Volume Control with advanced Bass management

[Hear Here]

That is another bunch of codswallop. More advanced DSP software lets you do what you want, and you can choose to leave the high freqs alone if you want to. In fact, I suggest that most DSP users leave the high freqs alone, unless they have high confidence that they have taken proper measurements and they know what they are doing.

You have missed an unavoidable fact about DSP. I agree users of DSP should limit its adjustment zone to sub 500 Hz or even less. BUT there is no way the higher frequencies can avoid encountering the DSP - digital signal processor – unless you pass the full signal though an active XO first and send the top end to a non-DSP amp while the bass goes through DSP and its own amp.

With typical amps with built-in DSP the ENTIRE SIGNAL has to pass through the filter despite adjustments being made to the bass only. As with any unnecessary processors (eg tone controls, graphic equalisers digital processors) there is a loss of ultimate sound quality. Don’t believe this? Just switch between No Filter and DSP filter when connected to very high-quality speakers. Play a bass-light track with plenty of top-end detail and you’ll hear the loss of top end detail.

Do you really believe that music couldn’t be reproduced in the home pre the invention of DSP? Perhaps in those days, people took a bit more care in choosing and setting up their kit. DSP is no substitute for this.

Keep your codswallop to yourself! Do you know the origins of that word by the way?

 

[meesbaker]

I think it is generally not advised to combine two usb-devices since you can't sync the timing between the two, at least not easily. There is a whole thread about this here somewhere, but I wasn't able to finde it just now.

Consumer-level multichannel-Dacs are rather rare, the Topping DM7 and Octo DAC 8 Pro have both been discontinued. More common are professional audio interfaces, but those are more commonly designed for studio use, so often lack "consumer features" like remote control for example. They also almost allways use balanced outputs, which neither your amplifier nor subwoofers may have, then you need adapter-cables, and so on...

I personally use a Audient Evo 16 for stereo and 3 subs and am quite happy with it, but there is definitely a learning curve for the interface, plus the DAW-integration, routing, Dirac quirks... an AVR is the easier way, plus has the advantage of decoding formats like Atmos which you may want in the future. But yeah, they certainly also have their disadvantages.

Hey I did find a thread about syncing but it seemed more a windows issue, someone said the combine into 1 aggregate in macos automatically syncs.

That topping dm7 you mentioned measures perfect in all disciplines - seems like the answer to all problems. How could they take that off the market :O

 

[Ropeburn]

Don’t believe this? Just switch between No Filter and DSP filter when connected to very high-quality speakers. Play a bass-light track with plenty of top-end detail and you’ll hear the loss of top end detail.

Musicians do this all the time during recording and performing - hearing the direct signal from the instrument side by side with the recorded or realtime processed one (AD->DA). And they're usually VERY hard, if not impossible to tell apart. The times of directly audible quality loss from one cycle of digital conversion even in cheap equipment have been over for a long time.

I see you added "very high quality speakers" as a caveat. Almost as if you know what you're doing there - "your system might not be resolving enough", that's what it sounds like.

 

[levimax]

Hey I did find a thread about syncing but it seemed more a windows issue, someone said the combine into 1 aggregate in macos automatically syncs.

That topping dm7 you mentioned measures perfect in all disciplines - seems like the answer to all problems. How could they take that off the market :O

There are some workarounds to combine multiple DAC's but they are fraught with issues. While the topping DM7 seems ideal I believe there were some issues and it is discontinued anyway. I have had really good luck with the pro interfaces. The Motu Ultralite MK5 has 10 channels and a lot of useful features and is well supported and just plain works. I have not tried it but there are supposedly "remote control" solutions. Outside of a remote I don't feel like I am missing any features for hi-fi use.

 

[meesbaker]

Okay - so a good multichannel usb interface! Got it!

Would optimally the interface itself regulate volume or the operating system, or dont matter ?

 

[levimax]

Would optimally the interface itself regulate volume or the operating system, or dont matter ?

I like having both... adds flexibility and "safety". One of the best features of the Motu is that you can set the volume control to control all the outputs simultainiously which is perfect for multi-channel and subs. Not all interfaces do this but I find it almost a mandatory feature.

 

[Keith_W]

With typical amps with built-in DSP the ENTIRE SIGNAL has to pass through the filter despite adjustments being made to the bass only. As with any unnecessary processors (eg tone controls, graphic equalisers digital processors) there is a loss of ultimate sound quality. Don’t believe this? Just switch between No Filter and DSP filter when connected to very high-quality speakers. Play a bass-light track with plenty of top-end detail and you’ll hear the loss of top end detail.

If the high frequencies pass through the DSP processor with no DSP applied, you still think there is a loss of quality?

Do you really believe that music couldn’t be reproduced in the home pre the invention of DSP? Perhaps in those days, people took a bit more care in choosing and setting up their kit. DSP is no substitute for this.

They had no way to control room modes apart from room treatment and other physical manipulations and very rough corrections, i.e. graphic equalizers. There was no ASR, no REW, measurement software and microphones were expensive, and most people only had a subjective idea that rooms made a difference with no way to objectively quantify it without the expense of hiring a professional.

