Ask an Acoustic Engineer (me)

Hi, thus what is the right way to keep it under control? Im using Carplay into head unit, then most adaptor toslink to Helix dsp. Should i keep volume on head unit around 70%, then control volume by Helix dsp? How to manage digital input? Is that control head unit volume is the same with control digital input? If i max out Head unit volume, does it mean i will max out digital input to dsp? What gonna happen doing it? Im a bit confuse.
Thanks.
 
Kb86 said:
Hi, thus what is the right way to keep it under control? Im using Carplay into head unit, then most adaptor toslink to Helix dsp. Should i keep volume on head unit around 70%, then control volume by Helix dsp? How to manage digital input? Is that control head unit volume is the same with control digital input? If i max out Head unit volume, does it mean i will max out digital input to dsp? What gonna happen doing it? Im a bit confuse.
Thanks.
Click to expand...

I have not used the equipment you have, so I'm not sure, but most digital devices can be run at any volume without problems.

My best guess if you should choose the volume knob you want to use, and then use it. If the system is not loud enough, then turn up the other volume knob. If the system is too loud or sounds bad, then turn down the other volume knob.

Your results may vary, I don't know how your equipment works.
 
Hi Jazzi, on your new V2 curve, what measurements method you use? i assume u waving mic, but around each ear or half circle around head?

Also, do you use that curve for global eq or each side?

Please be specific because its vary alot. Thank you very much.
 
Kb86 said:
Hi Jazzi, on your new V2 curve, what measurements method you use? i assume u waving mic, but around each ear or half circle around head?

Also, do you use that curve for global eq or each side?

Please be specific because its vary alot. Thank you very much.
Click to expand...

Hi!

I measured that curve using a 6-mic array. You can replicate the result with a moving mic average, without sitting in the seat. I will sometimes sit in the rear seat and reach up to the front seat to do the moving mic average.

I did not use that curve as a target. That was measured as the sum total of my entire system playing.
 
Thank you. it makes more sense now.
Could you explain further your tuning step?
1. Doing output EQ for each speaker?
How do you measure each speaker?
a. fix mix at headrest
b. moving mic while sitting in seat
c. mix array and averages?

2. Checking sum between speakers, how do you measure?
a. fix mix at headrest
b. moving mic while sitting in seat
c. mix array and averages?

The more specific the less confusing for us beginner. Thank you ��
 
Kb86 said:
Thank you. it makes more sense now.
Could you explain further your tuning step?
1. Doing output EQ for each speaker?
How do you measure each speaker?
a. fix mix at headrest
b. moving mic while sitting in seat
c. mix array and averages?

2. Checking sum between speakers, how do you measure?
a. fix mix at headrest
b. moving mic while sitting in seat
c. mix array and averages?

The more specific the less confusing for us beginner. Thank you ��
Click to expand...

Sure.

All measurements I do for frequency response are either moving mic method while sitting in the back seat, or using a 6-mic array when I was working at the office. They both measure very similarly. All measurements I do for time alignment or phase alignment is with a single microphone at the headrest, not moving.

My steps are:
-sanity checks, polarity, gain structure, and other setup steps
-equalize each speaker to the target, one at a time
-choose one speaker as my time alignment anchor (usually front left woofer)
-set time alignment and polarity of each speaker to match that anchor
-check speaker summing left and right in pairs (left and right woofers), then vs subwoofer
-play everything at the same time and check vs global target
-play bandwidth limited pink noise while sitting in the driver's seat
-adjust the relative left and right equalizers to center each of the pink noise tracks
-put the equipment away and play music I am familiar with
-make very small changes, if any

-sleep at least one night
-make small changes, if any
-sleep at least one night
-repeat
 
Justin Zazzi said:
Sure.

All measurements I do for frequency response are either moving mic method while sitting in the back seat, or using a 6-mic array when I was working at the office. They both measure very similarly. All measurements I do for time alignment or phase alignment is with a single microphone at the headrest, not moving.

My steps are:
-sanity checks, polarity, gain structure, and other setup steps
-equalize each speaker to the target, one at a time
-choose one speaker as my time alignment anchor (usually front left woofer)
-set time alignment and polarity of each speaker to match that anchor
-check speaker summing left and right in pairs (left and right woofers), then vs subwoofer
-play everything at the same time and check vs global target
-play bandwidth limited pink noise while sitting in the driver's seat
-adjust the relative left and right equalizers to center each of the pink noise tracks
-put the equipment away and play music I am familiar with
-make very small changes, if any

-sleep at least one night
-make small changes, if any
-sleep at least one night
-repeat
Click to expand...

Hi Jazzi,

- How do you set time alignment? Do you fine tune the timing after initial TA? If yes, how would you fine tune timing? Ear?
- Is there any case you give up correct timing to exchange for other benefit (for example phase?)

