Using convolver universally in HTPC

[antani]

In fact is the same, because the output level of all channels including sub is in any case levelled slightly below the sound card clipping.

In theory you might go both ways, but all the VST bass managers work with +10 Db, so there is no real option.

Hi,
Can you explain how I define in bass manager which channel is LFE? In your diagram I guess it is the channel 8. The bass manager user interface shows 8 channels (basic view) but it does not show any channel numbers next to the settings. Is the LFE fixed to rightmost (channel 8?) controls?
BR,
Ns

antani wrote:

You need to set up Directwire in this way:

It will route the audio streams from a WMD program like Powerdvd towards ASIO.
Then you need a VST host such as Console. You should create an Asio project.
The project should look like this:

The 5.1 channels will be processed in this way, from left to right:
- a convolution engine, Voxengo Pristine Space but you can use Convolver as well
- a parametric equalizer (SpinEQ)
- a time delay, to adjust speakers distance (Voxengo Audio Delay)
- a bass manager (Kelly Industries) to boost LFE channel by 10 Db and to route bass frequencies towards the subwoofer.

I doubt you can get a more sophisticated sound elaboration .

This is another example of final result, amplitude graph is slightly decreasing because I set the target curve in this way:

[antani]

I'm using the Adobe Premiere version from Kelly Industries because I had problem with the standard one.
In this version you can choose the sequence of the channels by clicking in the bottom left corner (L R Ls Rs C LFE or L R C LFE Ls Rs).
.

In my sound card LFE is channel 6, but with a 5.1 setup.

In any case it should not be difficult to identify the LFE with some trials.

Thanks,

Another question: As the bass manager is located AFTER the eq/room correction (pristine space) for me it looks like the room equalization can't be applied for the side channel's low frequencies routed to bass management sub channel. Or the correction is not correct if it is made based on side channel signal measurements.

Can you explain how this is handled? Are you using bass management also for measurement signal when recording the impulse response for side channels (i.e. actually sub channel is recorded/corrected for low frequencies of side channels)?

[antani]

nsiltala wrote:

...Are you using bass management also for measurement signal when recording the impulse response for side channels...

Exactly, in this way you optimize the integration of sub with other channels.

Console is not officially supported with Vista. Seems to work in some level but when I create an ASIO project with Prodigy it freezes. Do you know any other alternative plugin host for this setup with Vista?

UPDATE: Now I got it working - not sure what was the reason but when I created other parts first and added prodigy last then it didn't freeze anymore.

Now I can get the sound through test config in attached picture. However if I turn dry adjustment to minimum I don't get almost any sound. if I understand correctly the convolved signal is adjusted via "wet" level adjustments and by pass signal via "dry". Therefore it looks to me that the convolved audio path is not working.

Could you share screenshot from your prisitne space settings? Should the dry path be disabled fully?
Thanks

[antani]

It should look similar to this:


Each wav mono filter should have a slot, each channel should refer to a different slot. If you are using stereo wav filters you will use only 4 slots.

Dry is not important, as you said is a bypass, and should be muted. You should only take care of the wet. Each channel's wet level should be adjusted manually. You might want to try A-gain.

Thanks a lot again.

I forgot to say that I use audiolense for creating filters http://www.juicehifi.com/index.html.

With Audiolense I can select the speaker config and it creates filter with needed amount of channels. So I assume that then I should use only one slot in Pristine Space for all channels.

Anyway there is no sound even from channel one in wet mode, so there is some other problem. Need to study a bit more.

[antani]

I noticed you created a 44.1 KHz filter. I suggest to use 48 KHz. DVD and most of Blu Ray audio tracks use 48 KHz.

I'll take a closer look to Audiolense, it seems interesting.

Hi, I had problems with measurements with vista and now trying with XP.

With XP I have problem that sound is full of pops and clicks. Did you have this kind of problems?

[antani]

Try increasing the latency of the audio card.

Thanks for advice, I already bought a new display adapter nvidia 9600GT.

With that one it works now fine also for bluray. Not sure why but it fixed the audio issues as well.

By the way, is there some way during windows start up to start up also the console so that it automatically opens the specific configuration file and starts processing?

My system has lip sync problem, audio is clearly behind video (about 500ms) so it annoying. Any ideas how to adjust?

[BerntR]

You can use AC3 filter or FFDShow as DVD codec. Both has manual video delay.

[pakru]

Hi! Sorry for bumping this old thread...

I had a few newbie questions regarding this scheme, since I would like to try room correction/eq in a similar manner.

