Fastest Subs?

[edorr]

This is interesting. What does "192k ready" mean? Is it bit transparent up to 192k or only 96k?

I don't think DEQX has an AES out?

It means the system is (dacs / DSP engine etc.) is capable of running at 192/24, but the current version of the software does not support it yet. Trinnov has announced the Amethyst (basically a digitial preamp / streamers / DAC with Trinnov DRC build on the same platform) that runs at 192/24. This product has been announced half a year ago, and once this is up and runnning the ST2 Pro will be software upgraded to 192/24. They do this remotely by taking control of your Trinnov machine over the network.

This feature would make the Trinnov DSP engine one of the few that runs at 192/24 natively.

the digtial outs of the Trinnov pro line are a thing of beauty allowing you to hold on to your audiophile DAC while integrating with either a bunch of subs, or in a full blown MCH system.

 

[dallasjustice]

These are good suggestions Edorr. My only concern is volume control. I know deqx has volume control as well as trinnov but I don't know if I will be taking a step backwards there. The totladac has an unbeatable volume control and I would hate to compromise there. My ultimate plan is to have more than one totaldac and have Vincent link them together to have a unified volume control. If that's the case, then I would need at least 2 AES outs with independent delay, cross-over etc. I know the deqx has multiple digital outs but I don't know if they can be used contemporaneously with the analog outs.

 

[edorr]

These are good suggestions Edorr. My only concern is volume control. I know deqx has volume control as well as trinnov but I don't know if I will be taking a step backwards there. The totladac has an unbeatable volume control and I would hate to compromise there. My ultimate plan is to have more than one totaldac and have Vincent link them together to have a unified volume control. If that's the case, then I would need at least 2 AES outs with independent delay, cross-over etc. I know the deqx has multiple digital outs but I don't know if they can be used contemporaneously with the analog outs.

To keep the volume across four channels in synch you would need to use the digital volume control on the Trinnov. If you're concerned about loss of information in this setup, you should not be be and here is why. You could create some presets in the Trinnov for different listening volume levels, that ensure the Trinnov always operate in the 0 - 15 dB attenuation range. Here is how it would work.

To play "loud", your Totaldac would be set to the highest volume you can play on your Totaldac. You then attenuate say 0 - 15db on the Trinnov - no problem.
To play "intermediate", you turn down your TotalDAC 15db. You create a preset on the Trinnov that trims the subs relative to the mains by 15db. So your channels are still in synch and your master volume on the Trinnov still operates on the 0 -15db range (i.e. no loss). Effectively you are attenuating your subs by up to 30db digitally, but this will be absolutely fine
To play "low", you trim your TotalDAC by 30db etc.

This trick will allow you to always operate the Trinnov in the 0-15 db attenuation range, which should be absolutele fine.

You can select the low, intermediate and loud presets and the master volume control for the Trinnov using iRule and RS232 control. Works like a charm.

 

[edorr]

These are good suggestions Edorr. My only concern is volume control. I know deqx has volume control as well as trinnov but I don't know if I will be taking a step backwards there. The totladac has an unbeatable volume control and I would hate to compromise there. My ultimate plan is to have more than one totaldac and have Vincent link them together to have a unified volume control. If that's the case, then I would need at least 2 AES outs with independent delay, cross-over etc. I know the deqx has multiple digital outs but I don't know if they can be used contemporaneously with the analog outs.

An additional Total DAC just for Subs strikes me as overkill. Just one for mains and Trinnov DACs for subs should be fine. However, even if you go down the 2 x tottalDAC path, the Trinnov still meets all your processing needs and you can use 4 channels of digital out from the Trinnov with 2 TotalDACs.

 

[dallasjustice]

Badazz! Thanks. I was just considering the multiple totaldacs. No final decision on that yet.

To keep the volume across four channels in synch you would need to use the digital volume control on the Trinnov. If you're concerned about loss of information in this setup, you should not be be and here is why. You could create some presets in the Trinnov for different listening volume levels, that ensure the Trinnov always operate in the 0 - 15 dB attenuation range. Here is how it would work.