Please be details.

Thank you very much.
 
Kb86 said:
Hi Jazzi,

- How do you set time alignment? Do you fine tune the timing after initial TA? If yes, how would you fine tune timing? Ear?
- Is there any case you give up correct timing to exchange for other benefit (for example phase?)

Please be details.

Thank you very much.
Click to expand...

Phase and time go hand in hand. The only way to be in time but out of phase is if polarity of one is wrong. Otherwise to be in phase means you are also in time with the other speaker(s).

Looking at a graph and getting everything as good as possible via a graph is always a good beginning goal that is repeatable. This is a great first start.

Once you have the graphs pretty (rta, magnitude, phase, etc), then you can put those tools away that you used to make those graphs. You have achieved your baseline goal. Now it's time to sit in the car in your normal position and use your next tool, your ears. This is where the "flavoring" of the sound comes in. I first start with band limited 1/3 octave pink periodic noise and listen to the speaker interaction and if something is pulling left or right. I adjust accordingly by increasing one channel and decreasing the opposite by the same amount to keep overall tonality the same.

Once I have all that lined up I go to my test tracks that I know very well. I listen to them and do global eq (with my helix I use the input eq for this since my input is flat and doesn't need eq, but you can also use your head units eq here) to flavor the sound for the environment I am listening in. This is the key, every install/environment/equipment choice makes a difference in this flavoring. If you tuned two systems to the same pretty graphs and then listened, there is a very good chance they sound different.

I know Justin's process is pretty similar.
 
Kb86 said:
Hi Jazzi,

- How do you set time alignment? Do you fine tune the timing after initial TA? If yes, how would you fine tune timing? Ear?
- Is there any case you give up correct timing to exchange for other benefit (for example phase?)

Please be details.

Thank you very much.
Click to expand...

If I only have a tape measure, I use that to start with.
If I only have an RTA, I will use a technique called "nulling" where I flip the polarity of one speaker and then adjust time alignment so there is maximum cancellation between the two speakers (largest null in the RTA).
If I have a dual-channel FFT system like SMAART or OpenSoundMeter, then I use the phase response directly.

After I get close using the tools, I will fine tune by ear using the technique below. If I have tools then this process is much faster and I can match phase over a larger range of frequencies and use other corrections like all-pass filters which I find difficult to do by ear. If I have no tools available, then I use the technique below but it takes a lot longer.

To fine tune, I will set the lowpass filter to somewhere around 200hz and play only bass. I will play a mono track with a male voice like the introduction tracks on the Chesky Ultimate Demonstration Disc. I set the time delay for all speakers to the middle of the range on the equipment, like if the maximum delay per channel is 10.0ms then I set delay to 5.0ms on all speakers. I choose one speaker as the time anchor, usually front left woofer. Then I'll close my eyes and listen to a pair of speakers and adjust time delay until the mono sound appears to come from the "center" of the speakers. I will also check polarity by flipping the polarity of one speaker and ensure that it sounds more focused or not.

There are always cases when I give up "perfect" time alignment. The two most common are when a customer wants the sound to be more enveloping or surround-sound instead of precision focused in front. The other case is when I have a rear speaker that is very close to the listening position (like a rear speaker in an off-road vehicle) and the sound appears to be coming from that near speaker only. I will adjust the time delay of that nearest speaker slightly so that the sound stage moves away from that speaker and appears to be more balanced with the other speakers in the vehicle. A good example is the rear speaker in the Polaris Slingshot, where the rear speaker is directly behind your head about 6 inches away.
 
Justin Zazzi said:
There are always cases when I give up "perfect" time alignment. The two most common are when a customer wants the sound to be more enveloping or surround-sound instead of precision focused in front. .
Click to expand...

Hi, this is exactly what i am asking for.

Can you explain how adjust timing to be less perfect can make the sound more enveloping or surround like?

In my experiment, making midbass and midrange perfectly in time will lead to perfect sum at xover point (+6db) but crossover zone tend to be narrower.

Adjust timing to aim for wider crossover zone will make the sound blend better and more envelop?

Is that the case you talking here?

Would be great if you explain in technical perspective. This is very interesting.

Thank you.
 
Kb86 said:
Hi, this is exactly what i am asking for.

Can you explain how adjust timing to be less perfect can make the sound more enveloping or surround like?

In my experiment, making midbass and midrange perfectly in time will lead to perfect sum at xover point (+6db) but crossover zone tend to be narrower.

Adjust timing to aim for wider crossover zone will make the sound blend better and more envelop?

Is that the case you talking here?

Would be great if you explain in technical perspective. This is very interesting.

Thank you.
Click to expand...

I do not do this very tso it might be difficult to describe.

