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Convolver DSP plug-in for Windows Media Player 10 - Page 6

post #151 of 326
Thanks jrpavel - I will let you know how it goes - i'll post the .bat when I have it working.

I followed the link stereo impulse response. If I want to use that method, I presume I cannot use rec_imp - am I correct ?
post #152 of 326
Thread Starter 
That may well be right. I think that rec_imp produces only mono recordings, but perhaps Ed Wildgoose or Denis Sbragion may be able to say more.
post #153 of 326
Thread Starter 
Michele,

Yes, convolver (or its DirectShow filter forms) can do this. Eg, I have heard that they already work with TT2.2
post #154 of 326
Quote:
Originally Posted by jrpavel View Post

Michele,

Yes, convolver (or its DirectShow filter forms) can do this. Eg, I have heard that they already work with TT2.2

The "problem" is, I think, the convolver plug-in can only be used if analog audio is picked as output, while if someone would like to output via s/pdif it is not possible.

Just a small suggestion: why don't you get in touch with Milan Cukta in order to add your convolver filter to ffdshow audio?
post #155 of 326
Thread Starter 
Michele, I see. I'll have a look at how SPD/IF is handled.

For the moment my priority is to get Convolver to a stable state and perhaps improve the performance, particularly for matrix convolutions. I can then have a look at using the routines in other wrappers such as ffdshow.
post #156 of 326
Quote:
Originally Posted by jrpavel View Post

Michele, I see. I'll have a look at how SPD/IF is handled.

For the moment my priority is to get Convolver to a stable state and perhaps improve the performance, particularly for matrix convolutions. I can then have a look at using the routines in other wrappers such as ffdshow.

Great Pavel!
Obviously it was just a suggestion!

S/Pdif is somehow needed, if you want to convolve movies sound and getting a digital output, because a 5.1 (6.1) signal can only be routed through a single s/pdif output only if it is compressed (AC-3 or DTS).
Also a digital output is in someway mandatory, IMHO, for HTPC due to Reclock, which does not work with analog output.
post #157 of 326
Reclock works fine with analogue outputs - in fact that's what it was originally designed for.

I think trying to get a digital output for convolved 5.1 is a very bad idea - unless you have a soundcard and processor with 3 SPDIF outs and INPUTS, and I've never seen that.

To make digital work, you need to decode it, convolve it, then reencode it (like AC3filter can).

I think the reencoding with a lossy compressor will negate any benefits of convolving. To me the only way to use a 5.1 channel convolver is with the analogue outputs.

For stereo I use SPDIF, but that doesn't need reencoding.
post #158 of 326
Thread Starter 
For those of you interested in IR measurement, consider http://www.avsforum.com/avs-vb/showthread.php?t=529224

Do the new rec_imp builds work for people, or does it all still crash?
post #159 of 326
jrpavel
I have yet to fully test rec_impDS but this build certainly runs - the previous build crashed on my MCE2k m/c.

You mentioned the other Impulse Response prog - Room EQ Wizard - as I understood it, RoomEQWizard was for parametric adjustments of certain AV equipment - do you know if it can be used for generating an IR instead of rec_imp ? If so, any idea how? Would it generate file types which can be fed directly into DRC for example ? Why would it be better? or is it just an alternative?

Thanks,


EDIT*** - I see that the latest version of roomEQwizard now allows the saving of a wav IR file - I'll start experimenting as an alternative to rec_imp.
post #160 of 326
Quote:
Originally Posted by jrpavel View Post

For those of you interested in IR measurement, consider http://www.avsforum.com/avs-vb/showthread.php?t=529224

Do the new rec_imp builds work for people, or does it all still crash?

Works fine for me.
post #161 of 326
So, is rec_imp working with ASIO such as my my M-Audio FW410?
post #162 of 326
Quote:
Originally Posted by jrpavel View Post

For those of you interested in IR measurement, consider http://www.avsforum.com/avs-vb/showthread.php?t=529224

Do the new rec_imp builds work for people, or does it all still crash?

I have tried the rec_impDS version yesterday on my E-MU 1820M and that worked.
Thanks !
post #163 of 326
Thread Starter 
There is said to be a bug in the loop-back with rec_imp.

Can anyone confirm / refute, before I apply the suggested patch?
post #164 of 326
Thread Starter 
Quote:
Originally Posted by Mooneyass View Post

So, is rec_imp working with ASIO such as my my M-Audio FW410?

