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Multiple Amp Questions

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BoredErica

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The Topping PA5 does appear to have a lower self noise than the Buckeye NC252. For desktop listening distance, I believe the PA5 should be sufficient.

The PA5 can output 100 W into 4 ohm (<0.01% THN+N), which equates to 20 Vrms. With sensitivity of 85 dB/2.83V, the loudness unclipped volume at 1 m would be 85 + 20*log10(20/2.83) = 102 dB. Two speakers (and neglecting room gain) will raise it to 105 dB. Using my own personal rule-of-thumb, it will be adequate for 82 dB average listening volume, which IMHO should be plenty loud enough. The wall wart is an annoyance indeed.
Do you know of what mdsimon2 posted in the post above yours regarding some AB amps vs D amps? Do you think it's a problem and I should hold off? Difference of 1db there at 15khz+.

To connect your ODAC/O2 combo to the PA5, an adapter cable like this one should work, although probably not optimal, since it shorts the negative side of the balanced input of the PA5 to ground. You'll probably have to make your own cable if you feel absolutely have to "do it right".
If it'll work and be enough to test if the speakers are functioning as intended, then that is enough.

For some reason Amir stopped measuring amplifier frequency response in to complex loads. If you look at his Behringer A800 review you can see the frequency response deviation at higher frequencies.


@pma recently did some investigation comparing a UCD180 (precursor to Ncore) to a TPA3255 based amplifier which showed the TPA3255 amplifier had load dependent frequency response but the UCD did not.


Michael
This graph then for the Behringer? I don't see a graph here that is freq vs phase like pma's graph. If Amir's not testing it anymore and the Chinese tester isn't either, then perhaps we'll just never know how it performs in this metric...
index.php

Here is Genelec's speaker distance chart. For their coax based Ones (83x1), the smaller ones are good for distances >0.5 m. I'd expect the LS50 Metas to behave similarly.
The Neumann kh80 has a minimum listening distance is 0.5m and for recommended is at 0.8m which is waaay too far for me right now. The Genelec graph breaks it down into recommended vs not recommended. Perhaps for Amir's graph, the Klippel calculated a recommended distance of 0.6m. It's hard to say what basis Genelec and Neumann consider a speaker to be at 'recommended distance' vs 'minimum distance'. I've seen the Genelec graphic before, but it really squished the nearfield distances so it was hard to see what they're saying.
 
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BoredErica

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Please also note that KEF's are best listened to at >10 deg off-axis. Please see this post by Dr Jack Oclee Brown (VP of Technology at KEF).
Wouldn't having the Meta's tweeter not quite at ear height itself produce all the off axis I need? It's a challenge to get the tweeter at ear level even when I'm trying hard to make it happen since it would require a 14in height boost, which is not achieved with good looking desktop stands and achieved with some concessions with yoga block stacks. Or do I need to have the speakers pointing up?

1637974064354.png
When calculating angle from the speaker to ear I think this is the way to calculate it. Angle 2 would be the angle which is determined by how long side A is.

A=14.22in, Angle 2=36 degrees (No height boost for Meta)
A=10.22in, Angle 2 = 28 degrees (4in yoga block)
A=8.22in, Angle 2 = 23 degrees (6in speaker stand)
A=6.22in, Angle 2 = 18 degrees (2x4in yoga block)
A=2.22in, Angle 2 = 6.5degrees (3x4in yoga block)

edit:
Pa5 measurements here if you want them, to save you some time.
 
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NTK

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Do you know of what mdsimon2 posted in the post above yours regarding some AB amps vs D amps? Do you think it's a problem and I should hold off? Difference of 1db there at 15khz+.
WolfX-700 measured FR for 4 & 8 ohms, and there was no difference between them. I believe Topping designed the PA5 to be insensitive to load impedance.
Relative-Level-1-2.jpg
Wouldn't having the Meta's tweeter not quite at ear height itself produce all the off axis I need? It's a challenge to get the tweeter at ear level even when I'm trying hard to make it happen since it would require a 14in height boost, which is not achieved with good looking desktop stands and achieved with some concessions with yoga block stacks. Or do I need to have the speakers pointing up?

When calculating angle from the speaker to ear I think this is the way to calculate it. Angle 2 would be the angle which is determined by how long side A is.

A=14.22in, Angle 2=36 degrees (No height boost for Meta)
A=10.22in, Angle 2 = 28 degrees (4in yoga block)
A=8.22in, Angle 2 = 23 degrees (6in speaker stand)
A=6.22in, Angle 2 = 18 degrees (2x4in yoga block)
A=2.22in, Angle 2 = 6.5degrees (3x4in yoga block)
I'd recommend raising the speakers off the desk as far as you can. Not only because you want to match tweeter height to ear height, but also minimize the effect of the desktop bounce. You can adjust toe-in/out to get your desired off-axis angle too.
 
