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Battle of the DSP's - MiniDSP+REW vs FIR methods

I'm sorry, I don't think that is correct. Bit depth has no effect on latency as far as I am aware. Only tap count and sampling rate. Do you have a resource I can read?
You don't know what drivers are don't you?
Post in thread 'DAC Input latency tierlist for gaming? Hasn't been done before.' https://www.audiosciencereview.com/...ing-hasnt-been-done-before.53558/post-1967129
Let me reframe your milage will wary from different DAC and interface used. As already told I whose surprised by latency of cheap USB C Apple dongle (EU version).
Edit: if someone wants to play with low latency video streaming:
 
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After a long learning period I think I've got my system close - if not at - the end point of no more tinkering. This was done using REW and MiniDSP SHD Studio at every step of the way. Specifically, this entailed
  1. Optimize positioning (MLP, mains, 2 subs)
  2. Time aligning subs with each other
  3. EQing respective subs and creating a single virtual sub
  4. EQing mains
  5. Setting XOs
  6. Minor room response tweaks to manage impulse response and decay
  7. Repeat for finer tuning,
The end result was that the system sounded far better than ever before, just glorious, and measures well too. When I started I knew next to nothing about DSP of any sort except Audessy (as built into AVRs) and Dirac. My focus was on room correction/EQing and all that entails. More recently I've become more aware of the whole computer audio thing and all the associated tools, particularly HQPlayer. In some circles the REW-measure-MiniDSP approach is held in disdain in favor of FIR methods, often without measurements and, seeming, often with little emphasis on room correction.

So, can anybody comment on respective pros/cons of each approach? To be clear, my main interest is getting the system working well in the room it's in and making measurable improvements. Thanks and cheers,
You can try "directional bass" option where each sub works with just one speaker. Harder to get right but better end result for almost all room shapes with 2.2 systems. Additionally, you can slip-in speaker crossover phase linearization FIR filters in MiniDSP's FIR slots.
 
You can try "directional bass" option where each sub works with just one speaker. Harder to get right but better end result for almost all room shapes with 2.2 systems. Additionally, you can slip-in speaker crossover phase linearization FIR filters in MiniDSP's FIR slots.
Why would you say so? I never found directional bass useful and most people as well based on my forum experience - but might be worng.
 
You can try "directional bass" option where each sub works with just one speaker. Harder to get right but better end result for almost all room shapes with 2.2 systems. Additionally, you can slip-in speaker crossover phase linearization FIR filters in MiniDSP's FIR slots.
It's not that hard (2.2) when you get into it and brings lot of (small) benefits when done properly (significant enough together).
We don't judge on subjective opinion but measurements of how close he got and what might/need improving. Thanks reminding me MiniDSP Flex has FIR export directly from REW so it's cook it and insert. Is it to all chenel banks or input banks only and can it be used in combo? Asking to understand my self (as I don't have one) is it posible to keep single PEQ bank (or two) on input side for self filters reserved for ELC (with FIR). Let me remind you main purpose of 2.2 is to get ELC fit better on lower SPL. Please don't ask me to reapet detail how to setup 2.2 and why I did that a lot of times.
To answer Oddball shortly THD and better integration of psy how we hear together with better control of propagation of fundamentals for main peaks in great majority of ever recorded materials. Some are small, some you might not even hear and some become big more you cross threshold to lower SPL, never the less all real and substantial enough all together. Is it worth it is up to you.
 
To answer Oddball shortly THD and better integration of psy how we hear together with better control of propagation of fundamentals for main peaks in great majority of ever recorded materials. Some are small, some you might not even hear and some become big more you cross threshold to lower SPL, never the less all real and substantial enough all together. Is it worth it is up to you.
Sorry but not understanding the pitch. There are lots of words in there that don't translate into frequency, phase or decay response that I could understand, especially with the graphs. Any better way to present it?
 
