I got major problems with my soundcard while using measurement software. I am presuming the problem does lye with my soundcard, and some conflicting devices or even something connected with windows.
I use audiolense but I have same problem with REW. My input and output device don't support each other when I choose them.
I just want to go back to basics for now then delve deeper later.
The USB cable goes into the mic preamp(obviously) with the left and right mains(at the back of the preamp) going into the auxilary or the Preout 1 of the amplifier?
Have I missed something simple... Is the above correct?
Lastly, can it be that my soundcard is really not supporting it and that I purchased one that doesn't have enough features, this is highly unlikey as when I use my old Mobile Pre M Audio I have the same problem. Audio Box Itwo is my external soundcard.
It might be. You can always ask Presonus. You can get a Focusrite scarlett solo for 89 euros and that'll probably be current. As far as I know, some sound cards do depreciate.
All you do to connect an external interface is connect it via the data cable, in this case USB, to your computer, connect the main outs to your speakers - or mixer if you have one which is handling your physical routing - and a basic dynamic mic to one of your interface's preamps. This is all you need to verify everything is working.
Actually, I figured out how your meant to hook up the external soundcard. And to be honest its a little disappointing it took me 3 years to figure it out and it was due to a random forum post, nothing really explained it as well as this post. I know it sounds simple but I had no idea that the sound had to come from the external soundcard into the amp. I connected the soundcard to the amp with the RCA into the auxiliary of the amplifier with both the setting of the input and output device(on the software) on the soundcard not the amp as the input device.
Anyway, this isn't as clear as it should be, I might be clueless but even Ethans write up on Real traps isn't that clear. I had read it a least 50 to 60 times and it doesn't' spell out the above steps in a clear manner. Anyway, I have managed to get some measurement on my Audiolense software but on REW not yet. I am still experimenting and might have to come back to this forum to get it right. Plus I have a lot to learn about reading the graphs.
Attached is a before picture of my impulse response and a simulated response after applying correction to the measurement. Can anyone guide to what looking for? or how to read it in general?
I mentioned impulse response to see whether goat hair added more reflections in an iso box. Either way, the only time I've seen these be useful is when they're zoomed in to like a 50ms window. You can use this to see whether your RFZ is having the intended effect, the quality of diffusion later, or the effect of moving things around.
You can see an image at the center of my post herewhich illustrates a useful version of an impulse response.
Remember from above you mentioned "you can always ask Presonus" I did and this is the reply I got.
Tech Support April 11, 2019 17:27 Hi Masis,
Thank you for the update. The AudioBox is not designed for any loopback recordings. Yes there are applications that may work for you but we do not support third party applications. I can understand your frustration but the AudioBox is not designed for this function.
Regarding your questions, the AudioBox will record audio coming from an amp if that is the way you are configuring the setup.
I am just curious if the above gives us enough information to figure out if I can use my audiobox soundcard for the REW application or we need a soundcard that provides loopback that is if loopback is understood correctly in the above or by me for that matter.
PS> Presonus is the worst company I just have to say it I will never buy a product from them again.
Really, if you're just recording a mic input, there shouldn't be a problem. What's the difference between recording anything during a multitrack session and using REW? A loopback is when you plug the output back into the input.
In any case, either it will work or it won't. You can experiment with different Audio Systems other than ASIO like MME or even VoiceMeeter Banana to try to get a routing to work. I remember having some trouble with my old interface though it could be down to user error also.
That's about as much help as I can offer on this matter. If you want, you can also post screen shots and/or pics and video of your routing. Maybe there's a setting somewhere we can flip.
Okay, so from my understanding it seems that it should work fine. This time around I set the output through my amplifier/DAC with Direct Sound and recording with my external soundcard on ASIO. Seems like it worked and the measurement is descent enough. I will attempt with REW. Unfortunately, I need more experience with REW so it might take time.
Its seems i have this null between 40hz and 60hz I cant seem to shift it in anyway. I even added a large corner bass trap in one bottom corner(probably no enough) and made no difference. But adding more panels to the front with shifting the speakers closer to the front wall reduced the null(probably SBI) at around 120hz.
Im always reluctant to post graphs so I can force myself to do the research and I have thoroughly analysed some interesting technical analysis by you Hexpa and I will start a new thread which should help many others in understanding so many peculiarities of what you have said. But for now, can you please help me with this null between 40 and 60?
