact/index.php?topic=93720.msg646166#msg646166