We'd be interested in your experience of either the ASIO or DS rec_imp versions.
post #165 of 326
Thread Starter 
A slightly tweaked version of rec_imp_jrp is available on http://www.duffroomcorrection.com/wi...Measuring_Tool. It contains a small fix for loopback use.
post #166 of 326
John,

I have some problems with noise when running the convolution plug-in (Version 2.16).
The same filter is working fine with the Foobar plug-in. Maybe I got the config wrong?

44100 2 2 0
C:\\Program Files\\Convolver\\Convolver\\Filter\\center.wav
0
0.0
0.0
C:\\Program Files\\Convolver\\Convolver\\Filter\\center.wav
0
1.0
1.0

I'm trying to do a simple stereo setup with 2 in and 2 out, both channels using the same DRC filter file.

I have also tried the rec_imp files. Both versions are working, but there is a level difference compared to the Aurora plugins. Both measurements were done with an input level at -1 dBFS. The IR wav from Aurora was close to 0 dB, but the rec_impds version was 25 dB lower. The resulting frequency response was very similar, except for some differences below 30Hz. I will try the new version and compare some loop-back measurements.

I'm using a Digigram VXpocket V2 card.

Bjorn
post #167 of 326
Thread Starter 
Bjorn, The config file looks OK to me.

Do you get the same problem with different partition numbers (0, 2, 4)?

What are you trying to play back?

If you could email me your set-up (config + center.wav) I will have a look (jrp at dial dot pipex dot com)

I cannot really help much with rec_imp -- I've only compiled it, but not done any detailed comparisons with other approaches.
post #168 of 326
I'm trying to play a file ripped from a CD. 16 bit, 44.1 kHz linear PCM.

I only tried 0 and 2 in partition numbers.
Will do another test tonight.
Bjorn
post #169 of 326
Thread Starter 
Well I can't immediately hear anything untoward playing back an MP3 with this filter (albeit with the next release).

I'll keep this under review.
post #170 of 326
The noise is gone!

I had used the Default Direct sound device, but when I changed to VXpocket WDM as playback device it works fine.

I also compared the rec_imp/glsweep to the Aurora plugin. There are two differences.
The generated inverse sweep in Aurora is always at max level (0 dBFS), but in glsweep the level is very low. The file looks as if it's empty when you open it in Audition, but when you zoom in it's there.

This difference doesn't seem to cause any difference in result, though. A loopback results in a nice impulse response.

The cause of the level difference seems to be in the convolving prosess (convolving the recorded file with the inverse sweep). I cross-checked the files in Aurora and glsweep/lsconvolve, and the resulting impulse response was always about 5 dB lower with lsconvolve. The difference seems to be even greater with acoustical measurements.

I'm not sure if this is a real problem. It's not easy to see the real SNR in the impulse response with a log dB scale. A linear scale is more revealing, but Audition don't have this options, as far as I know.

BM
post #171 of 326
Thread Starter 
Good news.

I can't explain the differences between the different impulse response measurement methods ...
post #172 of 326
Quote:
Originally Posted by jrpavel View Post

A slightly tweaked version of rec_imp_jrp is available on http://www.duffroomcorrection.com/wi...Measuring_Tool. It contains a small fix for loopback use.

Dear jrpavel,

as you suggested I tried to implement the patch you made in order to being able to compile the fixed loopback rec_imp version under Linux.
Unluckly I am not really a C/C++ expert so...can you provide a little help?
post #173 of 326
Thread Starter 
Michele, I suggest that you start with the sources of Ed's version (available on the same page) and replace its rec_imp.c with the rec_imp.c from my version. You should then be able to use the makefile from Ed's version.
post #174 of 326
Thread Starter 
Convolver 2.17 is available via http://convolver.sf.net

This release:

* Allows WAV (or other libsndfile-compatible) files to be interpreted as filter paths (without the need for a config text file). The source channels are convolved with the corresponding channels of the WAV file (so not in matrix fashion).

* Allows a config file to comprise a list of filter path filenames, with automatic selection of the first compatible filter path (by number of input and output channels, sample rate), where a filter path filename is either a text config file, or a WAV file

There are some speed improvements.

This is a VC++ 2005 test build.

PS: Ed has put up a new rec_imp on http://www.duffroomcorrection.com
post #175 of 326
Great!

Thank you very much jrpavel!
post #176 of 326
Thread Starter 
Convolver 2.17 is available via http://convolver.sf.net

This release fixes a bug when WAV files are used directly as filters (rather than via a config file). Some slight performance improvements.
post #177 of 326
Thread Starter 
I have updated the http://convolver.sf.net site to make it easier to navigate. I have also added some example config files (http://convolver.sourceforge.net/configegs.html).