So yes, music could be reproduced, but it could not be reproduced accurately. Just like before the invention of digital audio, most people had no means to accurately reproduce sound.

To dismiss DSP is to misunderstand audio on a very fundamental level. I did not think I would come across such people on ASR ... but there you go.

 

[TheKoC]

Ive read people said something that it is "only 16 bit" , idk if that matters or what part of the measurements that refers to.

The Flex HT has 32Bit/96khz max DACs, DSP-Processing (including Dirac) is done at 32Bit/48 Khz. How much processing power it has compared to a DSP running on your PC i don't know, but you can allways just use it as an 8-channel DAC.
It also has an eARC HDMI Input which audio interfaces usally don't have, and a remote

 

[Hear Here]

If the high frequencies pass through the DSP processor with no DSP applied, you still think there is a loss of quality?

Yes - and easily demonstrated with high quality speakers

There was no ASR, no REW, measurement software and microphones were expensive, and most people only had a subjective idea that rooms made a difference with no way to objectively quantify it without the expense of hiring a professional.

Ah, what bliss!

We just enjoyed the music from systems we took great care in buying and seting up. Now we seem to think Signal Processing is all that matters - we hardly need to bother visiting showroom, arranging home demos, or putting hours of effort into setting up our chosen kit - we just leave everything to the magic of DSP.

Why should we spend big sums on accurate equipment is we trash its accuracy because we've been sloppy in our choice of kit, in setting it up or in a bit of audio housekeeping to minimise what we perceive as room problems!

Worse - Why should brands continue striving for accuracy if their customers chuck DSP at their systems to hide their poor response curves? I'm happy to measure my system's performance but not to mess with the signal to "improve" its performance. There are many better ways to recreate the excitement of listening to a live performance without resorting to DSP.

 

[meesbaker]

The Flex HT has 32Bit/96khz max DACs, DSP-Processing (including Dirac) is done at 32Bit/48 Khz. How much processing power it has compared to a DSP running on your PC i don't know, but you can allways just use it as an 8-channel DAC.
It also has an eARC HDMI Input which audio interfaces usally don't have, and a remote

I think it was about this part:

Distortion free range is usually like up to 22 bit on the top devices and someone in the comments of the review is holding that against it.

Since nobody else reacted to his comment in that thread idk how or if at all that part matters.

 

[TheKoC]

That's the review of the Flex, not the Flex HT, so an older 2channel product. Even so it would not concern me in practical use.

Here's a review by a member of the Flex HT:

P

'miniDSP Flex HT - Review and Measurements'

Nov 22, 2023

This is a review of the miniDSP Flex HT. The Flex HT is the latest iteration on the Flex platform. HT stands for „home theater“.

Note: The unit tested has been send to me by miniDSP for feedback and testing purposes. I am not associated with miniDSP nor did I receive compensation for this review. MiniDSP had no editorial influence.

The unit offers USB, coax SPDIF, optical SPDIF, bi-directional eARC. WISA will be added through a future hardware update. I tried to bring up HDMI on my PC, but was not successful. Will try it later on a denon receiver.

...

• Philipp Paul Klose
• Replies: 66
• Forum: DACs, Streamers, Servers, Players, Audio Interface

• 

 

[meesbaker]

That's the review of the Flex, not the Flex HT, so an older 2channel product. Even so it would not concern me in practical use.

Here's a review by a member of the Flex HT:

P

'miniDSP Flex HT - Review and Measurements'

Nov 22, 2023

This is a review of the miniDSP Flex HT. The Flex HT is the latest iteration on the Flex platform. HT stands for „home theater“.

Note: The unit tested has been send to me by miniDSP for feedback and testing purposes. I am not associated with miniDSP nor did I receive compensation for this review. MiniDSP had no editorial influence.

The unit offers USB, coax SPDIF, optical SPDIF, bi-directional eARC. WISA will be added through a future hardware update. I tried to bring up HDMI on my PC, but was not successful. Will try it later on a denon receiver.

...

• Philipp Paul Klose
• Replies: 66
• Forum: DACs, Streamers, Servers, Players, Audio Interface

• 

Awesome!

 

[RayDunzl]

That idea collides with everyone saying that DAC/streamer devices always have to be in fixed mode because digital volume control at this stage degrades the signal quality.

Not everyone. I don't say it.

Here is a visual of full volume vs -60dB digital attenuation.

-60dB will take you from a normal loud to barely hear anything at all.

Top, full scale digital signal
Middle, what -60dB digital looks like at the same zoom level
Bottom, what -60dB digital looks like zoomed in by a factor of 1000

Not much appears to be missing from the -60dB waveform except amplitude.

 

[Ropeburn]

Not everyone. I don't say it.

Here is a visual of full volume vs -60dB digital attenuation.

-60dB will take you from a normal loud to barely hear anything at all.

Top, full scale digital signal
Middle, what -60dB digital looks like at the same zoom level
Bottom, what -60dB digital looks like zoomed in by a factor of 1000

View attachment 466385

Not much appears to be missing from the -60dB waveform except amplitude.

