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Discussion Starter #61
So this is another how awesome rephrase is...

My sub amp has a dial crossover that cant be turned off.
RF bd1000a1 it has a dial from 50-250hz BW24

So I made a 269hz LR4 linearization than in phase eq was able to make phase flat
than added my LR4 linearization on to it. I sent that to the sub channel FIR and the FIR that does EQ is on the opendrc global EQ. Two separate fir.

Now the sub is perfect. I turned the knob on amp crossover up all the way to 250 added the linearization I custom made for the BW filter







 

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Discussion Starter #62
Got the sub dialed and styled

This week I’m doing this again 3rd attempt , this time I think I’m getting the hang of it.

The only part that’s still sucking for me is left and right coherence and the minimum phase filters on left and right de-correlation


I have questions out now to some experts to get a bit more info
I was just told today my soundcard I’ve been using for this might be erroneously spinning my phase backwards on some measurements. I’m hoping it was just the 2nd sub crossover causing all the headache, I have to work through Black Friday installs this weekend so won’t get to it till next week.

I’m bringing home my DMRTA from work this weekend and I’m starting all over again. That might be my only issue this whole time is the creative usb sound card (which is extremely flat response) I think it’s still fine but I’ll bring home the audio control sound card and bring out my big guns.

If all goes well , the link below is the procedure to use that is better than Dirac if you have the time. Even without doing seperate left and right eq as of now and just going off what phase I have been able to measure with the small sound-card it’s still better than Dirac: so we’ll see. I’ll post up every measurement and a entire ride along with the dmrta

Till then

https://www.dropbox.com/s/10xdhh83jokzbxv/REW_rePhase_tuto.pdf?dl=0
 

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Discussion Starter #63
cant wait till I can try this. Especially with the amazing help I’ve been getting on the DIY side from fluid.

I have a brand new set of beyma 8G40-4s arriving tomorrow. The 2118h is coming out and I’m also going to build up my kicks to be a lot thicker fiberglass and a bigger vent to the outside. I’m also going to eliminate my vent holes in the kicks and try to get the response more uniform and get rid of some of the back-wave interference

So I have a busy Black Friday weekend at work, next week the beymas go in and I start making my own midrange corrections.


I have an argument out now on comb filters and time delay. We’ll see what comes back and see what the rephase experts say about unequal path lengths.
I’m hoping for a easy to adapt procedure to the one I posted above.

We’ll see :)
 

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Discussion Starter #65 (Edited)
What didn't the legendary 2118 not do for you?
Short answer , 2118h plays to 70hz in .5 but don’t even think about anything more than 40w

What didn't the legendary 2118 not do for you?

The 2118h has been a fabulous midrange, in fact I can already say with confidence the 2118 will outperform the G40 above 800hz and have better midrange. I can’t think of an 8” that is as good as the jbl in the midrange

It’s been proven that the 2118h “breaks” at 300hz . Meaning it starts to drop off because it can’t act pistonic below 300 while keeping the same linearity as it’s upper band.

It makes no sense to eq down that magical midrange down to the midbass output when what little midbass can be made from the 2118h is riddled with distortion.

JBL states that it “shall” play down to 70hz. Which it can in a .7cuft bass reflex tuned around 45. the natural resonance gets lowered a tiny bit in that type of enclosure which also the port will start to control cone not allowing it to move as much as it would in IB. So that’s where they get it’s good down to 70.

The 2118 has a extremely thin diaphragm and a stiff compliance. It’s not mean to to move that much. The distortion in IB goes way way up under 300hz IB.


So I’m trading exceptional midrange for a little more bottom end performance. I have the 2118h in a a-periodic kickpanel pod that I have stuffed with fiberglass AP membrane that gets the jbl down to 200 using LR8 and that’s the bleeding edge for my kicks in AP. The kicks have a nasty ringing at 111hz very high q that has been difficult to tame.

I’m changing the kicks to IB and venting to the outside and want them to play down to 150. (80 would be great but it’s not possible while maintaining high enough efficiency for my goals) so the 8G40 is good down to 125hz and it has superb midrange under 1khz. The 8G40 has the best lower end vocals I’ve ever heard out of a speaker. I kicked myself several times for selling it. Now I have another brand new set from the 2014 2015 run of them. (The ones that has 6mm xmax vs the new “improved” that has 4.5mm).

