Summing or other processing (what kind ?) should give no problem but downsampling may not be so easy. Most softwares only work with one samplerate. Generally, downsampling is used by hardware processors to use less computing power on low frequency channels. But this should not be a probem with a modern PC.
If you really need downsampling, this could be done internally within a software but when this software is connected to another software (through Jack, VAC or ...) or passes the signal to the hardware soundcard, the sampling frequency is back to the original one.
A long time ago, I saw one paper on loudspeaker-room EQ where downsampling was used for other reasons than just processing power but I don't remember why, nor the authors (maybe Craven and Gerzon ?)...
gedlee said:
What exactly are trying to achieve?
Are you wanting to take 5.1 audio and place a HP at 200hz on the LCR and surrounds then route the stuff below 200hz all to LFE? Or are you wanting to do the same but without HP on the LCR and surrounds?
I'm just playing around in console right now and if you've got a clear and detailed picture of what you want to do then I should be able to set that up and show you.
ShinOBIWAN said:
Shin
Sorry for the OT, but the above comment got my attention and I'd like to come back to it if you don;t mind.
I have a dual core AMD processor and on Blue ray it is maxed out and won't play some disks. Player is WinDVD. Its not a slow processor, but apparantly the ATI HD2600 Pro video card is not taking any of the load from the CPU. I am not that "up" on video cards, is this one NOT capable of doing any of the video decoding or is it just not setup right? And if it is Not doing any of the decoding what is the lowest end ATI card that will. I'm tired of having to return Blu-Ray disks when they won't play because there isn't enough processing power.
Shin
I know precisely what I want to do.
First I want the 5.1 bit stream to go out to SPDIF unchanged. That has to happen.
Then I want to LP the sum of the LFE, Left, center and right at about 200 Hz. Now it would be totally pointless to process a signal LP at 200 Hz at 48 kHz. It only makes sense to resample at lets say 4 kHz to reduce the latency, CPU load etc. As above, CPU load is a big issue in my system.
Now that I have the LP filtered mono signal, I want to split it out to three signals, one for each sub, with arbitrary gain, phase, delay and some filtering. This can all be minimum phase, but FIR also works if its 4 kHz as the delay is not a big deal. But I'm pretty sure that IIR filters will be fine.
I'd also like to add some reverb to these signals, but it HAS to be different for all three signals. Not just different signals, but different initial delays and looped signal delays. Basically this randomizes the three signals - decorrelates them - relative to each other.
If there are any more deatils that you want please let me know.
Such a piece of software would make the JBL $1200 unit obsolete.
Thanks
I know precisely what I want to do.
First I want the 5.1 bit stream to go out to SPDIF unchanged. That has to happen.
Then I want to LP the sum of the LFE, Left, center and right at about 200 Hz. Now it would be totally pointless to process a signal LP at 200 Hz at 48 kHz. It only makes sense to resample at lets say 4 kHz to reduce the latency, CPU load etc. As above, CPU load is a big issue in my system.
Now that I have the LP filtered mono signal, I want to split it out to three signals, one for each sub, with arbitrary gain, phase, delay and some filtering. This can all be minimum phase, but FIR also works if its 4 kHz as the delay is not a big deal. But I'm pretty sure that IIR filters will be fine.
I'd also like to add some reverb to these signals, but it HAS to be different for all three signals. Not just different signals, but different initial delays and looped signal delays. Basically this randomizes the three signals - decorrelates them - relative to each other.
If there are any more deatils that you want please let me know.
Such a piece of software would make the JBL $1200 unit obsolete.
Thanks
gedlee said:
ATI HD2600 has a revision Avivo HD and should be able to deliver some assistance to CPU. You shouldn't need to buy a new card, although the more recent generation of cards will likely offer superior performance. An ATI 4850 is a good cost effective current generation card.
The video player you use has to be compatible with hardware decoding function of your graphics card though. Are you using WinDVD 9+ Blueray?