Yes indeed. The resolution is so high (24bit), if you attenuate by 60dB, you still have 14bit of data left, which is VERY nice for such a quiet signal and orders of magnitude better than our hearing resolution.

And that's just looking at the actual data and theoretically. The processing itself is done in 32bit, just like the volume control on a properly implemented DAC. You can attenuate 24bit data by 48dB in the 32bit realm before you lose ANY data.

 

[meesbaker]

This forum is the best thing ever

 

[sfogg]

Just throwing this out there but in some setups you can also do overall volume control after bass management and EQ. I am doing this with BSS Audio equipment for example. The stuff is crazy expensive new but it is possible to find it very cheap used on ebay. The biggest thing to watch for is what model you are looking at (some don't have DSP, and those that do the amount of DSP varies) and what input and output cards it has installed. .One chassis can hold 4 cards, each card has 4 channels. So up to 16 channels either input or output per chassis. There are analog input, analog output, digital input, digital output and AEC cards available.

You can also easily chain devices for more input/output channels too and the processing inside the box can handle much more than 16 channels.

For example, this processing is in one Blu160/JBL SDEC4500P. My Blu160 has (2) digital input and (2) digital output cards. My setup goes back to analog in (2) Blu120/SDEC4500x boxes.

Essentially, as an input this is taking the channels from two different surround processors (18 channels from Lexicon QLI-32 and 8 from a MC-12HD) and acting as a source selector between them. For the MC-12 it is using the mixer to derive 4 height channels. Then performing bass management to derive stereo subwoofers. Applying a high/low pass to every channel (NTM 52), putting a 30 band graphic EQ on every channel, putting a parametric EQ on every channel (10 PEQs each I think), delay for every channel (time alignment), then applying crossovers to my L/C/R to biamp them, PEQ for each of the drivers of the LCR (with delay), then it is going into the master volume control (with trims for each channel for channel balancing). From their the signal goes to limiters (just to prevent overload) and then two additional delay blocks (front/rear) so I can easily add AV Sync delay or to add extra delay to my surrounds and then output to the two Blu120 boxes which I use in two different locations to get back to analog.

All of this is using about 77% of the DSP capacity in Blu160. It also has many other processing types available including FIR filters.

Steep learning curve to this setup but it is enormously flexible. Right now I am programming it all to control other equipment in my theater. It is already controlling the QLI-32 over TCP and my scaler through RS232.

 

[meesbaker]

Just throwing this out there but in some setups you can also do overall volume control after bass management and EQ. I am doing this with BSS Audio equipment for example. The stuff is crazy expensive new but it is possible to find it very cheap used on ebay. The biggest thing to watch for is what model you are looking at (some don't have DSP, and those that do the amount of DSP varies) and what input and output cards it has installed. .One chassis can hold 4 cards, each card has 4 channels. So up to 16 channels either input or output per chassis. There are analog input, analog output, digital input, digital output and AEC cards available.

You can also easily chain devices for more input/output channels too and the processing inside the box can handle much more than 16 channels.

For example, this processing is in one Blu160/JBL SDEC4500P. My Blu160 has (2) digital input and (2) digital output cards. My setup goes back to analog in (2) Blu120/SDEC4500x boxes.

View attachment 469777

Essentially, as an input this is taking the channels from two different surround processors (18 channels from Lexicon QLI-32 and 8 from a MC-12HD) and acting as a source selector between them. For the MC-12 it is using the mixer to derive 4 height channels. Then performing bass management to derive stereo subwoofers. Applying a high/low pass to every channel (NTM 52), putting a 30 band graphic EQ on every channel, putting a parametric EQ on every channel (10 PEQs each I think), delay for every channel (time alignment), then applying crossovers to my L/C/R to biamp them, PEQ for each of the drivers of the LCR (with delay), then it is going into the master volume control (with trims for each channel for channel balancing). From their the signal goes to limiters (just to prevent overload) and then two additional delay blocks (front/rear) so I can easily add AV Sync delay or to add extra delay to my surrounds and then output to the two Blu120 boxes which I use in two different locations to get back to analog.

All of this is using about 77% of the DSP capacity in Blu160. It also has many other processing types available including FIR filters.

Steep learning curve to this setup but it is enormously flexible. Right now I am programming it all to control other equipment in my theater. It is already controlling the QLI-32 over TCP and my scaler through RS232.

Thats so interesting thank you man!

 

[sfogg]

No problem, if you have questions just let me know. Just started working with it all a couple of months ago but it is impressive, and if you shop well very little money. I bought both my JBL SDEC4500x boxes (Blu120 with all analog output cards) for under $200 each and the JBL SDEC4500P (Blu160 with 3 analog input and 1 analog output) for about $275 and another Blu160 with digital input and output cards for about the same.

You can play around with the software here. You want the Harman HiQNet Audio Architect. To be really confusing the hardware also has earlier software to design it called HiQNet London Architect. They are similar but not the same and designs aren't cross compatible. Some tutorials are using the old stuff, some use the new stuff.