Ive been looking for awhile now for a new old pair.

The 8G40 is also in 4ohms and has more motor than the jbl. The G40 is a amazing sounding loudspeaker. The 10G40 and the 6G40 don’t quite capture the same type of “big vocals” as the 8”. It’s quite a good speaker for what I am going to be changing into.

If I can get down to 150 and still be able to use a 4th order filter I’ll be estatic but I doubt I’ll be so lucky. I think it’s great potential but I don’t think I’ll get much below 150, maybe 125hz with 8th order.

Only one way to find out :)
 

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Discussion Starter #66
So when dealing with fir filters you have a finite amount of time windowing to address phase delay and group delay.

With rephase you can invert the time and invert the polarity separately
This can be useful especially with unequal path lengths.

So I would like to talk about something Dirac does quite well and will be key into following along with what’s what we’re going to be doing in coming days.

So Dirac can take a 2ch measurement and each of the channels could have substantial phase delay or group delay in the measurement and it can still correct for it very nice. You can have speakers with incorrect delay settings. Example the left tweeter could be .5ms out of arrival from the mid and the midbass could be 2ms out of time with the mid. That would show up as a bunch of wraps and. Much more than 180deg of shift.

The ddrc22d is limited to 23ms of output delay where it’s counterpart the opendrc 2x2 has 64ms of output delay at 6144t per channel. So the ddrc22d must be using iir type custom order allpass filters to deal with the phase delay.

The group delay and phase delay could be considered the same thing, phase delay I am speaking of is any delay that is in the measurement except time of flight delay.

I would like to share two videos on this subject so we all know how to read delay on a logarithmic time to frequency chart. This gives highlights on how to add or subtract time out of a small part of a measurement and repair it within rephase

Rephase allows you to export iir biquads to a minidsp file and use the rest of the correction in fir. It can split up the coefficients to iir and fir outputs on a single correction. Weather you use a mixed phase approach or seperate fir and iir biquads to do the time correction and get each channel to be doing the same thing.

Rephase has 16banks of eq and 7 or 8 banks (IIRC) of iir curves for all pass types. Pay attention to these and how to do the calculations within it. The calculations can greatly help figure out weather it’s better to keep a few wraps in the measurement as it might simply be group delay


Part 1


https://youtu.be/_KZ1_Je9fA8


Part 2

https://youtu.be/IcYBP51eCMk
 

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..
I have questions out now to some experts to get a bit more info
I was just told today my soundcard I’ve been using for this might be erroneously spinning my phase backwards on some measurements...
...
Phase and freq and related to each other by on integral and a derivative...

So play a tone, like 440 Hz... a "phase spinning backwards", will make it 439 etc.

Basically... trust your gauges, as the sound card is probably right!
 

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Discussion Starter #68
Phase and freq and related to each other by on integral and a derivative...

So play a tone, like 440 Hz... a "phase spinning backwards", will make it 439 etc.

Basically... trust your gauges, as the sound card is probably right!
Lol it is right .

It was my amp whole time. So I’m looking for new bass amp ....probably going with matching alpine type X the XA90M

Yeah very strange , even after the crossover linearization it still had super goofy polars
 

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Oabeieo you're a wealth of information and I really appreciate your sharing your experiences with us. I just stumbled on this thread right after installing a CDSP 8x12 DL. I've been following the CDSP 8x12DL threads, and in the last couple days I've been trying to figure out a workaround for the phase shift on the non-defeatable 10hz SSF and variable LPF on my sub amp. Funny to see you're kind of dealing with that right now too. It sounds like the best solution really might be to use a different amp with no x-overs.

The Dirac Live version still has appeal to me as although I'm pretty familiar with the basics of x-overs, slopes and time alignment, I start to lose it at phase and transfer function. But I have to wonder if it would make any sense to try and add one or two of these OpenDRC-DI's to my setup. It seems like you could do a bunch of tuning on the sub in an OpenDRC-DI, then (because you're supposed to apply all filters and eq applied to the sub to the entire front stage as well) forward that tuned signal via SPDIF to the CDSP inputs and allow it to run DL on the midbass, midrange and tweeter, or maybe even the sub again as well.