Why would you say so? I never found directional bass useful and most people as well based on my forum experience - but might be worng.
It's nearly impossible to implement correctly in D&M through the menus. That's why it was never even an option in A1 Evo but if you are doing DSP from a computer, it can achieve results stereo subs cannot.
 
Sorry but not understanding the pitch. There are lots of words in there that don't translate into frequency, phase or decay response that I could understand, especially with the graphs. Any better way to present it?
Hear we go again...
Represent what? More linear the cone behaviour lower the THD. When you cut what's hard to drive to a driver right above where phase impedance meet what do you get? Exactly that. Be so sound waves are directional until they begin (75 Hz) to sum (+3 dB for total of 6) fully into a standing wave (45 Hz). And they will do it to the lows of physics. You know what I can't hustle dig it from my previous posts if you wish.
Good old info graphic explains it all.
Instrument Freq Range (1).png
 
At the moment, FIR's greatest weakness is that it is a DIY solution. You will need a PC/Mac/Linux box or a RPi. So you will need some computer skills to get it up and running, particularly if you are running Linux. The other problem is that audio routing is not easy, particularly if you have multiple sources. It works best for a single source, you set it up once, and it works. This is FAR from something like a MiniDSP, where you can switch inputs and adjust the volume with a remote control. And the best thing about a MiniDSP is its robustness, no nightmare software updates or software glitches breaking your DSP chain.

I say "at the moment" because @mitchco has a hardware FIR processor in the works. Right now very little is known about it, but if my dreams are fulfilled it will provide MiniDSP-like functionality but with FIR processing.

The other major weakness of FIR is its latency, which is unavoidable. This is dependent on taps/sample rate (NOT on computing power!) and can range between 100ms to 1.5 seconds. This is not really a problem if you are listening to music, but it's a real problem if you need video because of lip sync problems. It can make your system feel sluggish and unresponsive, for e.g. if you change track or adjust the volume, it will happen after a short delay.

As to FIR vs. IIR, FIR is easier to design and more powerful. For e.g. all symmetrical LPF's and HPF's sum to flat, no need to check if certain configurations will have a non-flat sum due to phase interactions. It can do everything IIR can and more. It is the DSP of choice if you like tinkering. And not to mention, it's linear phase. The jury is out whether linear-phase sounds better. I think it does, but that's very much a subjective opinion.
Thank you for that succinct summary. In my own personal experience, when the audibility of an audio phenomenon has yet to be substantiated, I find that I either can't hear it or don't care about it, anyway.
 
I say "at the moment" because @mitchco has a hardware FIR processor in the works. Right now very little is known about it, but if my dreams are fulfilled it will provide MiniDSP-like functionality but with FIR processing.
Thanks, Keith. Some updated info on the stereo version of Hang Loose Processor.
The other major weakness of FIR is its latency, which is unavoidable.
Just wanted to mention that it is avoidable with minimum phase FIR filters, even with high tap counts, there is no inherent filter delay. While you don't get the excess phase correction, you still get excellent frequency resolution with 65,536 tap min phase FIR filters. Assuming the convolver is zero latency ( i.e. direct convolution like HLC) then the only delay is the output buffer. With a 48 kHz sample rate and an audio buffer size of 256 samples adds 5.3ms of delay. Well below audible thresholds and suitable for movies and gaming, but still get that smooth bass response with high tap count frequency resolution.
 
Thanks, Keith. Some updated info on the stereo version of Hang Loose Processor.

Just wanted to mention that it is avoidable with minimum phase FIR filters, even with high tap counts, there is no inherent filter delay. While you don't get the excess phase correction, you still get excellent frequency resolution with 65,536 tap min phase FIR filters. Assuming the convolver is zero latency ( i.e. direct convolution like HLC) then the only delay is the output buffer. With a 48 kHz sample rate and an audio buffer size of 256 samples adds 5.3ms of delay. Well below audible thresholds and suitable for movies and gaming, but still get that smooth bass response with high tap count frequency resolution.
I may have missed some details when this got initially announced, I like the idea it's open source and everything is installed and ready to go. I'm looking forward to when it gets released.
 
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