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post #271 of 326 Old 05-04-2006, 12:39 PM
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You betcha, just not too sure which arguments those are.

I hate Sony. I love BD. I hate Sony...


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post #272 of 326 Old 05-04-2006, 01:19 PM
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Ok, so I may have the source of my evil. There seems to still be remnants of my Delta 410 drivers still around.

I guess I need to clean my registry...

I hate Sony. I love BD. I hate Sony...


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post #273 of 326 Old 05-04-2006, 01:28 PM
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Can anyone help me out with Rec_Imp configuration for use with a mono mic?



Output: 0 channel is left, 1 is right
Input: my mic is on the left

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post #274 of 326 Old 05-04-2006, 02:11 PM
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Ok, I've wasted at least 10 hours on this due to lack of documentation. I promise to document a step by step method for this so that poeple can actually use it if you guys can help me through this.

Thanks,

Wes

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post #275 of 326 Old 05-04-2006, 02:24 PM - Thread Starter
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Pick up http://www.duffroomcorrection.com/im...Rec_impBin.zip from http://www.duffroomcorrection.com/wi...Measuring_Tool

and use rec_impds or the debug version rec_impdsd

That will use the windows native drivers, rather than the asio ones.
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post #276 of 326 Old 05-04-2006, 02:59 PM
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Ok, that helps a bit, it gives me a different picture on the devices. That being said, I still have no idea which is which in the rec_imp syntax

rec_imp.exe LeftSpeakerImpulseResponse.pcm 44100 20 21000 20 0:1 0:1
---------------------------------------------------------------^ ^ ^ ^
----------------------------------------------------------------1 2 3 4
1:?
2:?
3:?
4:?

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post #277 of 326 Old 05-04-2006, 03:09 PM
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Tried rec-imp -help and found my answer. Sometimes I need a slap....

I hate Sony. I love BD. I hate Sony...


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post #278 of 326 Old 05-04-2006, 03:18 PM - Thread Starter
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I don't understand precisely what problem you are encountering. Are you mixing up your outs with your ins?

As you will have discovered:

The rec_imp syntax is channel:device, indexed from 0

rec_imp -l will list your devices and give you a device number to use


rec_imp impulseFile rate hzStart hzEnd duration sweepOut sweepIn
where options,
impulseFile = file to save impulse response to,
rate = the sample rate,
hzStart = freq to start sweep,
hzEnd = freq to end sweep,
duration = length of sweep in seconds,
sweepOut = output channel and device to play the sweep,
syntax: channel:device, device=0 automatically picks first card,
sweepIn = input channel and device to record the sweep,
syntax: channel:device, device=0 automatically picks first card,
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post #279 of 326 Old 05-04-2006, 03:20 PM - Thread Starter
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Quote:
Originally Posted by Mooneyass View Post

Tried rec-imp -help and found my answer.

or even just rec_imp!
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post #280 of 326 Old 05-04-2006, 03:26 PM
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What do you guys get for a normal level for your sweep recordings? Miune seem quite low even though I'm at quite high volumes and mic level.

I'm getting alot of pooping during playback with the convolver on (JRMC 11.1)

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post #281 of 326 Old 05-04-2006, 04:05 PM
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rec_impds -l gives me 11 devices each with multiple I/O......

It all seems to work fine in WMP10 though.

I hate Sony. I love BD. I hate Sony...


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post #282 of 326 Old 05-31-2006, 03:44 PM - Thread Starter
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added mono convolverVST -- VST plug-in version: feedback welcome

put up property page when no filter loaded, so that the user has an opportunity to load filter first time through

internal interface tidying
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post #283 of 326 Old 06-21-2006, 01:32 PM - Thread Starter
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Added GUI for convolverVST. Thanks to Vera Kinter for some of the controls elements.

Fixed convolverCMD when number of input and output channels is different. convolverCMD now outputs in float format. (Thanks to Peter Fischer.)

Released Intel C++ 9.1 (which may be faster on P4 machines) and VC++ 2005 versions.


A panned binaural walkthrough by Peter Fischer is at http://convolver.sourceforge.net/binaural.html

Feedback welcome
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post #284 of 326 Old 06-23-2006, 04:26 PM - Thread Starter
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Version 4.2 now provides a multi-channel convolverVST. Let me know how it goes with a variety of hosts.