I just realised at the very least you need the room lengths. Length 17 foot by 13.6 foot and the height is 8.5 foot.
On the back right hand side there is a large 7 foot opening to the kitchen from about 10 foot from the front wall not ideally symmetric but the best I can do.
There is also a door right in the middle of the back which opens to another tiny room.
I should also note that the the speakers are not exactly in the middle and sit to the left of the room, with 1.3 foot from the left and about 2.9 foot from the right hand side. The speakers are specifically designed for wall placement.
I have about 25 traps most 4inch thick with a few 6 inch thick.
Anyway I hope that is enough info to get started....
Glad you got your rig to work. There are a couple things you can consider:
1. Do separate graphs: A. Full smoothed SPL, B. Unsmoothed SPL below 300Hz, C. Unsmoothed Waterfall below 300Hz with 300ms time window and showing at least 20dB of decay across the range. D. Unsmoothed decay plot with the same parameters as C. You don't have to post these but you do need to understand them if you want to optimize your space.
2. Below 63Hz doesn't matter as much though you most certainly have a modal issue. I find that SPL problems tend to be due to positioning. Adjust your listening position and/or speakers.
3. Though the first two octaves aren't our primary concern, you have tons of energy there. If you're using a sub, turn it down. Also consider moving it as per point 2.
4. Nulls above 100Hz can be very small things. For example, I had a null like that which I fixed by lessening the gap of my cloud absorber. Another null was due to some stuff in the front of my room. I've since rearranged many things in my room so I might have some more peaks and nulls but it's up to you how perfect you want to be.
If you're not planning on building more treatment then you need the graphs I suggest to analyze then test different placements. Make note of each problem area that doesn't fit your targets and see if you can't make several dB improvement there and there in your SPL and even and shorten your decay by a few ms.
Remember that my targets are +-10dB SPL from 63Hz-Schroeder Frequency. Above that, I just smooth the response. Decay target is 20dB of decay within 150ms and as even as possible. Always try to work with physical positioning before applying corrective EQ.
Your reply is greatly appreciated, I will work on it so if I have disappeared for some time rest assured your input is being put to use. Wealth of information thanks!
But like I said there was two other posts that really catapulted my knowledge forward, I think one was "A notch at 80-90" the other I can't remember. Anyway, after analysing the threads, I came to two sticking points which need clarification. This can hopefully also, with the your last post untangle many things for me.
"The last thing I'll say is that rockwool is definitely an option for super chunks or gapping. For whatever reason, maybe you're in a room with 7' and 14' dimensions and have a very specific need to tame 80Hz, the best thing you can do is use either a 3.5'-thick fluffy absorber or a 1.75' fluffy absorber with a 1x gap. You don't want to make a 6" rigid panel and gap it 3'. While that will target 80Hz, it isn't a best practice."
You don't want to make a 6" rigid panel and gap it 3'. While that will target 80Hz, it isn't a best practice."
Why isn't this the best practise? And your recommending that a fluffy corner bass trap is more efficient than a 4 or 6inch rigid trap?
"the quarter wavelength is the ideal place to put an absorber. That doesn't mean there needs to be a quarter-wavelength gap, mind you. All it means is the 1/4 and 3/4 parts of a wave are it's phases with maximum velocity and minimum pressure - perfect for velocity/porous/frictional absorbers. I included an image in this video and blog post I made contrasting this type with resonant absorbers."
OVerall this pargraph needs clarifying as I do not understand "All it means is the 1/4 and 3/4 parts of a wave are it's phases with maximum velocity and minimum pressure "
If that is above me, then I would like to know if I target around 60hz with a gap of around 1.5 metres does that also help absorb everything above 60hz or am I only targeting the 60hz?
A full-sized fluffy chunk is almost certainly better than a rigid panel. You'll have to test it yourself though. All I go on is the massive difference installing chunks gave me. Their main disadvantage is that they aren't as versatile i.e. they stay in corners.
If you gap a thin panel more than 1x, you get a 'hole' in its effectiveness similar to how a high pass filter can have a resonant peak but above that you might put a negative bell for clarity. The best thing you can do to 'target a frequency' is fill the entire 1/4 wavelength. Yes, this absorber will target everything above it but also have an effect below - just as any absorber would.