INTERACT FORUM
More => Old Versions => Media Center 17 => Topic started by: Matt on February 20, 2012, 05:03:52 pm
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Tonight's build allows playing the audio that's going to the default Windows sound card through the JRiver engine. It does this using WASAPI Loopback.
This is a poor man's virtual sound card. It can't do latency compensation, so if you use processing that adds latency, lip-sync will not be perfect.
To use this feature, you need to set the default Windows playback device to something other than the device Media Center is using. Then, start playback in Media Center using:
File > Open URL... > live://loopback Using a small buffer size in Options > Audio > Output mode settings... is recommended.
Sounds that go to the default Windows device will also go through Media Center.
As an example of a real-world use for this, I have a Desk and Theater zone at home. The Desk zone is where sounds from web browser, Netflix, and games go. When playing in the theater, I need those sounds redirected to my Theater zone where Media Center does bi-amping, etc.
I can now start the Theater zone playing live://loopback and then play a game or Netflix on the projector and get nice sound. Previously, I could switch the default Windows audio device but since Windows doesn't know how to handle the bi-amping of my subwoofer, all the sounds went to the wrong places.
Please keep in mind that this is an experimental project. There's no guarantee it will remain or be developed farther for v17.
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This works! I'm going to have to try it with multi-channel stuff at home on my Steinberg UR824. This USB audio device, unlike the firewire one I also have, only works in stereo with Direct Sound sources. I wonder if games will now be routed to all channels.
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I just streamed "Person of Interest" from CBS and upmixed it using JRSS. Pretty cool!
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I normally only play NBA2K12 on the treadmill, but I made an exception tonight so I could test this feature. It worked great. Durant put up a team record 72.
Fun stuff.
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Matt,
that's absolutely great!
The problem I have is that only ASIO provides multichannel support for my audio interface - the WDM driver only shows 4 stereo pairs as outputs. So to make use of this feature I would need it to route the system audio to my audio interface outputs via ASIO.
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I tried this at home last night with my HTPC and Steinberg UR824. The sound was always distorted and had dropouts. I tried changing hardware buffers and buffering in JRiver, but couldn't get anything to work. I didn't have any other audio devices so I download Virtual Audio Cable and used it. That is what I am using at work as the default device, but maybe I don't have it configured the same. I would think the loopback intercepts the audio before the device anyway so it shouldn't matter what audio device is used as the default as long as it is different than the one used in JRiver.
As I drove home from work last night I realized this should work for running REW sweeps through JRiver. I just tried it and if using the JAVA driver in REW with output set to Default Device it works! I also tried it with Parametric EQ and Mix Channels and you can route sweeps to any speaker or combination of speakers! If this doesn't really pan out for audio playback, it might be worth implementing just for the ability to work with REW.
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Great Function!!!
currently, I am playing game with JRiver engine handle sound part. ;D ;D
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As I drove home from work last night I realized this should work for running REW sweeps through JRiver. I just tried it and if using the JAVA driver in REW with output set to Default Device it works! I also tried it with Parametric EQ and Mix Channels and you can route sweeps to any speaker or combination of speakers! If this doesn't really pan out for audio playback, it might be worth implementing just for the ability to work with REW.
Great news, I'm going to try it later.
What do you mean by this statement: "using the JAVA driver in REW with output set to Default Device"? Is there something extra that needs to be done with Java?
Thanks again for all your efforts in getting REW to work with MC.
Rod
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The problem I have is that only ASIO provides multichannel support for my audio interface - the WDM driver only shows 4 stereo pairs as outputs. So to make use of this feature I would need it to route the system audio to my audio interface outputs via ASIO.
ASIO multi-channel output is fine in Media Center.
You just need a different device (say a cheap motherboard soundcard) to set as the default device. Then, you play loopback in Media Center out the ASIO and when Windows sends things to the motherboard soundcard, they go out Media Center.
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What do mean by this statement: "using the JAVA driver in REW with output set to Default Device"? Is there something extra that needs to be done with Java?
The latest version(s) of REW lets you select between a JAVA driver or ASIO driver. Older versions (I think before July of last year) only use the JAVA driver so no selection is required. Only the JAVA driver lets you then select Default Device so that it works with this loopback feature. If you had multiple ASIO devices and set one as the default, I guess the ASIO output should also work.
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ASIO multi-channel output is fine in Media Center.
You just need a different device (say a cheap motherboard soundcard) to set as the default device. Then, you play loopback in Media Center out the ASIO and when Windows sends things to the motherboard soundcard, they go out Media Center.
I'm in the same situation on my HTPC as TheLion. I think the problem is that when the default device uses Direct Sound, the sent audio from the game, etc. only goes to the first stereo pair. Then when it is looped back to JRiver there are only two channels. Maybe a fake ASIO default driver needs to be made and then the signal can be looped back to JRiver using the WASAPI Loopback. Actually, that still wouldn't work because it is the sending program (Game, etc.) that is using Direct Show output. Probably the hardware company just needs to make proper drivers. The Steinberg MR816 (Firewire) finally had proper multi-channel windows drivers in the last driver release. It took several years from when the product was released. The recently released UR824 (USB) does not have multi-channel windows drivers.
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I'm in the same situation on my HTPC as TheLion. I think the problem is that when the default device uses Direct Sound, the sent audio from the game, etc. only goes to the first stereo pair.
I would think it would work to configure your default device to be 5.1. Then the loopback format should be 5.1.
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I would think it would work to configure your default device to be 5.1. Then the loopback format should be 5.1.
You're right, the default device determines the output channels. I'm going to do some more testing during lunch.
If JRiver is handling the audio, the internal volume control needs to work at all times. When the loopback is live, all volume controls should be bound to JRiver.
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REW latest V5 per Matt & Mojave's settings is now sending sweeps to all 5.1 speakers. Wonderful!