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BoredErica

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WolfX-700 measured FR for 4 & 8 ohms, and there was no difference between them. I believe Topping designed the PA5 to be insensitive to load impedance.
Relative-Level-1-2.jpg
Wait, is that the graph I need to be looking at for that issue? Doesn't Amir have a similar test? Example from a review 2 days ago:
index.php

For some reason Amir stopped measuring amplifier frequency response in to complex loads. If you look at his Behringer A800 review you can see the frequency response deviation at higher frequencies.


@pma recently did some investigation comparing a UCD180 (precursor to Ncore) to a TPA3255 based amplifier which showed the TPA3255 amplifier had load dependent frequency response but the UCD did not.


Michael
AFAIK that graph by Amir has been in every review, no? Am I missing something?

I'd recommend raising the speakers off the desk as far as you can. Not only because you want to match tweeter height to ear height, but also minimize the effect of the desktop bounce. You can adjust toe-in/out to get your desired off-axis angle too.
It will be pretty rough finding desktop speaker stands that lift the speakers to ear height which fit my white theme. The best I found after an hour of search was the 6in stand. Yoga blocks are also an option, though they reduce stability with each new block added and don't look as good. So I want 2-3 yoga blocks by my calculation.

Now I think about it, having a speaker propped up by 2-3 4in yoga blocks which is smaller than the entire base of the speaker + adding toe in seems like a recipe for disaster...

What you (may) need is loudness compensation/correction. EqualizerAPO provides this functionality. But it only works if you adjust volume via the Windows volume control. Below is the screen shot of the loudness correction calibration screen.
In an ideal system, how would a 2.1 system be set up for volume control? Is everyone using powered subwoofers and passive speakers who don't use AV receivers just using Equalizer APO?
 
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mdsimon2

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Measuring in to purely resistive 4 and 8 ohm loads is not the same as complex loads which are NOT purely resistive (like an actual speaker).

Again, not saying this is an issue but it is somewhat unknown with the PA5. Of course if you buy the Mk5 you can rather easily measure the frequency response at the speaker terminals and either confirm that there is no issue or correct the frequency response deviations with DSP.

Michael
 
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BoredErica

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Measuring in to purely resistive 4 and 8 ohm loads is not the same as complex loads which are NOT purely resistive (like an actual speaker).

Again, not saying this is an issue but it is somewhat unknown with the PA5. Of course if you buy the Mk5 you can rather easily measure the frequency response at the speaker terminals and either confirm that there is no issue or correct the frequency response deviations with DSP.

Michael
True, DSP/EQ can make the problem go away if it's just a FR change. That does put things into perspective. Imo saving $179 and getting the PA3 far sooner with 10db or so less noise is very welcome. Its noise performance can compare with every single speaker amp Amir has tested apart from the Benchmark. Also, the marketing says the amp would stop any pop sounds when the amp powers on and off, and that would be a nice touch. I always got a pop sound when turning amps on and off, be it for headphones or speakers and it's mildly annoying.


I'm still scratching my head at how the average person controls volume with powered sub and passive stereo without an AVR. It seems like a pretty common use case. Maybe I'm just Googling the wrong key words. Guess people just use dac or digital volume controls?

--

index.php
This is not audible right? Doesn't look like it, though there was a 30 page discussion on it so I figured it's worth a passing look.
 
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mdsimon2

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With 2.1 setups if folks are not using an AVR I imagine most are using a miniDSP product like the 2x4HD where volume control is handled by the DSP.

Unfortunately most multichannel DACs are pro audio DACs with limited volume knob / remote volume control features. In my Ultralite Mk5 setup I implement DSP on a RPi running CamillaDSP and handle volume control in CamillaDSP.

Of course there are some multichannel DACs with better volume control options but they are either unavailable (Okto DAC8 pro) or much more expensive (exasound).

With a near field setup I would probably just use the volume knob on the Ultralite Mk5, only reason I do not do that is that I want remote control.

Personally I do not worry about the ESS hump. I can’t tell the difference between my Okto DAC8 pro and my MOTU Ultralite Mk5 when level matched. IMO nothing really touches the Mk5 in terms of sub-$1K multichannel DACs.

Michael
 
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BoredErica

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With 2.1 setups if folks are not using an AVR I imagine most are using a miniDSP product like the 2x4HD where volume control is handled by the DSP.
Unfortunately most multichannel DACs are pro audio DACs with limited volume knob / remote volume control features. In my Ultralite Mk5 setup I implement DSP on a RPi running CamillaDSP and handle volume control in CamillaDSP.

Of course there are some multichannel DACs with better volume control options but they are either unavailable (Okto DAC8 pro) or much more expensive (exasound).

With a near field setup I would probably just use the volume knob on the Ultralite Mk5, only reason I do not do that is that I want remote control.
Seems like the term is 'per channel volume control' looking at Okto's website. So the Motu lacks this functionality.