I use PowerDVD 8 Ultra myself and this worked with both my old ATI 3870 and the current 4870. As I said in an earlier post, without assist I see 80% and with, its down to 30%.
OK this does what you ask:
From input it sums the LCR and LFE channels plus applies a gain reduction to avoid clipping whilst at the same time applying a 60dB/oct LP at 200hz(configurable).
Next step is the signal is split into 3 identical feeds and routed to independent 10 band PEQ and gain for each of the 3 subs.
After that is independent and fully configurable reverb for each sub.
Final step are independent delays for the subs.
The processed signals then goes back to the soundcard for routing to whatever analogue outputs you desire.
Also note that CPU usage on a 3Ghz quad core is only 4.5% with all these plugins running. No need to mess around downsampling to 4Khz since its academic and adds unneeded complexity.
An externally hosted image should be here but it was not working when we last tested it.
From input it sums the LCR and LFE channels plus applies a gain reduction to avoid clipping whilst at the same time applying a 60dB/oct LP at 200hz(configurable).
Next step is the signal is split into 3 identical feeds and routed to independent 10 band PEQ and gain for each of the 3 subs.
After that is independent and fully configurable reverb for each sub.
Final step are independent delays for the subs.
The processed signals then goes back to the soundcard for routing to whatever analogue outputs you desire.
Also note that CPU usage on a 3Ghz quad core is only 4.5% with all these plugins running. No need to mess around downsampling to 4Khz since its academic and adds unneeded complexity.
ShinOBIWAN said:
I use WinDVD 9+ (the only one that will play Blueray) and the CPU load is never less than 80% and often goes to 100% and freezes the system or won't play audio smoothly. So even 4.5% CPU load is too much with my current setup.
Thanks for the posting on the processing. Looks like reverb is the CPU hog.
This still passes the full signal out to SPDIF?
gedlee said:
If that's the case then you really need to get GPU assist working and/or up your processing power. I'd recommend the Intel core2duo 3Ghz or faster processors.
Yes, its not shown in that routing scheme that I showed but its trivial to add it in and was explained on the previous page of this thread.
Also I just sent you a mail.
Hi
finally – several month after my decision to go that route – I took the time to set up my PCXO
Not sure if anybody ever did an open baffle speaker together with a PCXO....
First of all – Thanks a lot shin for inspiring me with your thread !!!
Also big thanks for kind help and for sharing so much first hand knowledge to Stig Erik at this place for his early support
http://memweb.newsguy.com/~stigerik/html/body_audiosys.html
With my short summary I hope to raise some further interest in PCXO in general by contributing some more details and "how to" from my experience so far.
Hardware is kind of different to what was discussed here and might be an interesting alternative to some .
PC used is a cheap old FS AMILO pro notebook
- quiet fan,
- IEEE 1394
- Celeron-M 1.4GHz
- XP
- 700MB RAM
-
running cool at a roughly 40% CPU power for a three way XO – enough to calculate new XO filters parallel to undisturbed music listening .
Soundcard is a Firewire Mackie Onyx 400F
http://www.mackie.com/products/400f/index.html
with
- 8 analogue pro outputs,
- switchable AES / SPDIF input / output
- MIDI in / out
- 4 analogue pro inputs
- 4 analogue pro mic-inputs
Software I use is the well known CONSOLE as a relatively cheap and pretty reliable host and the (until now
) not so well known RevolVerb from Red State Sound
http://revolverb.hostrocket.com/
Additional plugin are the versatile Voxengo SoundDelay (http://www.voxengo.com/product/sounddelay/ ) and the excellent NyquistEq. ( http://magnus.smartelectronix.com/ )
My biggest concerns about PCXO always have been stability on a daily usage basis.
It's really no fun to get speakers burned with flimsy software or due to bad hardware integration.
Therefor I went with a fresh and clean XP and did several weeks of stability testing - routing all sounds over the PC and using volume control and input switching - prior to setting up the PCXO.