To play "loud", your Totaldac would be set to the highest volume you can play on your Totaldac. You then attenuate say 0 - 15db on the Trinnov - no problem.
To play "intermediate", you turn down your TotalDAC 15db. You create a preset on the Trinnov that trims the subs relative to the mains by 15db. So your channels are still in synch and your master volume on the Trinnov still operates on the 0 -15db range (i.e. no loss). Effectively you are attenuating your subs by up to 30db digitally, but this will be absolutely fine
To play "low", you trim your TotalDAC by 30db etc.

This trick will allow you to always operate the Trinnov in the 0-15 db attenuation range, which should be absolutele fine.

You can select the low, intermediate and loud presets and the master volume control for the Trinnov using iRule and RS232 control. Works like a charm.

 

[dallasjustice]

Btw, I used to use Irule. I love it. I might go back to it later. I think Itai has really done great with it in the last few years.

 

[edorr]

Badazz! Thanks. I was just considering the multiple totaldacs. No final decision on that yet.

Synching up DAC volumes can be a problem. I actually have to synch up volume of another DAC with 4 channels of Trinnov in a 5.0 configuration (Trinnov DRC is not applied to my surrounds). I bought an NAD M51 because it has RS232, allowing me to integrate Trinnov + NAD volume controls through macros using iRule.

Your 4 channel volume control with Trinnov would be very simple, using the method I descibed. Absolutely no need for a second TotalDAC.

 

[DonH50]

My Rythmiks have a continuous phase control knob, an immense help integrating the subs with the mains. Other subs have built-in delays and other features, DSP-based or analog, but I have not researched them much (and not at all the past year or two).

 

[KlausR.]

Steve,

I am still anxious for someone to tell me what they are going to hear differently when group delay is manipulated or better yet if they can A/B the sound difference with certainty

No subs, but just to have an idea: my speakers have different factory settings with different group delays:



The curves include the time the signal needs to run through the filters, so for a comparison you'd have to shift each curve down to the zero line. The red curve is the linear-phase filter, the green curve the minimum-phase filter. I can switch between settings by pressing a button on the remote control. In the linear-phase setting the bass is more precise, somwhat tighter and better controlled, the difference to the two other setting is not huge, though.

Edit: I add the measurement of a German pro-audio mag, the speakers' -3dB point is at 27 Hz.



Klaus

 

[mojave]

Thanks Mojave! I owned audiolense about 4 years ago and I later sold it because it was too complex for me to use at the time. I believe it can be used as a plugin now with Jriver, right? Anyway, I would consider using it again, if I could find someone to teach me how to use it and set everything up properly. Let me know how the Tascam goes.

Audiolense creates filters based on your sample rate (44.1, 48, 88.2, etc) and your input selection (2.0, 5.1, 7.1, custom). You load a filter into JRiver's convolution engine and JRiver will automatically select the filter needed depending on your source file.

I had a chance this weekend to test out the Tascam US-366. It worked beautifully. It provides AES output on both the optical and coax outputs. I use an RCA to XLR cable that I used to use with my DCX2496 for digital input and set the Tascam driver for AES output instead of SPDIF. I think the digital output only goes to 96 kHz, but you can take a measurement at any sample rate and Audiolense will create filters for the other sample rates that you need. Since the Tascam uses USB power from the computer it needs to be connected. However, the mic input is routed directly to the digital output and doesn't go through the computer at all.

I used REW to do a loopback measurement of my signal chain. I used a DAC that is flat to DC for testing the Tascam's microphone input. In other words, I knew the output on the DAC wasn't going to rolloff at all and any measured rolloff would be from the mic input. On the Tascam's input it is flat down to 10 Hz and only down by .5 dB at 3 Hz. For all intents and purposes it can be considered completely flat. My microphone is down .25 dB at 5 Hz so my entire measurement signal chain is basically completely flat.

The Tascam uses an asynchronous USB connection and could be used as a USB to SPDIF/AES converter. In fact, I tried it with my DAC using optical SPDIF, coax SPDIF, and AES input. So, for about $200 you get a 4 channel DAC with asynchronous USB input, a USB to SPDIF/AES converter, a microphone preamp with almost a completely flat frequency response, and an ADC with optical/coax SPDIF/AES output.

I am currently trying to figure out which combination of hardware provides the best measurement for Audiolense and started a thread today on the Audiolense forum.

I have a Steinberg UR824 which can be used for playback/microphone input. It is has an Asynchronous USB connection and 8 channels of playback.