But....... (Why does there so often have to be one?) As the sweep is happening I get very loud ticks and pops. This may be related to the faint ticks I got in Theater View when background images change listening to music.(I have that feature turned off so no longer a problem). Reported here: http://yabb.jriver.com/interact/index.php?topic=68361.msg460080#msg460080 (http://yabb.jriver.com/interact/index.php?topic=68361.msg460080#msg460080) But this is so loud that it makes REW unusable through MC.
Others with Asus Soundcards have reported it here http://yabb.jriver.com/interact/index.php?topic=70206.0 (http://yabb.jriver.com/interact/index.php?topic=70206.0) and here http://yabb.jriver.com/interact/index.php?topic=62062.0 (http://yabb.jriver.com/interact/index.php?topic=62062.0)
Since I wasn't having any luck resolving it I just turned off the changing of background images in Theater View. Looks like now I need to get to the bottom of it because I really want REW to work through MC.
Rod
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Now that build 17.0.91 is public can we talk about this feature elsewhere (like friends, family, relatives, etc. ;D), or is it still "beta"?
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Now that build 17.0.91 is public can we talk about this feature elsewhere (like friends, family, relatives, etc. ;D), or is it still "beta"?
Experimental means there's no promise that it will work, or that we'll do additional work on it.
It's interesting, but it's too soon to say what we'll do with it.
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Matt
"It's interesting, but it's too soon to say what we'll do with it..."
...Please, please keep it going.
There is a growing thread at computeraidiophile.com about PC based digital room correction outputting to high-end multi-ch DACs and the capability of MC to access other PC streams, e.g. MOG, etc and external hardware sources, e.g. Tivo, etc running through your advanced DRC engine is key.
The implication of this feature is that stereo or multi-ch room correction down to the driver level, can be possible without adding hardware in the signal path and without additional a/d d/a conversion after the DAC.
Please don't give up on this.
Here is the thread: http://www.computeraudiophile.com/content/Room-correction-Stereo-Surround
Thanks for your amazing product!!
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I've been using the loopback successfully with games and Room Equalization Wizard. It works great with my 8 channel USB DAC that only allows for stereo input when playing games. I installed the motherboard audio drivers and use that as the default audio device set to 7.1. It took a little digging to figure out how to turn the motherboard's autosense feature off so I could set the motherboard as 7.1 channels even though nothing was connected to it.
I have a few suggestions:
1. It would be nice if we could set JRiver to be in loopback mode when not otherwise playing audio. This makes it more automatic at intercepting audio from other sources.
2. One has to use internal volume control with the loopback since the default audio device isn't playing anything and therefore the master volume control doesn't do anything. I can only use the internal volume with my keyboard with JRiver is the active screen. Since no other device will have audio in loopback mode, JRiver should take exclusive use of the volume controls.
3. Before installing the motherboard audio drivers, my USB DAC was my only audio device. To make it easy to use the loopback, JRiver should provide a fake 7.1 channel audio device driver. Enabling the loopback would automatically set this driver to default and start the loopback process.
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I've been using the loopback successfully with games and Room Equalization Wizard. It works great with my 8 channel USB DAC that only allows for stereo input when playing games. I installed the motherboard audio drivers and use that as the default audio device set to 7.1. It took a little digging to figure out how to turn the motherboard's autosense feature off so I could set the motherboard as 7.1 channels even though nothing was connected to it.
Can I ask what onboard sound you have and how you disabled autosense? I wanted to use this loopback a while ago but could get the on board sound enabled without something plugged into all the jacks. I have a realtek HD on board and it has the autosense thing. I found where I could "disable front panel jack detection", but this only enables the front panel which is only 2 channel and obviously not what I want.
Thanks
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Well I just took a pair of broken earbuds and cut the plug off the cable and just plugged that into the front speaker jack on the back which enabled the back outputs and allowed me to select them as the default audio device and set it to 6 channels.
I have found a different problem though. In DSP studio I set the channels to 5.1, turned on JRSS, enabled room correction, set the crossover for each speaker to 80Hz and moved bass to subwoofer. I then played a youtube video in firefox and noticed that sounds weren't being routed to the sub correctly. Looking at the analyzer shows all frequencies going to the sub instead of just what I routed in Room Correction. Then playing around I noticed that if I select "move center to front L/R" then routing to the sub is correct and only plays frequencies below my crossover point. With these same settings and playing audio from within MC everything works correctly either way.
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I have the Realtek HD Audio on the motherboard on my work PC and home HTPC. Disabling front jack detection in the Realtek HD Audio Manager is what actually enables the analog output from front and rear outputs. I then made sure the device was set as default and the proper number of speakers was selected (5.1 or 7.1).
. . . there are cases where small buffer sizes are important to reduce latency, like when using Media Center to process and output the audio from a game or web browser (which is a feature that's in development).
Its good to see that this feature is still in development. ;D
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Its good to see that this feature is still in development. ;D
This feature has been working well for me.
It just needs to be made easier to use, which we'll do at some point.
I was worried about latency, but at least in my installation, this isn't a problem. I have a bunch of input lag on my projector (my biggest gripe with the Epson 3010) so the audio is ahead if anything.
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I have a bunch of input lag on my projector (my biggest gripe with the Epson 3010) so the audio is ahead if anything.
Maybe you should use the "two built-in 10 W stereo speakers for amazing audio."
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I'm still having problems with this. The problem only happens when JRSS or Clone mixing is enabled and I'm playing back any 2 channel audio. If I playback a movie encoded with 5.1 channels everything is fine, I'm assuming that's because MC is smart enough to not use JRSS on already 5.1 material.
If I already have some 2 channel audio playing back and then start loopback in MC while watching the Analyzer, I can see the subwoofer signal and it is looks correct and is rolling off at what appears to be the correct frequency I have set in room correction. After maybe 4 or 5 seconds the subwoofer signal in the analyzer starts showing full playback of all frequencies far above the crossover point like there is no filtering happening and you can hear an obvious change in sound coming from the sub.