Like for everybody else one of the first questions that arise with active XO is about multi channel volume control.
The solution I finally settled with is top quality, easy and cheap but not really common.
Key is the ability of CONSOLE (like most other host's) to handle MIDI signals. This is kind of ancient technique but comes in very handy to control my plugin's.
All I needed is a cheap and universally programable MIDI controller – like the Behringer B-Control Nano for example - hooked up to the MIDI-in of my ONYX soundcard.
http://www.behringer.com/BCN44/index.cfm
Second I needed is a plugin that allows for gain control via MIDI – which are *many* (in fact almost all of them).
You can assign the same MIDI signal to as many plugin's you like – providing 100% synchronous multi-channel volume control.
For a 24 bit audio gear losses are neglectible IMO when doing digital volume control – at least its as good as any good NAIM analog preamps I have heard (even slightly better if this is possible at all).
There are beautiful looking plugins for free, designed especially for the purpose of gain control in DAW's, but I found that only NyquistEq (set flat) is stable *and* sonically transparent especially when it comes to low and very low volume settings.
Good thing is - you can also use this setup for switching between different audio sources connected to the soundcard inputs, or different source routing to your headphones with independent volume control - making the MIDI controller effectively kind of universally configurable preamp interface.
There are only very few VST freeware convolvers with SIR and CONVOLVER being the probably best known for Windows.
SIR I havent managed to get to work as there was too heavy CPU load for the purpose and CONVOLVER does run more than one instance only as long as you don't close CONSOLE – after a reboot, CONSOLE comes up with all CONVOLVER plugins filled with the same impulse response file – nothing you would like with a tweeter fed with bass signals for example.
Having asked John Pavel as the head behind CONVOLVER this is due to restricted registry access of his plugin. A patch / hack of the ConvolverVST-plugin allowed me to use multiple instances - though I couldn't get it working without hiccup.
As for now I cant say much about the benefits of my PCXO having to sort out some delay troubles between channels first. Hope to overcome it with the help of SoundDelay-plugin.
But after all the hurdles already taken everything is basically working stable and has a comfortable and very reliable user interface.
A simple RTA measurement – not so precisely relaying on synchronous channels – shows very encouraging first results.
One last word to the Mackie soundcard. It is really good sounding, has "better than decent" specs and is working absolutely stable and pop free. Also there is a bigger brother - the Onyx 1200F - if one needs even more channels - or a volume knob handling all channels simultaneously if desired .
There are cheaper USB soundcards out there but AFAIK USB *can* put you into troubles with time critical applications such as sound processing due to its "unpredictable" (time wise) bus protocol.
Firewire simply is superior / head ache free in this respect.
Downside is, that Linux support of firewire soundcards (ffado / bebob) still is in its early stage and BruteFIR on an USB stick isn't an option at the moment.
Oh, and yes - also thanks to Uli for his excellent before and after sales support of ACOURATE (and special thanks for making the tutorials available as downloadable files).
Michael
finally – several month after my decision to go that route – I took the time to set up my PCXO
Not sure if anybody ever did an open baffle speaker together with a PCXO....
First of all – Thanks a lot shin for inspiring me with your thread !!!
Also big thanks for kind help and for sharing so much first hand knowledge to Stig Erik at this place for his early support
http://memweb.newsguy.com/~stigerik/html/body_audiosys.html
With my short summary I hope to raise some further interest in PCXO in general by contributing some more details and "how to" from my experience so far.
Hardware is kind of different to what was discussed here and might be an interesting alternative to some .
An externally hosted image should be here but it was not working when we last tested it.
PC used is a cheap old FS AMILO pro notebook
- quiet fan,
- IEEE 1394
- Celeron-M 1.4GHz
- XP
- 700MB RAM
-
running cool at a roughly 40% CPU power for a three way XO – enough to calculate new XO filters parallel to undisturbed music listening .