I also have the Lynx AES16e, a 2 channel AES DAC and an 8 channel AES DAC for playback, and the Tascam US-366 for microphone input. The Lynx AES16e is set as the ASIO audio device in Audiolense. My two AES DACs use the same DAC chips and internal clocks and are designed to be used together for a total of 10 output channels. The two channel DAC uses two DAC chips per channel in a dual mono configuration for higher quality. I hope to use this system to test passive vs active crossovers with my speakers. The only difference in the signal chain would be that the passive crossovers obviously have to go through the crossover boards. With a fully active system, the signal chain would just be HTPC > DACs > Amps > Speakers rather than HTPC > DACs > Amps > Crossover > Speakers.

I have a 7.1 system with a phantom center and two infinite baffle subwoofer manifolds (4x15" drivers each). Each manifold gets it own channel and I run two channel music with the subs in a stereo configuration rather than mono. This means I need 8 channels normally. With active crossovers, then I would need 10 channels.

 

[dallasjustice]

The Tascam looks like a fine unit. What mic are you using? The Tascam 366 looks like a no brainer for this application.

I will use the aes16 to go out to 2 seperate DACs (main left and right + 2 subs). I am not sure I will use audiolense. I will try jriver first. Nyal said that I can loop back my mic into jriver and generate a test signal with REW. I am not sure how that works, but I will find out. My only concern with jriver is that it's IIR, not FIR. I don't know how much ringing, if any, it will add into my digital files.

Audiolense creates filters based on your sample rate (44.1, 48, 88.2, etc) and your input selection (2.0, 5.1, 7.1, custom). You load a filter into JRiver's convolution engine and JRiver will automatically select the filter needed depending on your source file.

I had a chance this weekend to test out the Tascam US-366. It worked beautifully. It provides AES output on both the optical and coax outputs. I use an RCA to XLR cable that I used to use with my DCX2496 for digital input and set the Tascam driver for AES output instead of SPDIF. I think the digital output only goes to 96 kHz, but you can take a measurement at any sample rate and Audiolense will create filters for the other sample rates that you need. Since the Tascam uses USB power from the computer it needs to be connected. However, the mic input is routed directly to the digital output and doesn't go through the computer at all.

I used REW to do a loopback measurement of my signal chain. I used a DAC that is flat to DC for testing the Tascam's microphone input. In other words, I knew the output on the DAC wasn't going to rolloff at all and any measured rolloff would be from the mic input. On the Tascam's input it is flat down to 10 Hz and only down by .5 dB at 3 Hz. For all intents and purposes it can be considered completely flat. My microphone is down .25 dB at 5 Hz so my entire measurement signal chain is basically completely flat.

The Tascam uses an asynchronous USB connection and could be used as a USB to SPDIF/AES converter. In fact, I tried it with my DAC using optical SPDIF, coax SPDIF, and AES input. So, for about $200 you get a 4 channel DAC with asynchronous USB input, a USB to SPDIF/AES converter, a microphone preamp with almost a completely flat frequency response, and an ADC with optical/coax SPDIF/AES output.

I am currently trying to figure out which combination of hardware provides the best measurement for Audiolense and started a thread today on the Audiolense forum.

I have a Steinberg UR824 which can be used for playback/microphone input. It is has an Asynchronous USB connection and 8 channels of playback.

I also have the Lynx AES16e, a 2 channel AES DAC and an 8 channel AES DAC for playback, and the Tascam US-366 for microphone input. The Lynx AES16e is set as the ASIO audio device in Audiolense. My two AES DACs use the same DAC chips and internal clocks and are designed to be used together for a total of 10 output channels. The two channel DAC uses two DAC chips per channel in a dual mono configuration for higher quality. I hope to use this system to test passive vs active crossovers with my speakers. The only difference in the signal chain would be that the passive crossovers obviously have to go through the crossover boards. With a fully active system, the signal chain would just be HTPC > DACs > Amps > Speakers rather than HTPC > DACs > Amps > Crossover > Speakers.

I have a 7.1 system with a phantom center and two infinite baffle subwoofer manifolds (4x15" drivers each). Each manifold gets it own channel and I run two channel music with the subs in a stereo configuration rather than mono. This means I need 8 channels normally. With active crossovers, then I would need 10 channels.