1. Set Motu and Windows volume to minimum. Play music on loop.
2. If I haven't destroyed the entire speaker system yet, turn Windows volume to 50%.
3. If it's too quiet, turn Motu's volume knob as needed. (Ideally Motu's volume would be max, but that will make probably overhaul attenuation Windows can do in volume settings.)
4. Set up DSP or Equalizer APO to match Meta volume with sub volume. (I guess if I don't do this either speaker or sub will be way louder than the other.)

I'm sorry if I'm not making sense or going back and forth retreading old ground. xD As always your input is greatly appreciated.

Personally I do not worry about the ESS hump. I can’t tell the difference between my Okto DAC8 pro and my MOTU Ultralite Mk5 when level matched. IMO nothing really touches the Mk5 in terms of sub-$1K multichannel DACs.
Following the guidelines in the stickied thread I don't see how the hump could be audible.

index.php

It is so far under the green line. I took Motu's IMD curve and drew the green line (here in purple).

1638012410812.png

Just put in an order for PA5. Not sure how the tracking works on Hifigo.
 
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BoredErica

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1638598274430.png

(Might have to zoom in for this one.)

I am thinking about the FR change based on listening height. I have 3 lines here, which would be the angle off axis with a 6, 6.9, and 9in stand. The 9in stand is the bottom black line and has the smallest FR change due to being off axis, almost at the "recommended 10 degree off axis" some people give for coaxial speakers. The 9in stand is stable enough, though perhaps not in an earthquake. 6.9in stand doesn't look as good but has a built in tilt up.

So my question would be, are these differences in FR realistically audible in music/computer content? I'm leaning towards no, as the differences in shades of red the 3 lines go across are pretty small.

Well, the taller the stand the farther away speakers are from the desk to minimize desk reflections. I can't find any good guidance on how off a speaker from a desk should be, and it might vary from speaker to speaker...

Or maybe just don't sweat it and just go for whichever looks better/is less likely to tip over?

Sorry, I'm probably going over every small thing with a fine grained sieve. It is an expensive purchase so I'd like to get it right.
 

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NTK

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Here are the FR response curves by elevation angles using the data in the included ZIP file in Amir's review for you to mull over.

Yeah. I think you are starting to think too much. I was once told the secret to happiness is to have realistic expectations (or the sure path to misery is to have unrealistic expectations).

FR Plots.png

[Edit] Oops. Wrong plot title. Replaced the above plot to correct the title. These aren't polar. The polar plots are ...
Kef Ls50 Meta v-polar.png
 
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BoredErica

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What would the optimal way to turn on the entire setup be? I was originally thinking of plugging all audio related things into one power strip and finding some kind of product that functions as a remote switch for my power strip so I can turn the strip on and off at my desk without walking to wherever the strip actually is. First I'd have to find such product. But then I realized, if I turned on the entire power strip at once, everything is getting powered on at the same time. They say amps are supposed to be turned last on startup and turned off first at startup. So I guess I'd need 2 switches on my desk, one for the amp and one for the dac + sub? >.>
 

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Assuming you are still considering the Ultralite Mk5 I like using the TOSLINK output with a Bobwire DAT1 to provide a 12 V trigger output when the Mk5 is playing. If your amp and/or sub does not have a trigger output you can use a trigger activated power strip like this -> https://www.adafruit.com/product/2935 to turn them on.

Personally I leave the Mk5 on at all times but use the audio detect functionality of the Bobwire to turn on the amps when an audio signal is detected on the TOSLINK output. When the audio signal stops the Bobwire will stop the trigger after 60 seconds and the amps shutoff automatically.

Michael
 
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Assuming you are still considering the Ultralite Mk5 I like using the TOSLINK output with a Bobwire DAT1 to provide a 12 V trigger output when the Mk5 is playing. If your amp and/or sub does not have a trigger output you can use a trigger activated power strip like this -> https://www.adafruit.com/product/2935 to turn them on.

Personally I leave the Mk5 on at all times but use the audio detect functionality of the Bobwire to turn on the amps when an audio signal is detected on the TOSLINK output.
Still planning on Motu but it's out of stock for probably over 3 more months so I'll have to live without it for a while. I always just assumed a 12v was some kind of physical power switch.

The benefit of the setup you're listing here is that without even having to have a power switch on my desk (reducing clutter), I can wake everything up just by sending some signal to the dac? Maybe I'm supposed to play something on my computer?

If I'm supposed to turn on the amp last and turn it off first, what I could do is keep the dac on 24/7 and just turn the sub/amp on and off instead. If the dac is already on and the amp and sub are off, turning them on would indeed cause them to be turned on last. If everything is on, and the power switch only turns off the amp and sub, turning them off would cause them to be turned off first.