I always prefer to have good time/phase alignment in the lowest frequencies so that the system has more bass output. Everyone wants more bass output, and even if you don't want more bass output, then you can still have the benefit of less bass boosting. So when I say I sometimes give up on perfect time alignment, I mean perfect alignment in the not-subwoofer and not-midbass frequencies first.

I do not have a good scientific formula because it do not do this very often, but some things that we can try are changing the polrity and time delay of the tweeters, and if that doesn't get the effect then try adjusting polarity and timing of the midrange speakers. If you only have a woofer/tweeter 2-way, then try adjusting the time delay of the woofers very slightly to shift the phase of the midrange frequencies without disturbing the lower frequencies as much.

I would try to adjust the front speakers VS the rear speakers first, and if that doesn't work then try adjusting the left VS the right speakers. Sometimes polarity and timing of the rear speakers can make a huge difference in envelopment.

Sometimes it is also very simple, just make the rear speakers a little louder.
 
I don't know if this has been covered already, so here goes...

I have read from one seemingly knowledgeable source that:

A: Port compression begins to reduce output at port velocities as low as 22m/s

...whereas another seemingly knowledgeable source says:

B: Port compression begins to reduce output at 60m/s.

What say you?

Here is a link to A: https://www.diymobileaudio.com/threads/improved-port-area-calculator.349265/#post-4820809

Here is a link to B: https://www.diymobileaudio.com/threads/winisd-port-velocity.465823/#post-6241236
 
I love Triticum Agricolam's method of taking more variables into account. I don't remember where their formulas come from but the theory of it seems sound. I wish we could also take the woofer's motor strength as a variable too since woofers with higher BL^2/Re ratios should be able to create more force per watt which means more air compression at port tuning and higher port velocities.

daloudin's post mentions they run the simulations at xmax. At port tuning when velocity is maximum, the cone motion is minimum. If the simulation is run at tuning frequency AND the cone is simulated moving all the way to xmax, wouldn't this result in dramatically higher port velocities than we would actually see? Or is the author saying they use xmax at other frequencies like below tuning frequency to look for air movement in the port. I am slightly unclear about the conditions of the simulation and what the data is showing in this particular example. I like how the author mentions there are other variables like proximity the listener, if the port is firing towards the listener or buried behind other stuff, etc.

Overall there are many variables that can determine how to design and it looks like both authors are trying to account for these instead of a one-size-fits-all method. But how do you choose which of these variables are important to you? If you're an SPL competitor then the sound of chuffing or other distortions might not matter as much as power compression and pure output right? If you are a pure SQ enthusiast then you have different tradeoffs you're willing to make. If you're making a commercial product and you know consumers like to shop with their eyeballs and big numbers in a review article are impressive, then there is motivation to make different design choices again.

The shape of the port matters too. Port area is super nonlinear in that a "two inch" port in the shape of a tube will behave much different from one in the shape of a rectangle. This is a boundary layer thing where air closest to the inside walls of the port is moving slowly but air in the center of the port (away from the walls) is moving fastest. The same thing is why water along the edge of a river is calm and not moving very fast, but you can find fast flowing water near the center of the river. Then the terminations on either end influence how the port "sounds" and if we get chuffing or other undesirable distortions. So many factors at play here, difficult to make super simple rules that apply to everyone.

By happy accident, I have some COMSOL training this week. Maybe I'll look at this port tuning and air velocity thing and see what I can find. It seems like there are two questions: how to minimize the port sounding "bad" and how to get the most output from it.
 
I asked the question in hopes of gaining some insight as to whether or not it would be preferable in terms of output efficiency to increase port area beyond that which would be sufficient to prevent audible port noise, assuming very close proximity of port to listener (maximum potential for hearing any port noise).

Triticum's viewpoint seems to be that it would be preferable in terms of output efficiency to increase port area beyond that which would be sufficient to prevent audible port noise, particularly in the case of a non-flared port.

Daloudin's viewpoint seems to be that increased port area beyond that which would ordinarily prevent considerable port noise (certainly considerable port noise would be produced through a non-flared port at 60m/s, and perhaps even through an aeroport?) would yield no increase in output efficiency.

Although both of their points of view appear to be based on the premise that port compression = decreased output efficiency, they differ in terms of the rate of port velocity at which port compression causes a decrease in output efficiency
 
I’m trying to hook up my new double din aftermarket stereo to my 2015 Ford F250, I figured everything out but the steering wheel controls. I have what’s labeled on the bracket/harness, and the new stereo connected and steering wheel controls still do not work. The original stereo had two harnesses, could the steering wheel controls be located on the smaller 16 pin harness? I’ve scoured the Internet, and cannot find the answer to this, and finally found your thread here, hoping to gain some insight! Thanks for any help you guys can provide!
 
That info should be available with a wiring diagram. You can get 48hr (or something like that) access for under $20. Grab all the electrical diagrams you need and you should be able to find your info.
 
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