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You might look at Audio Path to see the input, processing, and output information. Does it offer any clues?
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Audio Path
Input: 44.1kHz 32bit 6ch
Changes: Room Correction
Output: 44.1kHz 24bit 6ch using ASIO
Room Correction is set with all channels crossover at 100Hz, Routing is Move bass to sub.
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If I understand this correctly, I can then use any windows application, like Mediaportal or DvbViewer,
which are my preferred movie and tv front-ends (sorry), and then let JRiver handle the audio with all of it's superior DSP, eq, convolver etc..?!?
If that is correct, PLEASE PLEASE continue working on this... ;D
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Am I correct that Windows Media Center can now be looped back through MC for digital x/o, room correction etc?
I have to use Windows Media Center for CATV tuner cards e.g. HomeRun Prime, which I believe only play through WMC, due to DHCP protection.
Can I now route the audio from WMC through MC as described in the thread?
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Hi
I finally got my Mytek 8x192 DAC :-) and am now trying this for x/o and DRC.
Even with no DSP, I hear sound, but is "flutters", ie skips on/off very fast continuously. I have Realtek HD as my default device on a 64bit win 7 i5. In MC, Im using ASIO to the mytek via firewire.
Has anyone experienced this? Im really looking forward to it!!
Thanks
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Has anyone experienced this? Im really looking forward to it!!
You need to use a low latency setting for the output in Options > Audio > Output mode settings...
I had good luck with 0.05 seconds and ASIO. WASAPI - Event Style would be similar.
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Thanks Mat
Here's what I'm finding:
1) AISO Default MC settings: Flutters, as described above
Audio Path: In = 44.1/32bit Out= Same
2) WASPI Default MC settings: Error message saying "Playback could not be started using 44.1/32bit. Playback could be started using 44.1/24bit. Then it plays, but still stutters
4) WASPI Must have DSP change bit rate from 32 to 24. Then if reduce buffering to 50ms or 25ms plays well. If 10ms breaks up again
5) ASIO: Can remove DSP and play as 32bit. If reduce buffering to .03-.05s plays beautifully!!! Audio Path: In =44.1/32bit out same
Problem solved, but may I followup with a few questions?
a) Is there magic to buffering? Do I want it as large as possible before fluttering returns?
b) Why "native" bit rate 32bit? Can/should I set to 64 bit? Why does 32 bit play is ASIO and not WASPI? Does this mean I should use ASIO or doesn't matter?
This is a GREAT feature!!!!!!!!!!!!!!
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There's lots of information on bitdepth here (take note that bitdepth is ignored with ASIO):
http://wiki.jriver.com/index.php/Audio_Bitdepth
Buffering is tricky. For playback, normally latency doesn't matter so a big buffer is good. But for live playback like this, latency is bad so you want a smaller buffer.
So to use this feature, find the smallest buffer that never hiccups and use it.
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There's lots of information on bitdepth here (take note that bitdepth is ignored with ASIO):
http://wiki.jriver.com/index.php/Audio_Bitdepth
Buffering is tricky. For playback, normally latency doesn't matter so a big buffer is good. But for live playback like this, latency is bad so you want a smaller buffer.
So to use this feature, find the smallest buffer that never hiccups and use it.
This topic brought me some significant answers for my project.
I will receive my new exasound e18 DAC this week. Like we already discused, I will use 7.1 settings for 8 channels multiamp stereo, only RR will be free. Everything is ready now, x-overs are all set like mojave has explained.
Now I want to use Room Equalization Wizard's sweep for measurements. My e18 dac has an ASIO driver "exaSound has developed custom ASIO driver for e18 DAC, that is completely independent from the Windows sound system"
It looks that I will need to have a loopback setting for playing sweeps through JRiver.
I run in windows XP pro sp3. Which settings do you see for me to set this loopback?
By the way, Is there a way to save these settings while I wait for my new dac to arrive and send a normal 2 channels stereo output to my Motu unit? What would you do?
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jacqlan111
Others may be able to give additional detail, but I think it will work like this for REW:
A) For sweeps without existing correction filters (when you do initial measurements: Use your normal REW compatible sound card (with mic inputs, etc) and do your measurements.
B) To play corrected output (after you have loaded your correction filters into MC): Select the e18 as your output device, enable DSP/Room Correction and play your music
C) To do new Post-Correction Sweep measurements (Using REW to re-measure after you have applied room correction in MC): same set-up as in A, above. Then enable loopback as described above so MC loops back ("listens to") the REW compatible sound card, applies your loaded correction filters (via dsp/room correction), and outputs your room-corrected sweep. Compare this measurement to your uncorrected sweep to see if you need additional correction tweaks.
As to saving your settings:
A) You can create a "Zone" in MC with these settings and select that zone to enable the settings.
B) or, in MC select: Playback Options/Alternate Mode Settings
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Thanks again Caleb
I think that will work fine 8). Like you see I choosed the e18 DAC. We have discussed about my project on computeraudiophile (tympatique).
I'll be back when everything's connected, we'll see then how it sounds.
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Oh, Hi
I didn't know this was you.
I went for the exaU2I/Buffalo-III option as I described today here:
http://www.computeraudiophile.com/content/Anyone-heard-new-dac-exadevices
FYI, I also received the Mytek 8x192 with 8xRCA out. Its happily connected via firewire doing everything (except dsd for now) perfectly through MC.
(Sorry to others for going off-topic)
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jacqlan111
Others may be able to give additional detail, but I think it will work like this for REW:
A) For sweeps without existing correction filters (when you do initial measurements: Use your normal REW compatible sound card (with mic inputs, etc) and do your measurements.