 

[Nyal Mellor]

Audiolense creates filters based on your sample rate (44.1, 48, 88.2, etc) and your input selection (2.0, 5.1, 7.1, custom). You load a filter into JRiver's convolution engine and JRiver will automatically select the filter needed depending on your source file.

I had a chance this weekend to test out the Tascam US-366. It worked beautifully. It provides AES output on both the optical and coax outputs. I use an RCA to XLR cable that I used to use with my DCX2496 for digital input and set the Tascam driver for AES output instead of SPDIF. I think the digital output only goes to 96 kHz, but you can take a measurement at any sample rate and Audiolense will create filters for the other sample rates that you need. Since the Tascam uses USB power from the computer it needs to be connected. However, the mic input is routed directly to the digital output and doesn't go through the computer at all.

I used REW to do a loopback measurement of my signal chain. I used a DAC that is flat to DC for testing the Tascam's microphone input. In other words, I knew the output on the DAC wasn't going to rolloff at all and any measured rolloff would be from the mic input. On the Tascam's input it is flat down to 10 Hz and only down by .5 dB at 3 Hz. For all intents and purposes it can be considered completely flat. My microphone is down .25 dB at 5 Hz so my entire measurement signal chain is basically completely flat.

The Tascam uses an asynchronous USB connection and could be used as a USB to SPDIF/AES converter. In fact, I tried it with my DAC using optical SPDIF, coax SPDIF, and AES input. So, for about $200 you get a 4 channel DAC with asynchronous USB input, a USB to SPDIF/AES converter, a microphone preamp with almost a completely flat frequency response, and an ADC with optical/coax SPDIF/AES output.

I am currently trying to figure out which combination of hardware provides the best measurement for Audiolense and started a thread today on the Audiolense forum.

I have a Steinberg UR824 which can be used for playback/microphone input. It is has an Asynchronous USB connection and 8 channels of playback.

I also have the Lynx AES16e, a 2 channel AES DAC and an 8 channel AES DAC for playback, and the Tascam US-366 for microphone input. The Lynx AES16e is set as the ASIO audio device in Audiolense. My two AES DACs use the same DAC chips and internal clocks and are designed to be used together for a total of 10 output channels. The two channel DAC uses two DAC chips per channel in a dual mono configuration for higher quality. I hope to use this system to test passive vs active crossovers with my speakers. The only difference in the signal chain would be that the passive crossovers obviously have to go through the crossover boards. With a fully active system, the signal chain would just be HTPC > DACs > Amps > Speakers rather than HTPC > DACs > Amps > Crossover > Speakers.

I have a 7.1 system with a phantom center and two infinite baffle subwoofer manifolds (4x15" drivers each). Each manifold gets it own channel and I run two channel music with the subs in a stereo configuration rather than mono. This means I need 8 channels normally. With active crossovers, then I would need 10 channels.

Which DACs are you using?

 

[edorr]

Audiolense creates filters based on your sample rate (44.1, 48, 88.2, etc) and your input selection (2.0, 5.1, 7.1, custom). You load a filter into JRiver's convolution engine and JRiver will automatically select the filter needed depending on your source file.

I had a chance this weekend to test out the Tascam US-366. It worked beautifully. It provides AES output on both the optical and coax outputs. I use an RCA to XLR cable that I used to use with my DCX2496 for digital input and set the Tascam driver for AES output instead of SPDIF. I think the digital output only goes to 96 kHz, but you can take a measurement at any sample rate and Audiolense will create filters for the other sample rates that you need. Since the Tascam uses USB power from the computer it needs to be connected. However, the mic input is routed directly to the digital output and doesn't go through the computer at all.

I used REW to do a loopback measurement of my signal chain. I used a DAC that is flat to DC for testing the Tascam's microphone input. In other words, I knew the output on the DAC wasn't going to rolloff at all and any measured rolloff would be from the mic input. On the Tascam's input it is flat down to 10 Hz and only down by .5 dB at 3 Hz. For all intents and purposes it can be considered completely flat. My microphone is down .25 dB at 5 Hz so my entire measurement signal chain is basically completely flat.