Looking at the Bobwire product page I'm assuming it's this diagram:
1639119552362.png

So you use optical here because the device only accepts optical in. Neither Pa5 nor Rythmik f12 have 12v input/output, so I'm assuming it would go Motu optical -> Bobwire digital in -> Bobwire analoge out -> adafruit in? I don't know where it's supposed to go into the adafruit.
2935-17.jpg

I see 'always on', and that I think I understand. Not sure what 'normally on' and 'normally off' means, and what separates the two. Or what 'switch active is'. 'Power' can't be power in either, because there's the 3 prong connection at the end for that.

When the audio signal stops the Bobwire will stop the trigger after 60 seconds and the amps shutoff automatically.
Do you mean if the source stops playing music for 60s? Is that 60s configurable? There are times where I don't play anything for a period of time but want things ready anyways. Even a 1-2s turn on period between idle periods, 1-2 times a day would be enough to make me scrap the idea.


PA5 has a feature to mute pops when being turned off, but that only works with the power switch on the unit itself AFAIK. Simply cutting power to it elsewhere I think would still cause loud pops.

---

There's also the additional problem of switching between headphones and speakers. The Motu can do this in software I'm sure since there's no physical switch on the unit to do so. However, I don't want to keep Motu software running all the time and even more so, have it open and hit a button in the software to switch. Is there a way to switch with a macro that can be bound to a key on my keyboard? I'm used to using a physical Sescom switch to switch between what goes into my o2/odac.

---
Following the rule Johnyang and others have put out from my Googling to adjust volume at the end of the signal chain when possible, I would use PA5's volume knob to adjust volume, and fine tune it digitally. (The difficult part there isn't hooking up a USB digital volume knob up, but rather finding one that looks nice and feels nice. But that's something I should probably figure out alone.)

Sorry for the bother, I have and will have infinity questions likely until nobody wants to answer them anymore. :(
 
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A way to get around the problem of headphone/speaker output switching without having to open up the Motu software and changing the output (if it doesn't support hotkeys) is to use a seperate headphone dac/amp unit and plug my headphones into that. It will show up as a seperate audio device I can use as the playback device. Then with an Autohotkey script, I can toggle between the two. I already own o2/odac unit so I don't have to buy a new dac/amp for my headphones. This also gives me the freedom to upgrade my headphone dac/amp I suppose, though my o2/odac seems transparent to me and lacks any hiss I hear with many speakers.

First it must be established there's no way to use hotkeys using Motu's software. Then it must be established there's no way to use scripting programs to toggle output gracefully in Motu's software.

The only confounder I can think of is, if I'm using a system that autoshuts down the PA5/subs if Motu doesn't detect a signal for a while, Motu might shut both down if I'm using my headphones for too long. I want my headphones, speakers, and sub to be ready to produce sounds within 2 seconds of an input. Ideally <1s or virtually instantaneous.
 

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Lot's of good questions. I'll try to break them down in to sections.

Bobwire DAT1
Can function in two ways, 1) "audio detect" where it will send a 12 V trigger when audio is played and 2) "input active" where it will send a 12 V trigger when it senses an input (even if no audio is playing). MOTU does have a power on / off button so that would be a viable option to implement option 2. Option 1 is rather instantaneous, any delay will depend more on the start up characteristics of your amp than anything else, in my experience you get sound within 1-2 seconds. The Bobwire trigger signal is output via an 1/8" TS output.

Adafruit Controllable Power Strip
The two "normally off" outlets are controlled by the trigger. These will remain off unless a trigger is provided to the power strip. "Always on" is a non-switched outlet and "normally on" will be turned OFF when the trigger is provided. The trigger input here is in the form of the phoenix terminals on the side where is shows "+ -". You can either make your own cable to convert 1/8" TS to bare wire or use a premade one like this -> https://www.amazon.com/gp/product/B004GIGTQ6/ref=ppx_yo_dt_b_search_asin_title?ie=UTF8&psc=1.

PA5 / Sub Pops
You bring up some good points here and I have had different experiences here depending on amp used. Best thing to do would be test and see if you have bad pops when cutting the power (maybe with a cheap speaker to be safe). If the amp / sub have bad pops then none of this power sequencing stuff will work. I will say that the in my experience the Mk5 itself is completely pop free on turn on / off.

Volume Control
I do not see how using the PA5 volume control is a viable option in your setup as it will NOT control the sub volume. I would either use the MOTU volume control or some sort of software volume control in your computer (potentially with a USB volume knob or IR remote). Given the low noise of the equipment you will not have any issues using digital volume controls especially if using balanced interconnects.