B) To play corrected output (after you have loaded your correction filters into MC): Select the e18 as your output device, enable DSP/Room Correction and play your music
C) To do new Post-Correction Sweep measurements (Using REW to re-measure after you have applied room correction in MC): same set-up as in A, above. Then enable loopback as described above so MC loops back ("listens to") the REW compatible sound card, applies your loaded correction filters (via dsp/room correction), and outputs your room-corrected sweep. Compare this measurement to your uncorrected sweep to see if you need additional correction tweaks.
As to saving your settings:
Initial problem: I want to use Room Equalization Wizard's sweep for measurements. My e18 dac has an ASIO driver "exaSound has developed custom ASIO driver for e18 DAC, that is completely independent from the Windows sound system"
It looks that I will need to have a loopback setting for playing sweeps through JRiver.
I run in windows XP pro sp3. Which settings do you see for me to set this loopback?MIC
It looks like the suggested procedure didn't worked. Before using JRiver, I was using MMonkey with a M.Audio card. The loopback was from anal In/Out for time ref. . Now I cant get the sweep from REW with the exasound e18 8ch DAC. Cannot map the USB output from REW. I think the input via the M.Audio card is still good. Like I explained, the exasound ASIO driver dos not show in REW's browser. I have tried all available outputs and no sound is coming through.
Need an expert here.
Thanks
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That e18 is a nice looking 8 channel DAC. 8)
Try doing the following steps. I use it for REW measurements through JRiver both with or without any correction filters.
1. To use the loopback you have to have another soundcard in the system. I use the motherboard soundcard. Nothing is actually connected to it, but the drivers still need to be installed.
2. Set the above soundcard as 7.1 (with fullrange speakers) and as the default soundcard in the Windows Control Panel. You may need to turn off port autosensing for the soundcard to be available even when nothing is connected.
3. Set JRiver's Audio output to ASIO and the e18.
4. In REW, use the JAVA audio driver and set the output to whatever is the default card and the input to what you use for your microphone.
5. Before running the REW sweeps, put JRiver in loopback mode by going to File > Open URL and enter live://loopback.
Now when you play a sweep in REW it will be sent to the default driver and be intercepted by JRiver. JRiver will play it through its DSP and ouput via ASIO to the e18. REW will record back through the input you have specified.
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... I know it was implied but nos explicitly stated in the post above this one that:
To hear the sweep jriver needs be set to loop back ie to play loop back URL .
Be sure to test your loop back settings first using mog etc to be sure that works.
With this clarification I agree with the procedure described
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... I know it was implied but nos explicitly stated in the post above this one that:
To hear the sweep jriver needs be set to loop back ie to play loop back URL .
Be sure to test your loop back settings first using mog etc to be sure that works.
With this clarification I agree with the procedure described
I just added it to the procedure. Thanks.
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This is express services.
thanks, I'll try that tomorrow and will let you know what happend
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"Insert Quote
This is express services.
thanks, I'll try that tomorrow and will let you know what happened"...
...if it's "express" we want to hear back TONIGHT, or you're slowing progress and innovation ;-)
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When MC is put in loopback mode by File > Open URL and enter live://loopback
1. Does it stay in loopback mode as long as MC is running?
2. If so do we somehow need to stop it when going to regular playback?
3. Is it safe to assume that it is not re-enabled once MC is restarted?
Thanks,
Rod
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you're slowing progress and innovation
HaHaHa! You are so right ;)
The proof is that I can't make it work. See comments in blue. Can you spot the problem?
That e18 is a nice looking 8 channel DAC. 8)
Try doing the following steps. I use it for REW measurements through JRiver both with or without any correction filters.
1. To use the loopback you have to have another soundcard in the system. I use the motherboard soundcard. Nothing is actually connected to it, but the drivers still need to be installed. The M.Audio card 24/96 is still connected and available
2. Set the above soundcard as 7.1 (with fullrange speakers) OK and as the default soundcard in the Windows Control Panel OK . You may need to turn off port autosensing for the soundcard to be available even when nothing is connected. I have a loopback out/in for time ref connected
3. Set JRiver's Audio output to ASIO OK and the e18. It's exasound ASIO ?
4. In REW, use the JAVA audio driver OK and set the output to whatever is the default card default out is the only choice and the input to what you use for your microphone. M.Audio in and the mic work as shown in the level indicator
5. Before running the REW sweeps, put JRiver in loopback mode by going to File > Open URL and enter live://loopback. OK
Now when you play a sweep in REW it will be sent to the default driver YES it is showing in M.Audio panel and be intercepted by JRiver. JRiver will play it through its DSP and output via ASIO to the e18. REW will record back through the input you have specified. This don't work, maybe the e18 ASIO driver don't see it? I have tried ASIO4all without success, the e18 DAC may not like it?
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...hi
Before trying REW be sure you have mc loopback working p something like utube or mog.
Also I wasn't sure from your post if you understand you are NOT doing any physical loopback ( it's all virtual), ie no wires involved. You can use any sound device selected to default. Play mog to it and don't connect your speakers to it. Then set jams to play the loopback URL . In mc under audio setup select asio and select the exa device as your asio device. set latency to .05 seconds. Play the URL. ( do you see sound waves on jrmc indicating something is playing?) sometimes I have to then hit stop and play again in mc.
If trouble reread this thread, as it does work.
As a final test in the veiry unlikely case its the exa device at issue ( which i highly doubt) set your onboard sound as default and use audiophile 24/94 to play out from mc in loopback. This will eliminate exa for testing to be sure youre doing loopback correctly without it.
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When MC is put in loopback mode by File > Open URL and enter live://loopback
1. Does it stay in loopback mode as long as MC is running?
2. If so do we somehow need to stop it when going to regular playback?
3. Is it safe to assume that it is not re-enabled once MC is restarted?
Thanks,
Rod
Yes it can stay in loopback. You just push start/stop as usual.