1. For surround sound (5.1) room correction, for drc, do we generate the correction filters one channel after another (L, R, C, Ls, Rs) and include the sub in the measurement and correction process (a total of 5 room response correction filters)?
2. In the console snapshot, what is the purpose of the Equalizer after the correction filter convolution
3. If I need to convert two channel stereo to pro-logic II or dts neo:6, how do I add crossover filters into the chain if I want to completely bypass the a/v receiver and direct drive the amplifier from the soundcard? Also, is this advisable?
4. I would like to buy the audiotrak 7.1 prodidy card, but I have read that these have very low level (0.5-0.6volts) soundcard output signals on the analog outs. Will it be a problem if I need to drive the sound through 12feet of co-ax cables to the amplifier?

Thanks for your help!

[antani]

pakru wrote:

...
1. For surround sound (5.1) room correction, for drc, do we generate the correction filters one channel after another (L, R, C, Ls, Rs) and include the sub in the measurement and correction process (a total of 5 room response correction filters)...!

Yes, but the filters are 6: L, R, C, Ls, Rs and LFE.

pakru wrote:

...
2. In the console snapshot, what is the purpose of the Equalizer after the correction filter convolution
...

To Give some additional flexibility, however this is not very useful and currently I'm just using the convolution, without any additional equalization.

pakru wrote:

...
3. If I need to convert two channel stereo to pro-logic II or dts neo:6, how do I add crossover filters into the chain if I want to completely bypass the a/v receiver and direct drive the amplifier from the soundcard? Also, is this advisable?...

I'm not sure this is possible.

pakru wrote:

...
4. I would like to buy the audiotrak 7.1 prodidy card, but I have read that these have very low level (0.5-0.6volts) soundcard output signals on the analog outs. Will it be a problem if I need to drive the sound through 12feet of co-ax cables to the amplifier?...

Cannot tell, I am using short cables. Other options, of much higher quality and price, are RME sound cards. RME drivers allow loopback routing of audio streams.

[pakru]

thanks for the superquick response!

antani wrote:

Yes, but the filters are 6: L, R, C, Ls, Rs and LFE.

So, I am guessing that in the bass management, all the LFE frequencies are summed to the final LFE output. Also, how do i make sure the levels of the L,R,C,Ls and Rs for the room correction convolution with the sub included is the same? Is it done by calibrating each channel one by one during measurement with the sub connected for each channel?

antani wrote:

To Give some additional flexibility, however this is not very useful and currently I'm just using the convolution, without any additional equalization.

Sorry for another question...How did you estimate the audio delay in the voxengo audio delay vst?

antani wrote:

I'm not sure this is possible.

I was seeing that in the cyberlink powerdvd, they have the option to do dolby/dts psuedo surround. I guess if it can software encode this, then with directwire, it should be possible to do the console based correction chain...

antani wrote:

Cannot tell, I am using short cables. Other options, of much higher quality and price, are RME sound cards. RME drivers allow loopback routing of audio streams.

OK. I guess I will try my luck since my PC is a little farther away from the amp

Appreciate your answers to all these questions and some more as I move along...

[antani]

pakru wrote:

...in the bass management, all the LFE frequencies are summed to the final LFE output....

Correct.

pakru wrote:

...how do i make sure the levels of the L,R,C,Ls and Rs for the room correction convolution with the sub included is the same? Is it done by calibrating each channel one by one during measurement with the sub connected for each channel?...

Yes, first you should set the volume of the sub at a reasonable level compared to the other channels. You don't need to be precise because DRC will equalize the level anyway. Once all the filters are created you will need to balance the volume of the 6 channels again.

pakru wrote:

...How did you estimate the audio delay in the voxengo audio delay vst?...

(distance of the farthest speaker - distance of each single speaker)/speed of sound

pakru wrote:

...I was seeing that in the cyberlink powerdvd, they have the option to do dolby/dts psuedo surround. I guess if it can software encode this, then with directwire, it should be possible to do the console based correction chain...

If you use a sw player to apply pro-logic II or dts neo, then there is no problem. In theory you might apply the same effects with appropriate vst plug-ins, but I don't know if it's worthwhile.

[pakru]

antani, thanks for your answers and help...i just need to decide on the soundcard now.

