Oh and yes, you need to write a config for multichannel to work in convolver VST.
See the config file page on convolver homepage:
http://convolver.sourceforge.net/config.html
I also attached an example of a config file which sets up for 2 inputs and 8 outputs. This would allow a 2 channel, 4 way loudspeaker for example.
See the config file page on convolver homepage:
http://convolver.sourceforge.net/config.html
I also attached an example of a config file which sets up for 2 inputs and 8 outputs. This would allow a 2 channel, 4 way loudspeaker for example.
I think this is my config for convolver plugin
44100 6 6 0
0 0 0 0 0 0
0 0 0 0 0 0
C:\Programme\Convolver\FrontL.dbl
0
0.0
0.0
C:\Programme\Convolver\FrontR.dbl
0
1.0
1.0
C:\Programme\Convolver\Center.dbl
0
2.0
2.0
C:\Programme\Convolver\Sub.dbl
0
3.0
3.0
C:\Programme\Convolver\RearL.dbl
0
4.0
4.0
C:\Programme\Convolver\RearR.dbl
0
5.0
5.0
If i understand, this is 6ch input to 6ch output
mono filter
same input# as output# with no scaling, no delay
44100 6 6 0
0 0 0 0 0 0
0 0 0 0 0 0
C:\Programme\Convolver\FrontL.dbl
0
0.0
0.0
C:\Programme\Convolver\FrontR.dbl
0
1.0
1.0
C:\Programme\Convolver\Center.dbl
0
2.0
2.0
C:\Programme\Convolver\Sub.dbl
0
3.0
3.0
C:\Programme\Convolver\RearL.dbl
0
4.0
4.0
C:\Programme\Convolver\RearR.dbl
0
5.0
5.0
If i understand, this is 6ch input to 6ch output
mono filter
same input# as output# with no scaling, no delay
Your txt config looks fine to me. Just make sure the paths are correct and only use convolver VST within console.
If your running Vista then no need to worry about sample rate. Its a got a very nice SRC algorithm included.
If you go to control panel->classic view->sound then highlight your emu playback device. Click configure and set for your correct multichannel configuration including LFE management options.
Afterwards go to properties then advanced and set the default sample rate you want to work at eg. 24bit 192Khz etc. and from there the audio will either be bit perfect(should it match the source) or be upsampled/downsampled depending on the source rate.
You only need one set at filters at this sample rate you choose and everything will work automatically. This is the beauty of the Emu drivers and Vista working very well together.
If your running Vista then no need to worry about sample rate. Its a got a very nice SRC algorithm included.
If you go to control panel->classic view->sound then highlight your emu playback device. Click configure and set for your correct multichannel configuration including LFE management options.
Afterwards go to properties then advanced and set the default sample rate you want to work at eg. 24bit 192Khz etc. and from there the audio will either be bit perfect(should it match the source) or be upsampled/downsampled depending on the source rate.
You only need one set at filters at this sample rate you choose and everything will work automatically. This is the beauty of the Emu drivers and Vista working very well together.
Re: foobar volume to WVR
Unfortunately not.
WVR watches the event for master volume changes. As you probably already realise, the master volume does nothing with RME cards since the windows mixer is bypassed with RME's own DSP mixer and WVR works nicely to enable a large scope of control methods to be linked to this.
[email]JAMES@BOYD.ORG[/email] said:
Unfortunately not.
WVR watches the event for master volume changes. As you probably already realise, the master volume does nothing with RME cards since the windows mixer is bypassed with RME's own DSP mixer and WVR works nicely to enable a large scope of control methods to be linked to this.
No, i use XP.
I have build a second range of 48Khz filters and append a
second path for this in the config.
A little questions :
Is there a freeware, that do the same as Console ?
Or can i even use the Patchmixer insert ?
Console is very flexible, but i need only one vst plugin.
I have bought console, but i can say, that this convolver is not
the last i build, there are a bunch of interested people who wait
for my list
Other question :
A friend use Pristine Space, so i can try.
Is this plugin better than the convolver vst plugin ?
I have build a second range of 48Khz filters and append a
second path for this in the config.
A little questions :
Is there a freeware, that do the same as Console ?
Or can i even use the Patchmixer insert ?
Console is very flexible, but i need only one vst plugin.
I have bought console, but i can say, that this convolver is not
the last i build, there are a bunch of interested people who wait
for my list
Other question :
A friend use Pristine Space, so i can try.
Is this plugin better than the convolver vst plugin ?
yes, I'll make a quick guide when my fireface 400 has come back from hospital in germany...
Its quite simple if you are used to using a DAW.
meanwhile...