Soundcard is a Firewire Mackie Onyx 400F
An externally hosted image should be here but it was not working when we last tested it.
http://www.mackie.com/products/400f/index.html
with
- 8 analogue pro outputs,
- switchable AES / SPDIF input / output
- MIDI in / out
- 4 analogue pro inputs
- 4 analogue pro mic-inputs
Software I use is the well known CONSOLE as a relatively cheap and pretty reliable host and the (until now
) not so well known RevolVerb from Red State Soundhttp://revolverb.hostrocket.com/
Additional plugin are the versatile Voxengo SoundDelay (http://www.voxengo.com/product/sounddelay/ ) and the excellent NyquistEq. ( http://magnus.smartelectronix.com/ )
My biggest concerns about PCXO always have been stability on a daily usage basis.
It's really no fun to get speakers burned with flimsy software or due to bad hardware integration.
Therefor I went with a fresh and clean XP and did several weeks of stability testing - routing all sounds over the PC and using volume control and input switching - prior to setting up the PCXO.
Like for everybody else one of the first questions that arise with active XO is about multi channel volume control.
The solution I finally settled with is top quality, easy and cheap but not really common.
Key is the ability of CONSOLE (like most other host's) to handle MIDI signals. This is kind of ancient technique but comes in very handy to control my plugin's.
All I needed is a cheap and universally programable MIDI controller – like the Behringer B-Control Nano for example - hooked up to the MIDI-in of my ONYX soundcard.
An externally hosted image should be here but it was not working when we last tested it.
http://www.behringer.com/BCN44/index.cfm
Second I needed is a plugin that allows for gain control via MIDI – which are *many* (in fact almost all of them).
You can assign the same MIDI signal to as many plugin's you like – providing 100% synchronous multi-channel volume control.
For a 24 bit audio gear losses are neglectible IMO when doing digital volume control – at least its as good as any good NAIM analog preamps I have heard (even slightly better if this is possible at all).
There are beautiful looking plugins for free, designed especially for the purpose of gain control in DAW's, but I found that only NyquistEq (set flat) is stable *and* sonically transparent especially when it comes to low and very low volume settings.
Good thing is - you can also use this setup for switching between different audio sources connected to the soundcard inputs, or different source routing to your headphones with independent volume control - making the MIDI controller effectively kind of universally configurable preamp interface.
There are only very few VST freeware convolvers with SIR and CONVOLVER being the probably best known for Windows.
SIR I havent managed to get to work as there was too heavy CPU load for the purpose and CONVOLVER does run more than one instance only as long as you don't close CONSOLE – after a reboot, CONSOLE comes up with all CONVOLVER plugins filled with the same impulse response file – nothing you would like with a tweeter fed with bass signals for example.
Having asked John Pavel as the head behind CONVOLVER this is due to restricted registry access of his plugin. A patch / hack of the ConvolverVST-plugin allowed me to use multiple instances - though I couldn't get it working without hiccup.
As for now I cant say much about the benefits of my PCXO having to sort out some delay troubles between channels first. Hope to overcome it with the help of SoundDelay-plugin.
But after all the hurdles already taken everything is basically working stable and has a comfortable and very reliable user interface.
A simple RTA measurement – not so precisely relaying on synchronous channels – shows very encouraging first results.
One last word to the Mackie soundcard. It is really good sounding, has "better than decent" specs and is working absolutely stable and pop free. Also there is a bigger brother - the Onyx 1200F - if one needs even more channels - or a volume knob handling all channels simultaneously if desired .
There are cheaper USB soundcards out there but AFAIK USB *can* put you into troubles with time critical applications such as sound processing due to its "unpredictable" (time wise) bus protocol.
Firewire simply is superior / head ache free in this respect.
Downside is, that Linux support of firewire soundcards (ffado / bebob) still is in its early stage and BruteFIR on an USB stick isn't an option at the moment.