[pakru]

Thanks to this thread (and antani), I was able to get 5.1 channel audio working in this fashion using my onboard sound card. In case anybody else is trying to do this method of room correction without buying the prodigy/ESI sound card, here is what I did:

1. Install Virtual Audio Cable (VAC) which allows you to route from any software player (at least the ones I tried) to another device similar to the directwire driver in the ESI/Prodigy sound card lines
2. Install ASIO4ALL (Free)
3. Install Console
4. In console, choose ASIO4ALL as the audio device
5. In the ASIO4ALL configuration, choose appropriate sample rate, disable all inputs and outputs other than VAC as the input and RealTek HD audio output (in my case)
6. Set ASIO4ALL buffer size to max (2048 samples) and zero the input and output latency compensation. Also make the kernel buffer size to max (4)
8. Open VAC control panel and setup one VAC cable with 6 channels (for 5.1 setup)
9. On the motherboard soundcard driver, enable 5.1 playback
10. Using Media Player Classic or other players capable of doing DTS/Dolby decoding or virtual sorround decoding, play the media file and choose VAC cable 1 (setup above) as the output
11. In console route the 6 channel VAC input device to a bass manager as in antani's posts above and connect the output of the bass management VST (Kelly Industries) to the 5.1 output device
12. Enjoy (I also configured the receiver to 6 channel direct mode to pass the 6 channel audio through to the amplifier)

I was able to make this work with any software player that can decode to 5.1 channels (Media Player Classic, VLC, TheaterTek DVD, etc). Latency is not too bad for audio playback.

Next step is to get the room equalized in drc...

[mchabana]

Hi all,

thanks to that wonderful thread and the information mostly provided by Antani, i succeeded in using VST plugins with my Prodigy card on my HTPC system. Once thing bothers me however : i have been able to create filters with REW and then use them with pristine space but i have never been able to export a measurement with REW and then treat it with DRC. It seems that DRC expects a file format that is different from the wav files generated by REW and i experiment failures during the first phases of the file treatment . Could somebody point me to a procedure that would explain how REW can be used along with DRC ?

Thanks & Regards,

Al

[antani]

In REW you should select Export -> Impulse Response as wav and choose mono 32 bit.

In order to process the file with DRC, you need to convert the wav file to pcm 32 bit floating point.

You can do it with SOX or with Audacity.

With sox you should use this command:

sox.exe impulseresponse.wav -t raw -c 1 -f -4 impulseresponse.pcm

Then feed impulseresponse.pcm to DRC.

In the meantime I changed my Audiotrack with a EMU 1616M, a really nice piece of hardware, with superb performance and simple to configure with Console .

[mchabana]

Hi,

thanks to the explanation, i now have .pcm files processed and generated by DRC. I have a few more questions about that :

- in order to use Pristine space, you need to convert the pcm files back to wav files. I tried that with sox again and wanted to see how they looked like in REW but i only succeeded in producing 16 bits files (-2 option) that are compatible with REW (although they might be compatible with Pristine Space, i don't know). What is the process you use to generate your wave files back ?

- i saw that you applied a house curve to your FR. Did you produce that with DRC or did you apply a house curve filter to the DRC filter using REW ?

- i also noticed that you use the +512 option under Pristine Space. Why do you need it (ringing ?) and do you just need to validate that option or is it necessary to move some cursor in the GUI ?

Grazie Mille ;-)

Al.

[antani]

mchabana wrote:

...What is the process you use to generate your wave files back ?...

I use sox to create a wav file 32 bit floating point. You can't import it back in REW, I don't know why, but it's perfectly fine for Pristine Space. If you want to see the shape of filter in the frequency domain you can use Audacity.

mchabana wrote:

...
- i saw that you applied a house curve to your FR. Did you produce that with DRC or did you apply a house curve filter to the DRC filter using REW ?...

I set the target curve in DRC. However, with the latest release of DRC (3.0.1), I use a flat target curve corrected with the psychoacoustic target.

mchabana wrote:

...
- i also noticed that you use the +512 option under Pristine Space. Why do you need it (ringing ?) and do you just need to validate that option or is it necessary to move some cursor in the GUI ?...

In this way I can switch between dry and wet audio streams without any gap in the reproduction. It's relevant only for testing and to show to friends the effect of the equalization.
Even without the +512 option Pristine Space adds the latency required by the mixed phase FIR filter. If you are using Pristine Space for video reproduction, I suggest you not to use it to minimize latency.

[mchabana]

Hi,

this time i succeeded in generating the impulses with DRC thanks to the information you provided. As far as the measurement is concerned, it is almost perfect but it actually does not sound ok and i have too much attenuation in the treble range. Do you know if it is possible to only apply correction up to a certain frequency range with DRC ? I tried to modify the config file but it does not work as expected.

Thanks again for everything,

Al.