If you have any further examples (or interesting filters) that you would be willing to share, please post them to me at jrp at dial dot pipex dot com. Similarly, if you have some practical experience of generating filters, that would be helpful.
post #178 of 326
Thread Starter 
This release allows Convolver to run on all x86 cpus (not just Pentium 4).

Convolver is a (free) real time DSP audio plug-in for Windows Media Player.

The install package also contains two DirectX/DirectShow filter versions ConvolverWrapper or the equivalent ConvolverFilter for Adobe Audition and real time applications such as Zoom Player Pro, TheaterTek 2.2, J.River Media Center (when used with the DirectX host plug-in), or Console.

Convolver will take a set of impulse response (FIR filter) files and convolve them with sound paths mixed from the input channels, mixing the results into a set of specified output channels.

Applications

Why would you want to do this? There are several main applications:
  • With a suitable impulse response generated by a tool such as DRC you will be able to play sound corrected for your room response. For more details, see Ed Wildgoose's Duff Room Correction site.
  • You should also be able to use Convolver for bass management, cross-overs, cross-talk cancellation, equalization and other purposes that require the source signal to be filtered and redirected to different output channels.
  • You can also use Convolver as an effects filter (reverb) in Adobe Audition, or other sound or music application that accepts DX filters.

Features
  • Performance is excellent, possibly the best available under Windows, and subject continual improvement. A stereo 65536-tap filter, the largest that makes sense when applied to a 44.1kHz source, executes at 40 times real time, representing a 3% cpu hit, on a 3.4GHz Pentium 4.
  • Arbitrary-length convolutions for unusual applications (1 million tap limit imposed only as a sanity check)
  • Multi-channel input and output, PCM and IEEE Float
  • Mixing and scaling of both input and output channels
  • Wide range of filter file formats accepted (Microsoft WAV, SGI/Apple AIFF/AIFC, Sun AU/Snd, Raw (headerless 32-bit IEEE float), Paris Audio File (PAF)
    Commodore IFF/SVX, Sphere/NIST WAV, IRCAM SF, Creative VOC, SoundForge W64, GNU Octave MAT4/5, Portable Voice Format, Fasttracker 2 XI, HMM Tool Kit HTK)
  • Sample encodings supported include unsigned and signed 8, 16, 24 and 32 bit PCM, IEEE 32 and 64 floating point, U-LAW, A-LAW, IMA ADPCM, MS ADPCM, GSM 6.10, G721/723 ADPCM, 12/16/24 bit DWVWk, OK Dialogic ADPCM, and 8/16 DPCM.
    Windows Media Player plug-in and DirectShow filter interfaces
    Several filters can be loaded at once. The first to match the playback format (channels, sample rate) is automatically selected.

Requirements
  • Convolver runs on Windows XP/2000 with DirectX9. (It may also run on earlier versions of Windows.)
  • The Windows Media Player plug-in needs WMP10 or WMP9. Otherwise you will get a convolverWMP.dll failed to register message upon installation. If you continue installation in such circumstance, you may, nevertheless, get access to ConvolverFilter / ConvolverWrapper.
  • I have not found a way of running it with the Windows Media Center Edition (MCE) interface.
  • ConvolverFilter and ConvolverWrapper DirectShow filters need a DX host (eg, Adobe Audition).
  • From version 2.19, Convolver runs on all x86 cpus. Earlier versions needed a Pentium 4 class cpu.
post #179 of 326
Thread Starter 
Changes:

Speed improvements, particularly for shorter filters

New tuning rigour option: limit tuning to 1 minute

Fixed ConvolverFilter when more than one partition specified (bug introduced in 2.17)

ConvolverWrapper media format negotiation and properties page fixes (bug introduced in 2.17)
post #180 of 326
jrpavel - thanks for your excellent work on this plugin which I have only recently discovered. I am having the following problem when trying to configure the convolver (same error when using WMP10 plugin and convolverCMD.exe) :

Convolution error: Problem with filter paths: convolver-config.txt: Unexpected exception

My convolver-config.txt is the following :

44100 2 2 0
c:\\left_rps.pcm
0
0.0
0.0
c:\
ight_rps.pcm
1
1.0
1.0

Where left_rps.pcm and right_rps.pcm are the outputs from DRC. I have also tried using a single "filter.wav" file produced by the CoolEditPro process with the same result.

I had a quick look at the source code and it seems to be throwing an unhandled exception in channelpaths.cpp, line 239 but I can't see why. Any ideas what I may be doing wrong ?

Thanks,

Simon
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