Documentation at http://convolver.sf.net

Download from
http://sourceforge.net/project/showf...ease_id=427201
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post #285 of 326 Old 07-10-2006, 02:45 PM - Thread Starter
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Use FFTW 3.1.2 (which is faster and should run on more non-Pentium 4 platforms)

Disable "Lost filter" convolverVST error message, to give the user a chance to select a filter
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post #286 of 326 Old 07-19-2006, 03:01 PM - Thread Starter
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Added convolverPipe, which filters the standard input to the standard output (as opposed to convolverCMD, which filters files)

convolverWMP registration
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post #287 of 326 Old 01-17-2007, 10:34 PM
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Hi,

I just started using the convolver in conjunction with a correction filter generated by Acourate. Everything seems to work so far except for a strange interaction with the playback controls Jriver Media Center. I use the DirectX filter loaded into the DirectX-Host that comes with JRMC.

Whenever I switch a track the following happens.

The playback switches to a different part of the song a few seconds shifted and then with some delay the new song starts.

When I stop and then play a different song it first plays the remainder of the previous song before starting with the new one.

I appears like the convolver buffers about 2s worth of data and first drains this old data before starting with a new track. On pause this works well since it provides a gapless experience.

Is there a way to fix this behavior. I believe if the stream gets switched it would be better to flush the buffer. However, I don't know what a directX filter loaded inside the JERMC DirectX-host actually sees when a user switched a track, hits pause, or stops a track.

Ideally the buffer would be kept for a stop but flushed for a stop or switching a track.

Cheers

Thomas

P.S.: I will probably ask a few more questions on how to get seperate correction filters for 44.1, 48, 96 Khz when I have this all fully set up.
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post #288 of 326 Old 01-17-2007, 11:12 PM
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A certain amount of delay is unavoidable with FIR filters - and comes in addition to the buffer in jrmediacenter. But you should probably check that the buffer in jrmediacenter is set to minimum. Just in case.
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post #289 of 326 Old 01-18-2007, 02:54 PM - Thread Starter
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Sright. You could also try increasing the number of partitions.
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post #290 of 326 Old 01-18-2007, 10:06 PM
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I am not worried about the delay. I am worried about convolver not flushing the buffer when the stream is being stopped.

The first 2 seconds after playing a new track will be the samples that got stuck in the buffer from the previous track.

Can this be fixed?

Cheers

Thomas
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post #291 of 326 Old 01-19-2007, 12:00 AM - Thread Starter
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I'll have a look over the next couple of weeks.
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post #292 of 326 Old 01-20-2007, 08:09 AM
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Dear jrpavel,

did you notice that partitioned convolution patent as been revoked?
Are you planning to implement it on you plug-in?

Kind Regards,
Michele

see here: http://www.ludd.luth.se/~torger/brutefir.html#news

Bye
Michele
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post #293 of 326 Old 01-20-2007, 10:33 AM - Thread Starter
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Thanks. Not an immediate priority -- it is significantly more complex to implement. When I get some time, I will be working on correcting a few usabilty bugs/features.
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post #294 of 326 Old 03-16-2007, 09:35 AM
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hi, first of all i would like to thank jrpavel for all the work put into this program. it looks great.

i have read through the entire thread here and wanted to know if there are any example configuration files for some more complex setups.

i understand how to create the DRC files, and crossover filters after reading on the duffroomcorrection wiki.

i see how to do DRC or crossover in the config, but don't understand how to do both at once.

i think i understand the time delay options in the config file

I read here: http://convolver.sourceforge.net/config.html, but i think i need to see an example of a combination setup to understand how it works (DRC + crossover)

i am interested in taking a stereo signal and applying DRC as well as a 4-way xover with time delay otions (8 channel output)

it seems like this is possible, but i can't find any examples.

also, is there a list of soundcards that are known to work, or is any 7.1 sound card going to be ok??

thanks
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post #295 of 326 Old 03-16-2007, 11:52 AM
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check out the XMeridian card thread for an ideal solution to that question....


