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[berstuck]

WDM works for me with buffer, and latency both set to 10ms.  If I set them both to hardware minimum I get random pc crashes.  For example using xbmc with output device set to wasapi: JRiver speakers, and playing videos on the youtube app, switching videos causes my pc to reset.

[berstuck]

This works for me with buffer, and latency both set to 10ms.  If I set them both to hardware minimum I get random pc crashes.  For example using xbmc with output device set to wasapi: JRiver speakers, and playing videos on the youtube app, switching videos causes my pc to reset.

I would also like to add that I get crackling when using wasapi no matter what the buffer and latency are set to.  I only get crackle free audio on streaming content when using directsound.

[JimH]

Try toggling the WASAPI option called "Disable Event Style".

[gswaul]

Disabling WASAPI just makes it worse and goes from the skipping to the crackling problem.  It stops only when I switch away from JRiver as the default playback device in this one connection.

[JimH]

Don't disable WASAPI.  Please re-read what I wrote.

[berstuck]

Try toggling the WASAPI option called "Disable Event Style".

If this was in response to my post saying I get crackling unless I use directsound I believe you misunderstood.  JRiver is always set to output wasapi that is not the issue.  When using xbmc as my front end software and using wasapi:jriver speakers as my output it is impossible to avoid crackling on multiple different streaming sources.  No change to latency or buffer settings changes that.  I have to set xbmc to directsound:jriver speakers for this to work.  Of course that means resampling all of my high res audio files or having to set up a dual front end solution and use jriver for music.  Unfortunately I was really hoping this update will allow me to get rid of some complexity.

[RoderickGI]

If this was in response to my post

It was in response to you, and once again you have misunderstood what was said. There was a clarification post just after that post as well, for gswaul.

WASAPI has two mode: Event Style, and Non-event style (which was an earlier version of WASAPI, as I understand it.)

Don't turn off WASAPI. Change it from Event Style to Non-event style, by toggling the WASAPI option called "Disable Event Style".

It may not improve things. It may make it worse. But at least try what was suggested.

Then if it doesn't work for streaming sources, but Direct Sound does, study up on Zones, and automatic zone switching.

[Arindelle]

+1 to Rodericks response

When using xbmc as my front end software and using wasapi:jriver speakers as my output it is impossible to avoid crackling on multiple different streaming sources.  ...

Wait a second, you are using XBMC as a front end?  Can't tell why you are having this issue if you have another media player in the audio chain  (not the best plan for quality sound IMHO either, but not going there ... )

First try JRiver as your "front end" and output. Set your latency up so it works. THEN add XBMC. If you have problems, its not with JRiver.

Of course that means resampling all of my high res audio files or having to set up a dual front end solution and use jriver for music.  Unfortunately I was really hoping this update will allow me to get rid of some complexity.

Just to understand, what makes you want to not get rid of XBMC, Berstuck? Eye candy? or some technical feature that JR lacks?. Resampling files seems like a crazy lot of work and might be interesting to have a comparative of features in a separate post from someone who is used to the two players.

[berstuck]

It was in response to you, and once again you have misunderstood what was said. There was a clarification post just after that post as well, for gswaul.

WASAPI has two mode: Event Style, and Non-event style (which was an earlier version of WASAPI, as I understand it.)

Don't turn off WASAPI. Change it from Event Style to Non-event style, by toggling the WASAPI option called "Disable Event Style".

It may not improve things. It may make it worse. But at least try what was suggested.

Then if it doesn't work for streaming sources, but Direct Sound does, study up on Zones, and automatic zone switching.

At no point in time have I misunderstood anything.  If you scroll back up and pay attention to the names of the posters maybe you'll have a better grip on this conversation.

[berstuck]

+1 to Rodericks response

Wait a second, you are using XBMC as a front end?  Can't tell why you are having this issue if you have another media player in the audio chain  (not the best plan for quality sound IMHO either, but not going there ... )

First try JRiver as your "front end" and output. Set your latency up so it works. THEN add XBMC. If you have problems, its not with JRiver.
 

Just to understand, what makes you want to not get rid of XBMC, Berstuck? Eye candy? or some technical feature that JR lacks?. Resampling files seems like a crazy lot of work and might be interesting to have a comparative of features in a separate post from someone who is used to the two players.

I use JRiver for multizone audio (pool etcetera) but no longer as my htpc front end for numerous reasons.  The biggest of which is lack of pixel shader support.  As to your comment about sound quality it makes absolutely no sense.  You could chain audio through 10 different programs and if it isn't resampled the only thing that would be affected is latency.  Also this wdm driver was obviously written for people to use jrivers audio engine with other programs making your comment even more confusing.

[RoderickGI]

At no point in time have I misunderstood anything.  If you scroll back up and pay attention to the names of the posters maybe you'll have a better grip on this conversation.

Actually I have a good grip of the conversation thanks. Jim's comment was a response to you, even though gswaul piped up in the middle. You didn't respond to the suggestion to turn off WASAPI Event Mode, which was made to you.

Maybe you are just choosing not to clarify your use of WASAPI Event Mode, and want to emphasise that it works with Direct Sound.

But you may wish to note that JRiver MC uses WASAPI Event Mode by default. Event Mode can be turned off. This may improve the sound you are getting, or may not. But the point was;

Have you tried turning off Event Mode for WASAPI?

EDIT: I see you answered the Event Mode question in the WDM Driver thread. I guess this thread is dead then.

[Arindelle]

I use JRiver for multizone audio (pool etcetera) but no longer as my htpc front end for numerous reasons.

Just sometimes users, including myself, are not aware of some advanced features that exist which could make doubling software (or hardware for that matter) redundant. You evidently do not fall into that category as you seem to be well aware of every possible JRiver option. So if functions are lacking, potentially these could be isolated to be incorporated in future updates for the benefits of other users not as knowledgeable as yourself?

The biggest of which is lack of pixel shader support.

I see. Well maybe you could help me then?  I thought that there were 7 pixel shader options supported .. I guess I was wrong. What do theses options in this screenshot do?  Thanks in advance. Don't mean to hijack the thread, however

You could chain audio through 10 different programs and if it isn't resampled the only thing that would be affected is latency.  Also this wdm driver was obviously written for people to use jrivers audio engine with other programs making your comment even more confusing.

Gosh thanks, didn't know you could use multiple software packages in the chain and only affect latency. I did think it was this latency that was your problem. Trying to be empirical, I presumed it would be good to start with setting latency variables to work directly from an audio source external to the program (like a browser or a streaming service) first. Then adding the "front end". Maybe even try the asio driver which I heard had less latency and could be fine tuned. Looks like I'm not helping much, obviously.

As to your comment about sound quality it makes absolutely no sense.

Ah, yes I said I wasn't going to go there, and mentioned it anyway!   I'm just old-school and like to put the minimum between source and DAC in my audio chain. But what do I know! It  was a digression.  My bad, sorry.