Headphones
I see two options here. The MOTU has an independent volume control for the headphone output which is activated by pressing the volume knob. So you could turn the main volume control all the way down and then press the knob and now the MOTU volume knob will only impact the headphone output and you will have no sound coming from the main speakers. The other option would be a specific DSP profile to eliminate the outputs going to the PA5 / subs when using headphones. We haven't discussed many specifics around your DSP solution (and I personally have no experience with windows based DSP solutions) but imagine changing DSP configurations should be pretty easy. For example in my linux based streamer system running CamillaDSP it is very easy to switch configurations via the web based GUI and it would be possible to change configuration based on a remote input or hot key with some programming.

Michael
 
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BoredErica

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Lot's of good questions. I'll try to break them down in to sections.

Bobwire DAT1
Can function in two ways, 1) "audio detect" where it will send a 12 V trigger when audio is played and 2) "input active" where it will send a 12 V trigger when it senses an input (even if no audio is playing). MOTU does have a power on / off button so that would be a viable option to implement option 2. Option 1 is rather instantaneous, any delay will depend more on the start up characteristics of your amp than anything else, in my experience you get sound within 1-2 seconds. The Bobwire trigger signal is output via an 1/8" TS output.

Adafruit Controllable Power Strip
The two "normally off" outlets are controlled by the trigger. These will remain off unless a trigger is provided to the power strip. "Always on" is a non-switched outlet and "normally on" will be turned OFF when the trigger is provided. The trigger input here is in the form of the phoenix terminals on the side where is shows "+ -". You can either make your own cable to convert 1/8" TS to bare wire or use a premade one like this -> https://www.amazon.com/gp/product/B004GIGTQ6/ref=ppx_yo_dt_b_search_asin_title?ie=UTF8&psc=1.

PA5 / Sub Pops
You bring up some good points here and I have had different experiences here depending on amp used. Best thing to do would be test and see if you have bad pops when cutting the power (maybe with a cheap speaker to be safe). If the amp / sub have bad pops then none of this power sequencing stuff will work. I will say that the in my experience the Mk5 itself is completely pop free on turn on / off.

Volume Control
I do not see how using the PA5 volume control is a viable option in your setup as it will NOT control the sub volume. I would either use the MOTU volume control or some sort of software volume control in your computer (potentially with a USB volume knob or IR remote). Given the low noise of the equipment you will not have any issues using digital volume controls especially if using balanced interconnects.

Headphones
I see two options here. The MOTU has an independent volume control for the headphone output which is activated by pressing the volume knob. So you could turn the main volume control all the way down and then press the knob and now the MOTU volume knob will only impact the headphone output and you will have no sound coming from the main speakers. The other option would be a specific DSP profile to eliminate the outputs going to the PA5 / subs when using headphones. We haven't discussed many specifics around your DSP solution (and I personally have no experience with windows based DSP solutions) but imagine changing DSP configurations should be pretty easy. For example in my linux based streamer system running CamillaDSP it is very easy to switch configurations via the web based GUI and it would be possible to change configuration based on a remote input or hot key with some programming.

Bobwire/Adafruit:
To respond to your two options:
1: This won't work because it will take 1-2s (or worse, Neumann KH80's whopping 6.5s delay) before sound sound comes on after I press play. Then I have to go back and replay whatever I just played.
2: I don't want the Motu to be visible. Minimum it will have wires going out to amp, sub, and power. Maximum it will also have microphone and headphone connected in the front. On top of that, add the Toslink cable. It will have many cables going in and out of the device. Aesthetically it will be a nightmare. I can't be confident at all the Motu's power button will be within arm's reach.

I think a better solution would be a power strip that is controlled by a remote like this. 3 power strips, total 4 power slots. 1 remote which allows for seperate toggle on/off for each of the 3 power strips. This should solve the problem with only 1 small remote AND allow me to turn things on and off in the optimal order. Let me know what you think.

1639156658236.png


Pops:
I'm assuming how bad pops are when turning things on/off has a lot to do with the design of the amp and its power (max?) power output. My suspicion is the pops won't be bad enough to ever damage anything, but it will be enough to be a little annoying.

Volume Control:
Correct, and I screwed up the planning again. The calculated loudness of noise at 9db for Motu + PA5 has enough headroom. So, lower Motu volume as much as is needed so digital volume adjustment isn't weird, then fine tune it with digital volume.

Headphones:
Friend and I want to use Audiolense as DSP. I have almost no understanding of how to use Audiolense or general DSP principles. Friend's supposed to take care of that front and learn all they need to learn about DSP and fill me in when we both have systems set up. If I can't get that working for switching headphones/speakers via a hotkey or script bound to a key, then I guess switching between audio devices via AHK script bound to a key on the keyboard (Motu vs Odac) will be acceptable. The process would take about a second and is automated past pressing the right button.
 