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...hi
Before trying REW be sure you have mc loopback working p something like utube or mog.
First step The loopback does not work properly, not at all I say.
When I tried to install it, (file->open URL->insert the "live://loopback") then clic OK, JRiver crashes every time.
Therefore when I tried to play Any youtube or web-radio or "REW" noting is going through JRiver.
If I open a youtube video within JRiver it's fine, but not from web pages outside JRiver.
Also I wasn't sure from your post if you understand you are NOT doing any physical loopback ( it's all virtual), ie no wires involved.
Yes I understood, the physical loopback from/to M.Audio analog card, is necessary for REW time reference for accurate measurements. This physical loopback should involves only REW's operations, it is something else. It does the same without it.
You can use any sound device selected to default. Play mog to it and don't connect your speakers to it. Then set jams to play the loopback URL . In mc under audio setup select asio and select the exa device as your asio device. set latency to .05 seconds. Play the URL. ( do you see sound waves on MC indicating something is playing?) sometimes I have to then hit stop and play again in mc.
If trouble reread this thread, as it does work. Good idea and I did it several times
As a final test in the veiry unlikely case its the exa device at issue ( which i highly doubt) set your onboard sound as default and use audiophile 24/94 to play out from mc in loopback. OK I did that, I had to set JRiver at 2CH and nothing is going through the M.Audio 24/96. But if I put the M.Audio as the default output, it is working, "without loopback" I understand. So It should be the loopback crashes that causes trouble This will eliminate exa for testing to be sure youre doing loopback correctly without it.
I don't see why JRiver crashes when loopback installation?
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Are you using a current build of mc 17?
No one else has reported your issues
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Are you using a current build of mc 17?
No one else has reported your issues
Yea! they says that I'm one of a kind?
It may be the reason here. I am running on MC 17.0.99 (registered), update channels is set to "stable"
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Please try the build from the top of this thread.
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Please try the build from the top of this thread.
I have paid attention again of all replies from the top of the thread and I am pretty sure to understand almost everything concerning the loopback option. So would you suggest that I install the 17.0.117 beta version?
It was construct on a Wasapi driver, but work also on ASIO as I see it. Is a specific (exasound asio) driver would be compatible for sure?
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Well, must be a system problem.
I have tried to reinstall the "live://loopback
" in MC 17.0.117 and with the M.Audio driver's sound card and MC crashes again when I hit OK.
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When using the ASIO drivers in REW the same device is used for playback and recording. Since the e18 has no inputs, it probably will not be available in REW for ASIO ouput which is why you can't see it. Does it show up when selecting JAVA drivers in REW? If it doesn't, you will probably have to do all your measurements with another soundcard and then put the e18 back in for playback.
Can you try the loopback using a motherboard soundcard as the default output and the M-Audio soundcard as the output card in JRiver? It might help for you to get it working in some configuration at least.
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I dared trying this feature today but only with partial luck. I get very stuttering audio no matter the buffer size (did notice smaller is better in this case). But whether a large, medium or small buffer were chosen the audio stutters.
I noticed mojave seemed to have the same problem... any ideas?
My soundcard is an Asus Xonar HDAV1.3 Deluxe and the onboard is a realtek-something using VIA-drivers (with the onboard as default windows device and the Asus for JRiver). I'm using wasapi event-style but did also try asio just to be sure. It makes no difference... using .117-version of JRiver.
Hope you can help me - let me know what further info you need and I shall provide is as soon as possible.
Best regards,
Mikkel
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When using the ASIO drivers in REW the same device is used for playback and recording. Since the e18 has no inputs, it probably will not be available in REW for ASIO ouput which is why you can't see it. Does it show up when selecting JAVA drivers in REW? No it's the only one who doesn't show.If it doesn't, you will probably have to do all your measurements with another soundcard and then put the e18 back in for playback. I need 8 ch to match MC EQ and filters and I don't have a card doing this.
Can you try the loopback using a motherboard soundcard as the default output and the M-Audio soundcard as the output card in JRiver? It might help for you to get it working in some configuration at least.
Thanks for your patience,
I wonder why nobody seams to bothers with the fact that MC crashes when I try to put the live loopback in? I tough this was crucial?
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If you are still trying to set up loobback (without introducing REW until you have LB working) and if you are still getting stuttering even at .05 sec latency and below, try disabling (unchecking ) all dsp in MC.
Dont forget to stop and restart MC between all settings changed (not pause, but stop/start)
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I dared trying this feature today but only with partial luck. I get very stuttering audio no matter the buffer size (did notice smaller is better in this case). But whether a large, medium or small buffer were chosen the audio stutters.
I noticed mojave seemed to have the same problem... any ideas?
My soundcard is an Asus Xonar HDAV1.3 Deluxe and the onboard is a realtek-something using VIA-drivers (with the onboard as default windows device and the Asus for JRiver). I'm using wasapi event-style but did also try asio just to be sure. It makes no difference... using .117-version of JRiver.
Hope you can help me - let me know what further info you need and I shall provide is as soon as possible.
Best regards,
Mikkel
Loopback is an excellent idea!
There seems to be some effect of varying the buffer size in my setup, 0.1s seems to be fine (but I do not trust it yet...). Any other buffer size longer or shorter causes violent stutter. This is my experience when I tried this first time today using ASIO. Work perfectly without Convolution in the audio path, but stutters violently when room correction filter is activated and using other buffer sizes than 0.1s.
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Loopback is an excellent idea!
There seems to be some effect of varying the buffer size in my setup, 0.1s seems to be fine (but I do not trust it yet...). Any other buffer size longer or shorter causes violent stutter. This is my experience when I tried this first time today using ASIO. Work perfectly without Convolution in the audio path, but stutters violently when room correction filter is activated and using other buffer sizes than 0.1s.