Oh, and yes - also thanks to Uli for his excellent before and after sales support of ACOURATE (and special thanks for making the tutorials available as downloadable files).
Michael
Cool stuff Michael.
I always like how people find their way through to different solutions yet still unified by the same goal.
The Mackie 400F interface is a good price at ~£450 with all that I/O connectivity. I got excited about the larger 1200F when you mentioned it but sadly it still only has the same 8 analogue outputs as the 400F. Do you know if you can use multiple interfaces on the same computer to increase channel count? RME allows this with the Fireface series so it might be the same here.
For people looking at more than 8 channels, I recently spotted the Echo AudioFire 12. It supports 12 analogue outputs and seems good value at about £500:
http://www.echoaudio.com/Products/FireWire/AudioFire12/index.php
The midi control is something I looked at myself but in the end chose to use scripting software to tie IR remote commands with volume control or whatever else is needed.
PS. Convolver can be configured to offer upto 16 channel of input and output with one instance using text configuration files but I agree that using multiple instances is messy with that software. Thanks for the link to Revolverb, I didn't know about that one. I'll try it out to see if it offers better sound than convolver.
I always like how people find their way through to different solutions yet still unified by the same goal.
The Mackie 400F interface is a good price at ~£450 with all that I/O connectivity. I got excited about the larger 1200F when you mentioned it but sadly it still only has the same 8 analogue outputs as the 400F. Do you know if you can use multiple interfaces on the same computer to increase channel count? RME allows this with the Fireface series so it might be the same here.
For people looking at more than 8 channels, I recently spotted the Echo AudioFire 12. It supports 12 analogue outputs and seems good value at about £500:
http://www.echoaudio.com/Products/FireWire/AudioFire12/index.php
The midi control is something I looked at myself but in the end chose to use scripting software to tie IR remote commands with volume control or whatever else is needed.
PS. Convolver can be configured to offer upto 16 channel of input and output with one instance using text configuration files but I agree that using multiple instances is messy with that software. Thanks for the link to Revolverb, I didn't know about that one. I'll try it out to see if it offers better sound than convolver.
Any way to route audio to Art Teknica Console using Jack for Windows
I don't know much about the Jack port for Windows, but I'm hoping I can use it to supply Art Teknica's console with audio input (from any arbitrary Windows program that outputs sound/music)
Does anyone know how to do that?
Is there a better program out there?
I don't know much about the Jack port for Windows, but I'm hoping I can use it to supply Art Teknica's console with audio input (from any arbitrary Windows program that outputs sound/music)
Does anyone know how to do that?
Is there a better program out there?
ackcheng said:
Yes I could only get the VST version to include more than 1 in and 1 out when using text configuration. The DX plugin came up with errors when trying to create 2 in and 8 out for me. A shame because I would like to use more than 16 partitions since my CPU is very fast and can easy handle 64 or even 128 giving 4-8 times less latency than 16.
Re: Any way to route audio to Art Teknica Console using Jack for Windows
Jack for windows is currently only really useful to connect software that uses ASIO drivers (mostly DAW software) - jack presents a dummy ASIO hardware interface in this case. Unfortunately there is no direct sound plugin. It is possible however to write software that natively supports jack under windows.
It's a very useful tool though, eg
Foobar2000 + ASIO output plugin -> jack -> reaper -> jack -> soundcard
Daveis said:
Jack for windows is currently only really useful to connect software that uses ASIO drivers (mostly DAW software) - jack presents a dummy ASIO hardware interface in this case. Unfortunately there is no direct sound plugin. It is possible however to write software that natively supports jack under windows.
It's a very useful tool though, eg
Foobar2000 + ASIO output plugin -> jack -> reaper -> jack -> soundcard
Re: Re: Any way to route audio to Art Teknica Console using Jack for Windows
Having trouble figuring out howto use Jack for Windows still.