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BoredErica

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2. My personal desire for amplifier power is ~100X (= 20 dB) the power at my average listening volume. If you want to account for occasionally increasing the listening distance by 3X, a 5 dB padding is reasonable. That matches your +25 dB figure. Also, how the amplifier behaves when clipping can be important too.
Okay, I got another question xD

What about for subwoofer frequencies?
1641198986114.png

Following the equal loudness contours, for a 20hz tone to sound as loud as a 65db note, I'd be looking at bit above 110db SPL. In Sweetchaos' thread, the Rythmik f12 goes up to 97.6db at 2m away before considering room gain. So let's say 3db of room gain, taking me up to 101.6db, which is still about 10db off.

People say music is mixed to compensate for our hearing curves, so unweighted the SPL is way higher in low frequencies. But... 20hz is not 40db or more higher in any track I've seen. Wouldn't that make the dynamic range of such a track 40db if not more?

I guess for the purposes of looking at how loud a sub must play at 20hz:
1. Start with 65db SPL at 1khz
2. Analyze spectrum analysis in Audacity and look at how much lower 1khz is vs 20hz. Add difference to 65db.
3. +3db room gain
4. Add or subtract based on subwoofer distance.
 

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Okay, I got another question xD

What about for subwoofer frequencies?
View attachment 176357
Following the equal loudness contours, for a 20hz tone to sound as loud as a 65db note, I'd be looking at bit above 110db SPL. In Sweetchaos' thread, the Rythmik f12 goes up to 97.6db at 2m away before considering room gain. So let's say 3db of room gain, taking me up to 101.6db, which is still about 10db off.

People say music is mixed to compensate for our hearing curves, so unweighted the SPL is way higher in low frequencies. But... 20hz is not 40db or more higher in any track I've seen. Wouldn't that make the dynamic range of such a track 40db if not more?

I guess for the purposes of looking at how loud a sub must play at 20hz:
1. Start with 65db SPL at 1khz
2. Analyze spectrum analysis in Audacity and look at how much lower 1khz is vs 20hz. Add difference to 65db.
3. +3db room gain
4. Add or subtract based on subwoofer distance.
The way I understand the application of the equal loudness curve is this:

Say, when you are at a live (acoustical) concert listening to live instruments, the SPL of the bass is what it is, and the SPL of the high is also what it is. When we hear it we hear a certain ratio of the loudness of the bass vs the high, and that is how they sound "natural" to us. For example, if the SPL of the 1 kHz is 80 dB (= 80 phon) and the 100 Hz is 100 dB (= ~92 phon), the "natural" loudness ratio is 80 phon to 92 phon.

If we listen to its recording at home at a lower volume of 60 dBSPL at1 kHz (= 60 phon), without any loudness compensation, the 100 Hz will be 80 dBSPL (= 62 phon). The loudness ratio is now 60 phon to 62 phon, and is quite different from the original 80 phon to 92 phon. Therefore, we'd want to compensate for it be boosting the low frequencies.

The goal is not to make the 20 Hz sound equally loud as 1 kHz. The music dictates how "loud" the 20 Hz is and how "loud" the 1 kHz is. The mixing and mastering engineers eq the mix to have a ratio that sound "natural" at their specific target listening level. If we listen at a different level and want to reproduce the music with the same loudness ratio, we'll need to apply eq (loudness compensation).

Below is the spectrum of a piano note A0.ff (27.5 Hz, very loud), you can see that there isn't a whole lot of energy in the low and sub-bass, and that's how it sounds natural.
Picture from post: https://www.audiosciencereview.com/.../missing-fundamental.18930/page-6#post-621408

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I don't know if I answer your question. If not, let me know and I'll try again. Happy new year.
 
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BoredErica

BoredErica

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Say, when you are at a live (acoustical) concert listening to live instruments, the SPL of the bass is what it is, and the SPL of the high is also what it is. When we hear it we hear a certain ratio of the loudness of the bass vs the high, and that is how they sound "natural" to us. For example, if the SPL of the 1 kHz is 80 dB (= 80 phon) and the 100 Hz is 100 dB (= ~92 phon), the "natural" loudness ratio is 80 phon to 92 phon.

If we listen to its recording at home at a lower volume of 60 dBSPL at1 kHz (= 60 phon), without any loudness compensation, the 100 Hz will be 80 dBSPL (= 62 phon). The loudness ratio is now 60 phon to 62 phon, and is quite different from the original 80 phon to 92 phon. Therefore, we'd want to compensate for it be boosting the low frequencies.

The goal is not to make the 20 Hz sound equally loud as 1 kHz. The music dictates how "loud" the 20 Hz is and how "loud" the 1 kHz is. The mixing and mastering engineers eq the mix to have a ratio that sound "natural" at their specific target listening level. If we listen at a different level and want to reproduce the music with the same loudness ratio, we'll need to apply eq (loudness compensation).
I think it's illuminating and answered the main question. I heard people say that a lot of music is mastered at 75-85db. I'm assuming that's 75-85db a-weighted. I still find the question of how loud I actually listen to music and how loud the speakers have to go to be confusing.