This may seem a silly question, and please pardon my ignorance, but does forwarding the tuned signal to the CDSP as its main input signal solve the need to apply all filters and EQ from the sub to the mains (as far as phase etc is concerned)? Or do you actually need to send the original signal again to the mains, and actively apply all EQ and filters that were applied to the sub to the each individual channel in the mains?

And one step further, could you work on both the sub and the midbass in a single OpenDRC if it was just part of the chain before a CDSP? Or would you need an OpenDRC for the sub and another OpenDRC for the midbass?
 

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Discussion Starter #70 (Edited)
Oabeieo you're a wealth of information and I really appreciate your sharing your experiences with us. I just stumbled on this thread right after installing a CDSP 8x12 DL. I've been following the CDSP 8x12DL threads, and in the last couple days I've been trying to figure out a workaround for the phase shift on the non-defeatable 10hz SSF and variable LPF on my sub amp. Funny to see you're kind of dealing with that right now too. It sounds like the best solution really might be to use a different amp with no x-overs.

The Dirac Live version still has appeal to me as although I'm pretty familiar with the basics of x-overs, slopes and time alignment, I start to lose it at phase and transfer function. But I have to wonder if it would make any sense to try and add one or two of these OpenDRC-DI's to my setup. It seems like you could do a bunch of tuning on the sub in an OpenDRC-DI, then (because you're supposed to apply all filters and eq applied to the sub to the entire front stage as well) forward that tuned signal via SPDIF to the CDSP inputs and allow it to run DL on the midbass, midrange and tweeter, or maybe even the sub again as well.

This may seem a silly question, and please pardon my ignorance, but does forwarding the tuned signal to the CDSP as its main input signal solve the need to apply all filters and EQ from the sub to the mains (as far as phase etc is concerned)? Or do you actually need to send the original signal again to the mains, and actively apply all EQ and filters that were applied to the sub to the each individual channel in the mains?

And one step further, could you work on both the sub and the midbass in a single OpenDRC if it was just part of the chain before a CDSP? Or would you need an OpenDRC for the sub and another OpenDRC for the midbass?

Thanks zacjones for the kind words, excellent questions also.



So an opendrc would definitely be able to untwist that phase up to 64ms with the 48k plugin. 10hz is 100ms so you would be able to do most of it. Most of it is inaudible however the air between 10hz and 20hz really helps the entire spectrum get in tune. It’s crazy how much the sub keeps the system together. It’s easily the easiest driver to get right and the most important speaker in the system (if I had to pick one and label it as such) I would go for a different amp if possible and use the dsp on other things.



If you can add the same exact subsonic filter to all other channels without having to buy anything else and without massive dsp losses that would solve the issue as well.



A pair of opendrcs would be quite the system. So far my favorite configuration has been my opendrc upstream of 3 other dsps and one of the 3 being a ddrc24 Dirac that is only for midrange. It is excellent sounding.



To be completely honest after studying what Dirac is doing, it’s very hard to beat the algorithm. A computer in charge of the biquads with a missive amount of programming and fine tuning to the algorithm is hard to top when staring at the controls of rephase and some of the answers you need just can’t be thought up as fast. If you like tuning and figuring out how’s and why’s rephase is excellent and does produce a better sounding tune.



So loudspeaker crossover linearization is by far the most beneficial thing a system can have and the removal of any group delay. That’s half of the room correction right there.

A pair of opendrc might be two much dsp for 4 potential channels. But if you used one as a sub dsp and the other as a highs dsp that would work excellent as a pair of global dsps that work on everything. So a few ways to do it. I would like to make a suggestion though that maybe will help.



Focus on speaker linearization first, than do the room correction. Getting the crossovers to work together perfectly is by far the biggest improvement I’ve had above all else as far as time domain goes. So much so when I was on Dirac I opted to use 3 2x4hds with a ddrc22d upstream because of linearized crossovers. Instead of using the all in one 8x12DL that does so much except driver linearization.