I face the same behavior with a M-Audio Profire 610 on a Q9550/2Gb RAM. Without loopback, I run Room correction and Andiolense convolution filters (up to 7 x realtime) without any problem, even with BRDs, but loopback is still stuttering.
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I face the same behavior with a M-Audio Profire 610 on a Q9550/2Gb RAM. Without loopback, I run Room correction and Andiolense convolution filters (up to 7 x realtime) without any problem, even with BRDs, but loopback is still stuttering.
I am not sure if DSP processing is the direct cause of the stutter. I ran Spotify through the loopback, and with stereo audio my audiolense filter was processed more than 50x real time. With my old computer I only occassionally heard a glitch even when processing 15 paths at 2-3x real time.
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Is there any way (logs/options) that could help to investigate the problem?
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Seems like a really great feature, but i have not been able to get it to work it, i get a "something went wrong with playback", with ASIO and WASAPI, and very choppy playback with direct sound (often sound that is speeded ut for a second, and then a pause for half a second, and so on) with direct sound. I tried with MC tunning on a S/PDIF output, and the default windows soundcard being a usb-soundcard.
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Well my initial problem with loopback was a PC XP problem, some other failures came out after (DVD burner, system restore crashes..).
I've switch everything on my more recent PC running win 7 and the M.Audio 24/96 card 2ch. Now the loopback feature work but likes others I have experienced stutters while using REW's sound sources, with or without the MC's EQ.
Will have to find a way for sweeps measurements.
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Matt
Are you able to help out here? A number of your users are having issues.
Thanks!
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Does anybody have REW playing good sweeps with output to an ASIO device?
If so, please provide a few setup tips. I want to loopback motherboard audio to my ASIO Lynx USB DAC.
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Does anybody have REW playing good sweeps with output to an ASIO device?
If so, please provide a few setup tips. I want to loopback motherboard audio to my ASIO Lynx USB DAC.
...Im doing more tests tomorrow, but thus far WASPI Event mode is working much better than ASIO.
Also, FYI REW is close to implementing the ability to do REW measurements on external and/or recorded (vs live) sweeps...
http://www.hometheatershack.com/forums/rew-forum/56977-external-sweep-rew.html#post515181
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Does anybody have REW playing good sweeps with output to an ASIO device?
If so, please provide a few setup tips. I want to loopback motherboard audio to my ASIO Lynx USB DAC.
Try doing the following steps. I use it for REW measurements through JRiver both with or without any correction filters.
1. To use the loopback you have to have another soundcard in the system. I use the motherboard soundcard. Nothing is actually connected to it, but the drivers still need to be installed. You might need to turn autosense off in the motherboard drivers. I have the Realtek HD Audio on the motherboard on my work PC and home HTPC. Disabling front jack detection in the Realtek HD Audio Manager is what actually enables the analog output from front and rear outputs.
2. Set the above soundcard as 7.1 (with fullrange speakers) and as the default soundcard in the Windows Control Panel.
3. Set JRiver's Audio output to ASIO and the Lynx.
4. In REW, use the JAVA audio driver and set the output to whatever is the default card and the input to what you use for your microphone.
5. Before running the REW sweeps, put JRiver in loopback mode by going to File > Open URL and enter live://loopback.
Now when you play a sweep in REW it will be sent to the default driver and be intercepted by JRiver. JRiver will play it through its DSP and ouput via ASIO to the Lynx. REW will record back through the input you have specified.
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3. Set JRiver's Audio output to ASIO and the Lynx.
I agree with your workflow, but get stutter in ASIO ( no matter what delay/latency i use) and good results in WASPI Event Style mode with wide flexibility in latency setting. Note: matts advice above is to use lowest latency/delay setting as will yield stable result. For me works best in waspi than asio.
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I agree with your workflow, but get stutter in ASIO ( no matter what delay/latency i use) and good results in WASPI Event Style mode with wide flexibility in latency setting. Note: matts advice above is to use lowest latency/delay setting as will yield stable result. For me works best in waspi than asio.
I found that by unchecking "use large hardware buffers" and then manually selecting large buffers in the ASIO drivers for my audio device I was able to get better stability. For my firewire audio device (Steinberg MR816x) it was necessary to set the firewire buffers to large (an obscure setting using an obscure program by Steinberg) in addition to having the ASIO buffers set larger.
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Thanks Michael, I'll try that
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I wonder why nobody seams to bothers with the fact that MC crashes when I try to put the live loopback in? I tough this was crucial?
> It ( loopback ) works for the others commenting within this thread because they are obviously using one of the operating systems that supports WASAPI ( XP doesn't ) .
> Please NOTE: WASAPI is only available from within VISTA & WIN7 or newer . SEE ; WASAPI Notes ( from this very web-site ) . (http://wiki.jriver.com/index.php/WASAPI)
> If you are keen to use this "Loop-back" feature ( from within MC ) you'll need to use a supported "Operating System" ( like WIN7 ) to get the necessary WASAPI support .
:)
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> It ( loopback ) works for the others commenting within this thread because they are obviously using one of the operating systems that supports WASAPI ( XP doesn't ) .
> Please NOTE: WASAPI is only available from within VISTA & WIN7 or newer . SEE ; WASAPI Notes ( from this very web-site ) . (http://wiki.jriver.com/index.php/WASAPI)
> If you are keen to use this "Loop-back" feature ( from within MC ) you'll need to use a supported "Operating System" ( like WIN7 ) to get the necessary WASAPI support .
:)
thanks for your concern, Matt had this reply on the subject for me in an other topic
"loopback help needed win XP sp3"
I don't know how to refer to this, but that was the reply
In a coming build:
Fixed: Starting loopback playback (live://loopback) would crash on XP instead of fail gracefully (loopback requires Vista or newer and WASAPI).