This URL is really similar to what I'm trying to do...
http://www.eggheadcafe.com/software/aspnet/33315737/looking-for-help-with-a-s.aspx
Connect foobar2000 to Thuneau allocator lite using Jack.
ack_connect foobar2000ut1 Allocator:in1
jack_connect foobar2000ut2 Allocator:in2
jack_connect Allocatorut1 systemlayback_3
jack_connect Allocatorut2 systemlayback_4
jack_connect Allocatorut3 systemlayback_5
jack_connect Allocatorut4 systemlayback_6
jack_connect Allocatorut5 systemlayback_7
jack_connect Allocatorut6 systemlayback_8
fb said:
Having trouble figuring out howto use Jack for Windows still.
This URL is really similar to what I'm trying to do...
http://www.eggheadcafe.com/software/aspnet/33315737/looking-for-help-with-a-s.aspx
Connect foobar2000 to Thuneau allocator lite using Jack.
ack_connect foobar2000
ut1 Allocator:in1jack_connect foobar2000
ut2 Allocator:in2jack_connect Allocator
ut1 systemlayback_3jack_connect Allocator
ut2 systemlayback_4jack_connect Allocator
ut3 systemlayback_5jack_connect Allocator
ut4 systemlayback_6jack_connect Allocator
ut5 systemlayback_7jack_connect Allocator
ut6 systemlayback_8
ShinOBIWAN said:
Shin, I haven't tryed by myself but that's what Mackie says:
http://www.mackie.com/products/400f/download/v3.2.8/400F_328_ReleaseNotes.pdf
RELEASE NOTES Onyx 400F Software Update • July 11, 2007 Version 3.2.8
New Features and Improvements Multi-unit Support
The 400F now supports daisy-chaining of multiple units, to form one large recording device. For example, you could daisy-chain three 400Fs together for a 30 input system, with 12 mic inputs, 12 line inputs, and 6 digital inputs.
On a Windows system, the sum of inputs and outputs will appear as one large ASIO device to your Windows DAW of choice. On an OS X system, you will need to combine the units into one large “aggregate device” in the
“Audio MIDI Setup” application, located in Applications/ Utilities.
For anyone interested in the Onyx, Soundcard there is an amazing thread over there in Harmony central where Anderton and Mackie people discuss simply each and any facet of this piece of gear
http://acapella.harmony-central.com/forums/showthread.php?t=1097071&highlight=asio
Regarding text configuration of CONVOLVER – I too haven't managed to get it working – and besides that you would loose the main benefits of CONSOLE to have free routing capabilities as text configurtation connects directly to the soundcars in / out AFAIK.
This scripting software you use to control volume via IR – sounds like a very cool idea - where can I find additional information about that?
Michael
Re: Re: Re: Any way to route audio to Art Teknica Console using Jack for Windows
OK, have you tried using qjackctl? Also, for me foobar had to be actively playing for qjackctl to see its outputs. Once it was configured and saved, it would automatically connect.
Daveis said:
Having trouble figuring out howto use Jack for Windows still.
This URL is really similar to what I'm trying to do...
http://www.eggheadcafe.com/software/aspnet/33315737/looking-for-help-with-a-s.aspx
Connect foobar2000 to Thuneau allocator lite using Jack.
ack_connect foobar2000
jack_connect foobar2000
jack_connect Allocator
jack_connect Allocator
jack_connect Allocator
jack_connect Allocator
jack_connect Allocator
jack_connect Allocator
OK, have you tried using qjackctl? Also, for me foobar had to be actively playing for qjackctl to see its outputs. Once it was configured and saved, it would automatically connect.
Hi Michael
You’re not the alone..
Those are some pics of my system:
An externally hosted image should be here but it was not working when we last tested it.
An externally hosted image should be here but it was not working when we last tested it.
An externally hosted image should be here but it was not working when we last tested it.
I’m currently running Bidule as VST host using the internal Wav files player as source.
Hardware is RME HDSP 9652 and Apogee DA-16x
Thanks to Shin for inspiring me using pc-based xover solution
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