I haven’t had a need to cross below 80hz and you simply don’t need to. The 2048taps at 96k works just fine for crossovers and you also have 4ch of fir power per unit. Minidsp makes a 8ch version that has even less taps so a pair of 2x4hds is plenty, or a 3rd for the sub if you want to take it that much farther but it’s really not at all needed. I only have a third because of my rears.



So having fir channels at every output is the very first thing to acquire. You really can’t do a linearization on a gloabal dsp. It needs to be done at the driver level. And here’s the thing.



Being that hear the sum of all reflections and all the speakers playing there’s a lot going on and it needs to be broken down and sorted out to get a proper linearization. With Dirac and the room correction side of a correction your correcting for the sum of all frequency and time related problems. So in that instance you can linearize the sum of all speaker and reflections (room correction). At the driver level you linearize each individual sound source before they sum. That’s where things make the biggest difference and makes dealing with the room a lot easier.



When we’re talking about the sum of multiple drivers the crossover time distortions would be viewed as a group delay and not a linearization. So adjustments to the time domain in a global output would affect the sum of everything and should be measured as such.



When you correct the group delay in the time domain gloabally without doing any loudspeaker linearizations first you can’t truly linearize the crossovers only the sum. The sound quality is still massively improved vs doing nothing at all. In a global correction if you for example fix the group delay of a high pass you making the low pass time smear worse as a low pass and high pass go opposite direction (except a few even order filters in that case the polarity is reversed on the low pass). If the crossovers sum perfect and let’s say your car breaks all the problems of a car and the Linkwitz-Riley alignment actually works the way it should in a home setup it than it would be okay. The problem is in a car, the alignment topologies don’t quite operate to the textbook they were as much written. They still perform mostly as textbook however the farther the drivers are mounted away from each other and the more path length issues are presented the alignments perform worse and worse, or shall I say different and unpredictable, because it may work out based on luck. So while you can still get superior tonality and sound with a single global correction , driver linearization is where the cake is baked and the global correction after is the icing.



Doing separate linearizations you can get them to sum perfectly no matter how you have them mounted or what axis there on or the path length differences. All you do is make the driver flat for an octave past crossover at listening position and turn on your textbook linear phase crossovers in a nutshell. There’s more you can do of course but doing just that is 75% of the time domain distortion.



Once the driver linearization is done you go to do your global correction. You can count of the the speakers summing right so no worries about crossover cancellations as a result of any global correction. Than you measure each side and sum the measurements. Do minimum phase eq until the sum of L and R are flat. Than do separate left and right measurements and look at excess group delay and make it minimum phase. No de-correlation.



Using separate eq settings on left and right is the worst thing one can do because of unequal path length until I prove otherwise or someone can prove otherwise. I’m still working on that part of it and have been using linear phase eq after doing summed L and R eq and have been having some good luck in the higher frequencies. Pre-ringing is audible in the lower frequencies with linear phase eq.
It works excellent in stop bands or at the filter knee. I can’t hear pre-ring at high frequencies. Linear phase eq also helps shape the DI or power response. More on that later.

Hope that helps
 

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Hope that helps
It absolutely helps. Thanks for the detailed explanation. Well for starters I guess I will go ahead and ditch the Zapco Z-3KD with non-defeatable variable SSF and LPF. Why they made a top tier amp without a defeatable filter switch I cannot imagine.

I will likely run the C-DSP 8x12 DL for a while to get a feel for it and see if I feel the need to make improvements. The concept of linearizing each driver individually at the listening position sounds pretty amazing. It sounds like this could be most easily accomplished by simply adding a couple 2x4HD's into the end of the mix. The only downsides are the additional A/D and D/A conversions required, and with the C-DSP we're still stuck on Dirac 1.7.

Will a single 2x4HD have enough taps to process a sub and a pair of midbass drivers?

Maybe a great place to start would be with adding a single 2x4HD for the sub and midbass drivers. That way I'm not overly concerned with the additional A/D and D/A conversions since we're only talking about 20-300hz of audible frequencies. We're still going all digital through the midrange and tweeter, and I can let Dirac handle those for now. If I find I'm able to handle tuning with the single 2x4HD and want to try my hand on the mid/tweet, I can always add another at any time on the mid/tweet, and see if the benefits of driver linearization outweigh the additional A/D and D/A conversions.