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I am following the convolution thread and this one for long time but this is my first question. Before asking question I believe you need to know my hardware:
1. HTPC> JRiver 17> Asus Xonar ST + H6 board> 6 analog out ( front channel through passive switch)> 6 analog in Denon AVR 5700> 5.1 speaker setup with 2 sub. This way I listen multichannel audio
2. HTPC > JRiver 17>passive switch > Eastern Electric Minimax DAC plus> 2ch
My intention is with Audiolense and Convolution engine in JRiver to create 2 sets of filters :
a) One for 5.1 set up ( including 2 sub) through Asus Xonar ST + H6. My understanding is that should not be a problem, of course I will have to learn a lot
b)In the moment I am not able to incorporate my 2 sub with my DAC for 2.1 channel stereo, or at least I don't know how. My DAC is able to use WASAPI mode only, does not work with Asio driver. My question is , is it possible with the help of loop function in JRiver to incorporate the 2 sub with my DAC and all the measurement with Audiolense XO, separately for 5.1 set up through Asus Xonar ST +H6 board and 2.1 setup with 2 sub and EE Minimax DAC plus ?. I am so eager to get the YES answer. Sorry for my English
Thank you
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My intention is with Audiolense and Convolution engine in JRiver to create 2 sets of filters :
I think you should start by creating two Zones: one for the 5.1 setup and one for the 2 channel setup. This will make it easier to manage the settings and convolution for each system.
One for 5.1 set up ( including 2 sub) through Asus Xonar ST + H6. My understanding is that should not be a problem, of course I will have to learn a lot
Since you have 8 channels with the ST & H6 (I have the same) you can set Output Mode in JRiver's DSP Studio to 7.1. Then copy the subwoofer channel to channel 5 (Rear Left) using the Parametric Equalizer DSP with Mix Channels. You can then connect one sub to channel 4 (subwoofer) and the other sub to channel 5. Channels 7 and 8 will be your surrounds. This will give you a little more flexibility with Audiolense and convolution. Does this make sense?
In the moment I am not able to incorporate my 2 sub with my DAC for 2.1 channel stereo, or at least I don't know how. My DAC is able to use WASAPI mode only, does not work with Asio driver. My question is , is it possible with the help of loop function in JRiver to incorporate the 2 sub with my DAC and all the measurement with Audiolense XO, separately for 5.1 set up through Asus Xonar ST +H6 board and 2.1 setup with 2 sub and EE Minimax DAC plus
It is not possible to use the sub for each system. The loopback feature won't help with this. I don't know what sub you have, but it may be possible to change the cables and settings to get it to work with the DAC system, but it might not be worth the effort each time.
Do both the Asus Xonar ST and Minimax go through the passive switch and then to the Denon AVR5700?
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--. Do you know you only need loopback for non- media center sources? For mc sources skip loopback and just use convolution
--. Assuming you need/want loopback, do I correctly assume your source material will be 2-ch? If so, your decault sound playback device under audio control panel in windows should be set at 2-ch (in windows control panel). In jriver mc set playback to your m-ch sound device as waspi (or asio). Now in dsp studio set eq to on and configure it to duplicate channels to split left / right signals to l/r main and to l/r sub
I'm on iPhone not at pc so this is a general description
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[It is not possible to use the sub for each system. The loopback feature won't help with this. I don't know what sub you have, but it may be possible to change the cables and settings to get it to work with the DAC system, but it might not be worth the effort each time.]
I have two B@W ASW 300 active sub connected in a serial to sub port od Denon. All cables are under the floor ? so not much flexibility here./quote]
Yes , both sound card and DAC go through passive switch . My idea was to use only analog connection between sound card and Denon to be sure that no processing whatsover is done in Denon, both for 2 ch and 5.1 set up. Denon is always set to 6 channel in for both 2ch and 5.1 ch setup. To be onest this loop back feature is not very clear to me, its just my shot if somhow can be uset to incorporate my subs in 2 ch stereo. Exually the problem start whe I incorporated the DAC in my system?:-[
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Is there a way to enable loopback in theaterview or using MCC commands to enter a url ? i tried entering it as a url in theaterviewbut it comes back with an error?
TIA,
AL
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Is there a way to enable loopback in theaterview or using MCC commands to enter a url ? i tried entering it as a url in theaterviewbut it comes back with an error?
I think this shold work:
Play from Standard View, then add the file to a playlist named something like 'Loopback'.
Then play the playlist from Theater View.
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I think this shold work:
Play from Standard View, then add the file to a playlist named something like 'Loopback'.
Then play the playlist from Theater View.
Awesome! That worked.
Thank you.
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Hi,
I am using Zune to stream music to JRiver in loopback mode. JRiver is using my Lynx L22 sound card with ASIO driver. With digital loopback on my Lynx card, I can route JRiver's output to my amp/speakers.
With Convolution in the path for DRC, I am getting dropouts, clicks, and pops. I have turned off use large buffers and set the buffer to 0.02 secs. No change. Without Convolution in the path, sound plays fine.
For fun, I have the old convolverVST plugin in JRiver, so I checked off Convolution and tried convolverVST. So far, no dropouts and crystal clear sound...
I don't get it. Anyone have any thoughts?
Cheers,
Mitch
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Loopback doesn't support DSP effects that add latency in v17.
However, v18 (early versions coming in a few weeks) will address this.
Here's the history entry:
NEW: Loopback mode adapts the target latency automatically so that it works with effects that create delays like resampling, convolution, Dolby Digital output, etc.
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Loopback doesn't support DSP effects that add latency in v17.
However, v18 (early versions coming in a few weeks) will address this.
Here's the history entry:
NEW: Loopback mode adapts the target latency automatically so that it works with effects that create delays like resampling, convolution, Dolby Digital output, etc.
Cool Matt, thanks. If you are looking for a Beta tester... :-)
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Yet another great feature. Thanks Matt.