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https://www.audiosciencereview.com/...ef-ls50-meta-review-speaker.25574/post-873573 to the post.
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ZLYiHSr.jpeg


The above post in Meta thread is what got me thinking (note the 5 likes). I remeasured with my a weighted db meter and at the loudest I'd ever critically listen using the "max" button (highest spl measured is shown on db meter) with a variety of tracks and got 69.5db, so let's round up to 70db. There's going to be some averaging going on there with the db meter. Subtract 3db from 70 since there will be 2 speakers, another 3 for being close to a wall so 63.5db. Let's say I swap to an OLED tv as a monitor and speakers are now 1m away from me. I have over 23db of headroom for music.

If music is mastered at louder average levels than I listen then I'd be boosting bass a bit so there's going to be some EQ there, but only down to 120hz. I guess the spooky part is if room interferences force me to boost bass frequencies above 120hz and the Metas get stressed.

I don't understand what I'm looking at with their imgur link. I'm assuming they're saying a track playing at 70db on average is somehow peaking at near 97db but I have no idea how or why.

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I'm trying to compare the thd graph to the graph of how loud music are at different FR and comparing that to thd audibility graphs and I'm having a hard time. I'm assuming if use a sub and crossed over at 120hz, anything above 120hz in thd would improve over Amir's measurements, or is that only for IMD? I guess I can pull up some spectrograms via Audacity for some of my tracks to see how they land or use this averaged graph from 'AES paper 8960'.

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And... this distortion threshold graph I don't entirely understand:

Masking-thresholds-for-a-1kHz-tone-masker-at-different-levels-masking-tones-from-15.png

I think distortion is not nearly as bad of a problem as people make it out to be absent of insane levels. Amir listened to Metas and said it's fine unless you want 'the room to shake under you and to really feel the music', while I listen at below average levels. I think distortion fears with Metas are overblown when paired with a sub and listening at sane levels. Still, I'm a paranoid person and I also just like to know a little bit more about the specifics of the limits of what is audible or not, and what could potentially be problems in rare cases, in part just for my own education. So, I list what people criticize about gear I'm considering/already bought and my responses in hopes you or somebody else more knowledgeable can weigh in.

----------------

Dr. Jack Ocklee Brown who helped develop the Metas wrote on ASR in a post you or mdsimon shared about how coaxial speakers are best listened to 10 degrees off axis. Erin said in a Kali IN review that EQ should not be based on on axis response since you're want to EQ based off of at least 10 degrees off axis response, and probably 20 degrees (for the Kali). Do you agree with this assessment? Easy EQ via automatically generated Spinorama github results would probably be off then.

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One area that is a bit unknown with the PA5 is frequency response in to complex loads. Hypex shines in this area as it has frequency response that is very independent of load which is not often the class D amplifiers.

Moving to the amp side, as you probably know pma and Restorer-John often bash cheaper class D amps. Drama aside, pma/Restorer-John has brought up 4 points of contention:
1. Class D amps have problems with FR in 'complex loads'.
2. Effective speaker impedance is not simply the lowest part of the impedance graph. Phase angle matters too. Some amps can be underpowered when 4ohm testing suggests they are not.
3. thd+n behavior of many class D amps are not very linear and vary a lot based on frequency + power. They can also be poor in 15khz-20khz range.
4. Bandwidth for class D amps is insufficient and should be up to 200khz.

Here are my thoughts on each:
1. This is the most concerning one. Simply measuring with a mic and applying EQ? I suppose the problem areas tend to be far above frequencies where the room or desk reflections? affect it? Seems like people can't agree if it's feasible to test FR in 'complex loads' and to decide which load, but then I don't understand how pma managed to do it with just a speaker and a mic. Amir said he tested FR through speaker load and said it was only a problem with cheap <$100 amps.

Seems like FR with the 4ohm (resistive?) load Amir does in his tests is a small indicator of amp behavior. Aiyima the pma tested with the 2db deviation, and in Amir's testing 20khz was off by 0.7db while PA5 was off by 0.2db. So, in this limited jank indicator, PA5 exhibits less than a third of the deviation of the Aiyima. I also can't hear above 15khz, so the problem areas would be 10.5khz-15khz and <1db based on that measurement with Aiyima which I consider worst case. Every measurement in Amir's review except max power output shows PA5 outclassing Aiyima 07 and I don't think I have reason to think it won't be the case here. If the error is <1db I fear measurement inaccuracies with my mic will outweigh the correction and do more harm than good.

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Here is pma's testing of Aiyima:
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2. I don't think I listen loudly enough for PA5 to struggle. I don't think this matters in my case. pma would bring out EPDR graphs but that only works for class AB amps and we just don't know how class D amps interact.