Jim White
St. Petersburg, FL
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post #296 of 326 Old 03-16-2007, 12:24 PM - Thread Starter
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Quote:
Originally Posted by jmelan View Post

i am interested in taking a stereo signal and applying DRC as well as a 4-way xover with time delay otions (8 channel output)

it seems like this is possible, but i can't find any examples.

also, is there a list of soundcards that are known to work, or is any 7.1 sound card going to be ok??

I don't have more detailed guidance, but suspect that you will need to combine the DRC filters with the frequency filters (for xover). I would have thought that Audition or some such would do the trick, or you could try console to set up a sequential (input -> DRC filters -> frequency splitting filters -> output) network.

Your 7.1 card should work if the drivers are OK. Let us know how you get on,

Best, J
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post #297 of 326 Old 03-16-2007, 09:16 PM
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Quote:
Originally Posted by jimwhite View Post

check out the XMeridian card thread for an ideal solution to that question....


thanks, that soundcard looks to be perfect - 5Vrms 8 channel analog outputs with user replaceable OPAMPS if i am reading the specs right!

at $200 its not cheap, but it looks to be worth it.


Quote:
Originally Posted by jrpavel View Post

... I would have thought that Audition or some such would do the trick, or you could try console to set up a sequential (input -> DRC filters -> frequency splitting filters -> output) network.

Your 7.1 card should work if the drivers are OK. Let us know how you get on,

Best, J

thanks for the ideas, i have no idea on how to combine filters with audition, but I will try to figure it out.

i was trying to avoid buying any more software, but i guess console would be good to have anyway.

also, i was looking at convolverPIPE. i haven't been able to find much info on it outside of your website. it looks like it can be used to filter audio from an external source through the default input and back out the default output?
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post #298 of 326 Old 03-17-2007, 07:07 AM
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If you run filter and xo in sequences, it will only increase the load and the latency of the playback graph.

The best solution is to convolve the filters with the XO's. You need a tool to do the following steps:

For each channel:
1) convolve filter with xo. Now your filters are twice as long.
2) cut filter down to preferred size. Preferred size should be power of 2. If the filter has centered peak, the peak should also be centered after the cutting. If it is a minimum phase filter (peak at the far left, the should be cut only on the right side
3) Apply a blackman window to the filter. If it's a filter with centered peak, an ordinary black man window is fine. If it is a minimum phase filter, only a right-sided half blackman should be applied.
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post #299 of 326 Old 03-17-2007, 01:37 PM
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thanks for the advice!! i really do not have much experience with any of these programs.

would i be able to use audition to try this?

as of now, i used room eq wizard to capture the left and right impulse response with a ECM8000 and m-audio mobile pre. i may use the plugins for audition and compare the results.

i plan on setting up a more basic stereo filter to start and test it by comparing altered music files.

i am hoping with a little trial and error i can begin to understand how to do what you just said
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post #300 of 326 Old 04-07-2007, 09:55 PM
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Hi,
First off, jrpavel thanks for all of your work on the Convolver Plug-in. I have only played around with it so far, but now I am ready to get serious.
I have been working with Adobe Audition to find the best method for converting a regular stereo CD into a synthesized binaural CD, and have begun to get decent results. Now I would like to use the Convolver plug-in for Windows Media Player to do the same thing.
I have read all of the documentation I can find, but there is still one thing I cannot figure out. I want to take the original signal and convolve it through a wav file, then crossfeed it and recombine it with the original signal.
Yr = Xr + Xl * Hl
Yl = Xl + Xr * Hr
I also need to adust the volume of the original signal and the convolved crossfeed individually when they are recombined to get the right balance and avoid clipping.
The part I don't get is how to pass the original signal through without convolving it through a file.
If I write this;
44100 2 2 0
0 0
0 0
C:\\impulses\\Binaural.wav
0
0.0
1.0
C:\\impulses\\Binaural.wav
1
1.0
0.0
then I believe I will get the crossfeed that I want, but then how do I add it back in to the unfiltered signal which I also want to attenuate approximately 3dB
???? Original ?????
0
0.0
0.5
???? Original ?????
1
1.0
1.5
Any help you can give would be much appreciated.
Thanks,
Adrian
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