David
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However, v18 (early versions coming in a few weeks) will address this.
Here's the history entry:
NEW: Loopback mode adapts the target latency automatically so that it works with effects that create delays like resampling, convolution, Dolby Digital output, etc.
I'm using v18, and just started using convolution. I'm using Output Format, Volume Leveling, Convolution, PEQ and Room Correction for DSP. Getting bad clicks and pops with Loopback, has this been addressed? Changing the buffer has no effect, though I'm getting stuttering audio with a low buffer and no loopback (.2 seconds right now for clean audio with no loopback). MC is set to resample everything to 48khz, my interface is set to 48khz, mic preamp is set to 48khz, and the filters are built for 48khz.
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I'm using v18, and just started using convolution. I'm using Output Format, Volume Leveling, Convolution, PEQ and Room Correction for DSP. Getting bad clicks and pops with Loopback, has this been addressed?
Next build has a new algorithm for latency management during loopback playback that should work much better with "lumpy" DSPs like Convolution.
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Next build has a new algorithm for latency management during loopback playback that should work much better with "lumpy" DSPs like Convolution.
Cool!
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Next build has a new algorithm for latency management during loopback playback that should work much better with "lumpy" DSPs like Convolution.
Matt, thanks so much for this. Works perfect! I use ASIO line in at 24/96 with Audiolense and REW to pass sine wave sweeps through JRiver's audio engine. I can toggle Convolution off/on to get before/after measurements to compare. There are no clicks, pops, or stuttering at all. Thanks again!
Cheers, Mitch
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Matt, thanks so much for this. Works perfect! I use ASIO line in at 24/96 with Audiolense and REW to pass sine wave sweeps through JRiver's audio engine. I can toggle Convolution off/on to get before/after measurements to compare. There are no clicks, pops, or stuttering at all. Thanks again!
Cheers, Mitch
Cool. Gotta try this.
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I've no pops or clicks, but I still have stuttering. How the hardware is influencing all this?
I've a M-Audio ProFire 640, a Q9550 and 2 GBytes of memory.
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Matt, thanks so much for this. Works perfect! I use ASIO line in at 24/96 with Audiolense and REW to pass sine wave sweeps through JRiver's audio engine. I can toggle Convolution off/on to get before/after measurements to compare. There are no clicks, pops, or stuttering at all. Thanks again!
Cheers, Mitch
How does that work? I thought the line in was for running external sources. Are you doing an actual physical loopback?
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I'm still not having any luck with the loopback and convolution. It works to where I can do measurements with the loopback + convolution but I have to redo them quite a bit because of dropouts during the sweep.
I just realized a bit ago that its "WASAPI" loopback (duh!), and I've been using ASIO. The loopback works just fine without convolution in ASIO, but turn convolution on and it falls apart. I can't get the buffer small enough to play clean from my pc source and work with the loopback as well. 1.6 ms is as low as I can go and get clean playback from my pc source.
So I switched to WASAPI. With the smallest buffer (.05) I can get clean playback, with the occasional click and pop. The abnormal stuff seems to happen when I'm working on other things on the computer. When I just listen and not do anything else I get clean playback. Loopback works for maybe 20-30 seconds before going into static.
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How does that work? I thought the line in was for running external sources. Are you doing an actual physical loopback?
No, this is a software loopback which is done at WASAPI level ( so it only works on Vista, 7 and 8 ).
You need two soundcards in your configuration, the motherboard's soundcard is usually used for that.
The other applications broadcast the sounds to it, and JRiver routes the WASAPI stream to its own DSP.
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How does that work? I thought the line in was for running external sources. Are you doing an actual physical loopback?
In my case, I am using a Lynx Hilo which has an internal 32x32 digital mixer that allows flexible routing. I made a diagram so I can remember the channel numbers and application offsets.
REW is also used in place of Audiolense on the diagram with the same routing. Both applications run at 24/96 ASIO as the Lynx driver is multi-client. The bit-perfect indicator is on in JRiver when I run the sweeps, without the filter in Convolution.
I use Audiolense to measure, design, and generate the 64-bit FIR correction filter. Audiolense/REW sweeps are passed through JRiver's audio engine to be included in the measurement as the complete signal path. The FIR correction filter is hosted in JRiver's 64-bit Convolution engine. Before and after filter measures are achieved by toggling Convolution off/on.
This is one approach.
Cheers, Mitch
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In my case, I am using a Lynx Hilo which has an internal 32x32 digital mixer that allows flexible routing. I made a diagram so I can remember the channel numbers and application offsets.
REW is also used in place of Audiolense on the diagram with the same routing. Both applications run at 24/96 ASIO as the Lynx driver is multi-client. The bit-perfect indicator is on in JRiver when I run the sweeps, without the filter in Convolution.
I use Audiolense to measure, design, and generate the 64-bit FIR correction filter. Audiolense/REW sweeps are passed through JRiver's audio engine to be included in the measurement as the complete signal path. The FIR correction filter is hosted in JRiver's 64-bit Convolution engine. Before and after filter measures are achieved by toggling Convolution off/on.
This is one approach.
Cheers, Mitch
Hi Mitch,
So, from what I understand, you are using the rewire facilities of your Lynx sound card. I don't understand in your schema how you use the WASAPI Loopback.
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No, this is a software loopback which is done at WASAPI level ( so it only works on Vista, 7 and 8 ).
You need two soundcards in your configuration, the motherboard's soundcard is usually used for that.
The other applications broadcast the sounds to it, and JRiver routes the WASAPI stream to its own DSP.
I understand what the loopback is, and have been using it for some time. Please reread my other post. MitchCo is not using the loopback. He's using a new feature called ASIO Line In.
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Hi.
Just getting my head round using this feature. So I was wandering if it has been taken any further. For instance, is there a short-cut available, or could you set a zone up linked to loopback?