The Tascam uses an asynchronous USB connection and could be used as a USB to SPDIF/AES converter. In fact, I tried it with my DAC using optical SPDIF, coax SPDIF, and AES input. So, for about $200 you get a 4 channel DAC with asynchronous USB input, a USB to SPDIF/AES converter, a microphone preamp with almost a completely flat frequency response, and an ADC with optical/coax SPDIF/AES output.

I am currently trying to figure out which combination of hardware provides the best measurement for Audiolense and started a thread today on the Audiolense forum.

I have a Steinberg UR824 which can be used for playback/microphone input. It is has an Asynchronous USB connection and 8 channels of playback.

I also have the Lynx AES16e, a 2 channel AES DAC and an 8 channel AES DAC for playback, and the Tascam US-366 for microphone input. The Lynx AES16e is set as the ASIO audio device in Audiolense. My two AES DACs use the same DAC chips and internal clocks and are designed to be used together for a total of 10 output channels. The two channel DAC uses two DAC chips per channel in a dual mono configuration for higher quality. I hope to use this system to test passive vs active crossovers with my speakers. The only difference in the signal chain would be that the passive crossovers obviously have to go through the crossover boards. With a fully active system, the signal chain would just be HTPC > DACs > Amps > Speakers rather than HTPC > DACs > Amps > Crossover > Speakers.

I have a 7.1 system with a phantom center and two infinite baffle subwoofer manifolds (4x15" drivers each). Each manifold gets it own channel and I run two channel music with the subs in a stereo configuration rather than mono. This means I need 8 channels normally. With active crossovers, then I would need 10 channels.

Quite intriguing. Depending on what other pioneers like Nyal have to say about audiolense, I may give this a whirl myself. No rush, as I am waiting for more passive treatment to come in first. The 90 seconds "demo" stikes me as a bit of a joke though. Is there any way to thouroughly test this software (and compare to Dirac) before having to commit to buying it?

 

[mojave]

The Tascam looks like a fine unit. What mic are you using? The Tascam 366 looks like a no brainer for this application.

I use an iSEMcon EMM-7101-CHTB with custom calibration down to 5 Hz. The mic is very linear, has a high dynamic range, and low noise. It requires an SMB to XLR cable which is part #CX-BFXM-2m and cost $35. You then connect this directly to an XLR mic cable and a preamp with phantom power like the Tascam.

Which DACs are you using?

There is a local retired audio engineer named Charles Martin who, with his wife, hand makes DACs, ADCs, and preamps for audio engineers all over the world. The 2 channel DAC is made with dual mono Burr Brown PCM1794A chips and ADA4898-1 opamps. The 8 channel DAC is similar, but made with singe PCM1794A chips for each output. I'll send you a PM.

 

[Imperial]

For group delay to even work, at these low frequencies, that is to blend perfectly. Some point to remember:
The human ear can place a sound heard by both ears equidistant within a 1 degree zone.
If it's to either side, from the head, it's only accurate to within a 15 degree zone. 360 degrees describing a full circle...

So when dialing in subwoofers. Also try to sit 90 degrees, that is facing directly at a side wall while listening, then flipping around 180 degrees again. Facing the other wall.
There will be a slight difference... sitting either "side" way one of them should work better. That's you dominant ear. The dominant brain half is of course the other side of you noodle.
Now you have found the side your sub should be placed on. (If you have only one..).

Confused? Listen with the ear you usually hold your cellphone...
If you start out placing the sub outside the 15 degree window on you non dominant side, you will struggle dialing in.
If you place it correctly, according to you dominant side, it will sound, one's dialed in ... "Faster" than on the other side... is it placebo?

The theory, is that your dominant ear has a greater intelligibility, or detects easier Minutiae in discrepancies regarding blend and placement.

SO....
Place two people in the same chair, aske them: Is it a fast sub?
Depending on the dominant ear, one will say yes, the other might not... thinking it does not integrate fully or otherwyze..
(However, if you flip the sub or mirror image it, he might be more positive...) All things else being equal..

This is lessened with two or more subwoofers, I think.

Just something I think could make sence...
Am I far off or even totally lost?
I think I'm on to something

So what is the give here?
Do please try to place your lone subwoofer and run any and all digital auto adjustments on BOTH sides of the room (mirror image the placement..) before making say an decision to not purchase, or trade in to something else.
Who knows, right?

Imperial.