Anyway I hope I haven't strayed too far off topic here. When I actually get knee deep into rePhase I'll be back. Thanks again!
 

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Discussion Starter #72
It absolutely helps. Thanks for the detailed explanation. Well for starters I guess I will go ahead and ditch the Zapco Z-3KD with non-defeatable variable SSF and LPF. Why they made a top tier amp without a defeatable filter switch I cannot imagine.

I will likely run the C-DSP 8x12 DL for a while to get a feel for it and see if I feel the need to make improvements. The concept of linearizing each driver individually at the listening position sounds pretty amazing. It sounds like this could be most easily accomplished by simply adding a couple 2x4HD's into the end of the mix. The only downsides are the additional A/D and D/A conversions required, and with the C-DSP we're still stuck on Dirac 1.7.

Will a single 2x4HD have enough taps to process a sub and a pair of midbass drivers?

Maybe a great place to start would be with adding a single 2x4HD for the sub and midbass drivers. That way I'm not overly concerned with the additional A/D and D/A conversions since we're only talking about 20-300hz of audible frequencies. We're still going all digital through the midrange and tweeter, and I can let Dirac handle those for now. If I find I'm able to handle tuning with the single 2x4HD and want to try my hand on the mid/tweet, I can always add another at any time on the mid/tweet, and see if the benefits of driver linearization outweigh the additional A/D and D/A conversions.

Anyway I hope I haven't strayed too far off topic here. When I actually get knee deep into rePhase I'll be back. Thanks again!
“Only 1.7” .....oh man any Dirac is downright amazing and the 8x12DL is one powerhouse that is just downright badass. 1.7 is a amazing platform.

An 8x12DL will sound impeccable. Remember the crossover distortion is not very much audible. It’s still audible, but it’s small. And if the sum (what you actually hear) is linear (phase and frequency) than honestly what difference does it make. The differences are tiny. If you like to chase perfection than do 3 2x4hds with a ddrc22d upstream. I’m trying to beat that setup right now manually and being honest I’m not sure if I’ll ever get there it’s that dam good.
And I only say that because I measured a perfect Dirac impulse on both left and right channels. So it’s hard to beat perfect. I think I just want to learn how to get there on my own all the way.

A 2x4hd fir bank is good for a crossover linearization and not really much more than that. Maybe some small eq in the fir but it falls apart fast with much more than a crossover. It’s really obvious they designed it to be a linear crossover. Just not enough taps for anything else. It work great for a crossover though. And it’s a great dsp on top of that. At low frequencies you can center the impulse ahead a few ms and get a low frequency crossover to work with the 1024taps. One you add eq on top of it it just isn’t powerful enough as eq would require different centering in most instances if your trying to eq out of band.

If you have a digital source honestly a ddrc22d and a pair of 2x4hds is downright amazing hands down. Otherwise , the 8x12DL if your only wanting Dirac.

I would avoid the ad/da da/ad nonsense.
The quantization errors is audible as it’s all floating point. If it was fixed point it wouldn’t be such a big issue at 24/32bitAd but with floating point I can hear the difference. It’s grainy and looses depth. At least not over this, you want the signal to be right and stay digital with this much dsp being done.
 

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It's good to see you're still at this Andy. I've been enjoying my system for the last couple months and taking a break. I need you to get your hands on a Bit One HD and give me your opinion. :)

I really like the fact that I could choose any crossover slope and retain phase correlation between the active speakers. An example of how I was able to use this to my advantage was that I had an issue with 250-300hz where there was excess group delay due to car interior and speaker mounting position. It caused 250-300hz to steer to the right excessively. I could never get low mids to sit in the center. Even when I tried to EQ it out it just didn't work. The left and right amplitude were match but there was excess group delay on the right speaker which EQ couldn't fix. I used a 48/db slope and crossed mid+midbass over at 300hz and the issue went away completely. Solid center image for male vocals. I basically resorted to having my midbass playing 250-300hz by using a very steep slope while maintaining phase correlation between the active speakers. I totally agree on what you're discovering because I think I noticed that as well when I'm using the Bit One HD with FIR filters.