1. in Reaper - set audio device to asio Rearoute
(Options - Preferences - Audio - Device )
2. Then open foobar. (the order is important). Set output to - asio virtual devices - rearoute asio
(file - preferences - playback - output - asio virtual devices - rearoute asio)
3. in reaper you then add a receive track and as many output tracks as you want. the manual is very good at explaining how to do this. You could use the routing/grouping matrix - or you can setup sends and receives from the buttons on each track.
its actually all quite intuitive if you have a play...
Its quite simple if you are used to using a DAW.
meanwhile...
1. in Reaper - set audio device to asio Rearoute
(Options - Preferences - Audio - Device )
2. Then open foobar. (the order is important). Set output to - asio virtual devices - rearoute asio
(file - preferences - playback - output - asio virtual devices - rearoute asio)
3. in reaper you then add a receive track and as many output tracks as you want. the manual is very good at explaining how to do this. You could use the routing/grouping matrix - or you can setup sends and receives from the buttons on each track.
its actually all quite intuitive if you have a play...
Maybe I should make a new thread on this to get a better answer. I was wondering if there is a generally accepted way to handle a mono subwoofer?
For instance when I use Waves Q-Clone to probe M-Audio's Delta series bass managment software, the subwoofer comes back at around -6dB. Wouldn't that mean the software is turning the bass down 12dB? Because you get +6dB when you combine the 2 signals right?
Anyway I am just a little confused because I set up my bass managment software to where Q-Clone was showing the bass flat on the mono subwoofer feed. But it seems like the bass is really hyped up and too loud at that level - about 3dB to my ears. And of course this IS 6dB louder than what m-audio's official bassmanagment is using.
I expect some hype in the bass area with the way most CDs are mastered. That is one thing that is throwing me off - badly mastered CDs with hyped up bass.
Ideally I would just use 2 matched subwoofers. But eh this is the only one I have and it is just sort of bugging me that I am unsure how to handle the levels on the mono sub - I am mixing music so it must be pretty exact.
For instance when I use Waves Q-Clone to probe M-Audio's Delta series bass managment software, the subwoofer comes back at around -6dB. Wouldn't that mean the software is turning the bass down 12dB? Because you get +6dB when you combine the 2 signals right?
Anyway I am just a little confused because I set up my bass managment software to where Q-Clone was showing the bass flat on the mono subwoofer feed. But it seems like the bass is really hyped up and too loud at that level - about 3dB to my ears. And of course this IS 6dB louder than what m-audio's official bassmanagment is using.
I expect some hype in the bass area with the way most CDs are mastered. That is one thing that is throwing me off - badly mastered CDs with hyped up bass.
Ideally I would just use 2 matched subwoofers. But eh this is the only one I have and it is just sort of bugging me that I am unsure how to handle the levels on the mono sub - I am mixing music so it must be pretty exact.
Help !
I try to save a measurement with acourate Logsweep recorder,
but all i get is a flat line in the amplitude window.
I think, the error is the routing over the e-mu mixer.
I want to record the Mic A, so i have insert a SEND Command
to Asio in 1/2 Host.
For output i´ve made a strip with Asio out 1/2.
LSR inout seems correct, because the peak meter is green and
yellow. How i manage the record on harddisc ?
I try to save a measurement with acourate Logsweep recorder,
but all i get is a flat line in the amplitude window.
I think, the error is the routing over the e-mu mixer.
I want to record the Mic A, so i have insert a SEND Command
to Asio in 1/2 Host.
For output i´ve made a strip with Asio out 1/2.
LSR inout seems correct, because the peak meter is green and
yellow. How i manage the record on harddisc ?
Sonopanic:
Try if you get managed to record a signal with the help of
http://www.emu.com/support/files/download3.asp?Centric=759&Platform=1&File=379
Uli
Try if you get managed to record a signal with the help of
http://www.emu.com/support/files/download3.asp?Centric=759&Platform=1&File=379
Uli
acourate Logsweep recorder
Hey Sonopanic,
I'm not using Acourate yet. But, I had a similar issue with Audacity.
What fixed my issue was : When I created the strip for my Mic A, I too initially inserted a send to ASIO. I was not able to record - I was just getting a flat line. Then I changed the send to make it send to WAVE instead of ASIO and that fixed it.
I don't know if this helps with Acourate or not. But, it's easy enough to try.
Good Luck,
Steve
Hey Sonopanic,
I'm not using Acourate yet. But, I had a similar issue with Audacity.
What fixed my issue was : When I created the strip for my Mic A, I too initially inserted a send to ASIO. I was not able to record - I was just getting a flat line. Then I changed the send to make it send to WAVE instead of ASIO and that fixed it.
I don't know if this helps with Acourate or not. But, it's easy enough to try.
Good Luck,
Steve
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