3. I don't think it's at audible levels, and the power level of music at 15khz-20khz playing normal music rather than test tones is simply too low to matter. I think this issue is a non-issue.
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4. I struggle to understand the technical reasons behind this objection, but multiple people who seem pretty technical do not agree there is good evidence such wide bandwidth is important.


Happy New Year to you too! Again, sorry if I'm just bringing up infinite questions that may seem simple. I've done... hundreds of hours of reading around the forum at this point and that's before EQ process even begun. Your patience and knowledge is always very appreciated. :)

Meta buying time is when PA5 is on the verge of being delivered and that day is fast approaching. Getting some last minute questions in.
 
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NTK

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One thing first, if you are listening at a reduced volume from the "reference level" (i.e. the level the music was monitored during mastering), you need to accept there is going to be differences. We can at best minimize them, but not totally eliminate them. People going to a symphony orchestra will expect a certain loudness level (almost without exception high). If the same orchestra is playing much softer, the audience is not going to be happy. So there are inherent differences due to volume levels. What I am saying is that it is unproductive to try to optimize everything to the n'th degree. It is just one of the many compromises we make listening to reproduced music, and it is far from a major one.

Then there is what Dr Toole called the circle of confusion. There is no standard for music studios. You really don't know what the "reference level" for each piece of music is. Ideally, the calibration would be something like playing a test signal (say, a wide band pink noise) and adjust the volume until the sound meter reads a certain number at the listening position. Then keep the system volume (gain) setting fixed at that during the mastering session. The listeners at home will calibrate their rooms the same way, and then they know they will be listening at the same volume as the engineer(s) in the studio if they are using the same volume knob setting. If you dial back the knob because you want to listen at a lower volume, you'll also know exactly how much the level has been reduced from the "reference". But that isn't happening, so we can only do it with our ears and educated guesses. And good luck with the self produced music mixed and mastered not in "professional" studios.

Now on to how much SPL we need. The good thing about digital is its max output level is fully deterministic. If you have a voltmeter, disconnect your speakers, set the volume knob at the highest you'll listen to, play a single frequency tone signal at a known level, say -10 dB full scale (you can use Audacity or some websites to supply the signal), and measure the output RMS voltage. (Use a 60 Hz signal if you are using a cheap multimeter.) You can easily convert that into power at whatever load resistance you are interested in. Scale that number to 0 dBFS and you've the max power your amp will put out. Apply whatever headroom you need for EQ, and you can easily judge whether you have enough and if your Metas can handle it. (I personally wouldn't worry too much with intersample-overs. But if you do, then multiply the number by a headroom factor of up to 2 to get your required maximum power.)

Tchaikovsky's 1812 -- the perennial audiophile favorite demo track for its canon shots and therefore dynamic range. You don't listen to it with a pair of desktop speakers. LOL.

The graphs show auditory masking. They show, for a 1 kHz signal, the masking thresholds. If you look at the area under the topmost curve (the Lm = 90 dB curve), that's the area we are not be able to hear in the presence of a 1 kHz 90 dBSPL "masker" tone. For example, we cannot detect a ~60 dBSPL 2 kHz signal in the presence of the 90 dBSPL 1 kHz masker. Since the second harmonic of 1 kHz is 2 kHz, we cannot detect a second harmonic of more than 30 dB down from the 1 kHz fundamental at 90 dBSPL. (Note that the masking thresholds change with the SPL of the masker.) That means our ears aren't capable of detecting less than 3% (= -30 dB) HD2 at that listening level.

I don't have the references at the moment -- for music, we usually can't hear a few percents THD. At low frequencies, the threshold easily goes up to above 10%. So, yes, unless you are exceeding the "useable" capacities of your speakers and amps (and people do), distortions are not a problem. So basically, distortion is not a problem (and therefore can be ignored) until it is a problem.

In the case of the Metas, I'd EQ the listening window (LW) curve instead of the on-axis.

On the class-d amplifier fears, class-d amps are designed to take advantage that our audible range stops at about 20 kHz. They amplify audio by converting it into high frequency pulses (in the case of PA5, ~600 kHz), which the electronics can operate with little losses. The pulses are reconverted back to audio with low pass filtering. Therefore, they, by design, do not have bandwidth to >100 kHz.

Problem withs "complex loads" are only problems for poorly designed amplifiers, be they class AB or D or something else. There is no evidence that PA5 is one of them.

What the EPDR is for is to give a rough idea of the amount of heat dissipation a class AB amp will give when driving those speakers. When there is no problem with overheating, that is not something I worry about.

The FR curves pma showed aren't any surprise. Amir didn't measure the A07 FR at different loads, but he did for the A05. You can see the FR differences at the highs with different loads, and I expect the A07 to be similar. It is no big surprise that the FR of an A07 as measured with speakers is not the same as the Hypex UcD (assumed invariant to loads). When you look at the PA5 FR, there is no difference for 4 & 8 ohms until beyond 20 kHz. Again, not something to worry about.
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