Your ears are keen to be hearing A/D conversions haha. But that's what makes you good at this. Every time you pump a signal through a device you're degrading or coloring it in some ways. Less is more there.

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Discussion Starter #74
Thanks Tony

yeab I would love to hear your car I bet it’s amazing. My buddy Evan has the n7 3 ways in in Ram with pods a lot like yours and it’s very nice to listen to his car with a helix and brax amps. Very different than horns and 8s but good. Dam good actually. (Just can’t get loud without blowing a 500$ 3” speaker which is what sucks) aside from that man oh man the sq is insane.

But yeah it’s a lot of fun.

I’ve finally got a few good base tunes going.
I finally figured out a MASSIVE problem I have been having and completely unaware

My new laptop had installed windows drivers and made my soundcard run at 96k and I’m taking 48k measurements.
Also rew says if using a usb mic a soundcard calibration isn’t necessary.

Only after digging on the rew forums I’ve found where John has stated it is still a good idea to calibrate it for phase related measurements, so I did and everything is working much much much better.

It was off at 10k so it moved the phase only a tiny amount, but that tiny amount up high is a substantial amount down low when your alignment of things is sample by sample.

Gang, make sure your soundcard has a loop back, is calibrated,and is running at the right sample rate in windows and in rew.

I metered my web configuration so it can’t update anymore and hopefully no more drivers update and change themselves.

I can’t express how important it is to make sure your rig is calibrated and has the right sampling within entered. Also make sure your RTA is using a rectangular window ONLY if using PN noise and the fft lengths match! So so incredibly important
 

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Hmm might invest in a USB sound card for the MacBook. I don't think I could get loopback with stock sound card.


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Focusrite 2i2 is a good microphone preamp if one wants to run an external microphone. And it has 2 inputs and 2 outputs so a loopback can be setup.
 

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Discussion Starter #77
Focusrite 2i2 is a good microphone preamp if one wants to run an external microphone. And it has 2 inputs and 2 outputs so a loopback can be setup.
I was just lookin at that haha

I have a old creative sb! Soundcard that has a nice flat responce and all kinds of in/outs
It was 50$ and has worked great , I have the dmrta that I can’t seem to get asio to work right in rew. So I might get one, but so far the sb! is working again now that I figured out what was wrong.

If I can’t get the dmrta to work with rew I think that or the us-366 tascam will have To be the next thing
 

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Discussion Starter #78
Hmm might invest in a USB sound card for the MacBook. I don't think I could get loopback with stock sound card.


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Do you have a line in? Like a aux in or mic in ? If so you can loopback , you have to just configure it .....in a Mac......Idk .

The loop back is the only reliable way I’ve been getting good phase measurements
And when doing more than one speaker it’s critical. The acoustic timing reference works on tweeters pretty worthless on anything else.

Using loopback I can take measurements and they actually do what it says. No goofy sounding corrections.

There’s nothing worse than taking a bunch of measurements just for them to trick you into thinking there accurate and than not accurate at all. The phase is a dead giveaway something is wrong. The FR can completely send you on a wild goose chase.

Even tho my soundcard plays flat adding the calibration made my phase spot on again.
It was riddled with problems for like 3 weeks to a month before I caught it. Such a headache going from excellent sounding tunes to what the f*** happened.
 

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Do you have a line in? Like a aux in or mic in ? If so you can loopback , you have to just configure it .....in a Mac......Idk .

The loop back is the only reliable way I’ve been getting good phase measurements
And when doing more than one speaker it’s critical. The acoustic timing reference works on tweeters pretty worthless on anything else.

Using loopback I can take measurements and they actually do what it says. No goofy sounding corrections.

There’s nothing worse than taking a bunch of measurements just for them to trick you into thinking there accurate and than not accurate at all. The phase is a dead giveaway something is wrong. The FR can completely send you on a wild goose chase.

Even tho my soundcard plays flat adding the calibration made my phase spot on again.
It was riddled with problems for like 3 weeks to a month before I caught it. Such a headache going from excellent sounding tunes to what the f*** happened.
MacBook only has a headphone output. No line in.


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