REW measurements done. Please help interpret. Amazing? Horrible?

[Snoochers]

Again, fully agree with this important point.

In any technical/scientific study/test, it is critical/indispensable that you (we) do not change multiple parameters at once; change only single parameter, and then compare "before and after".

If you change e.g. two parameters at once, the pros and cons would happen at once to give no improvement or even worse effect or overlapped improvements; in this situation you cannot identify/assess the pros/cons of each of the two parameter changes.

Yes I know but this is partly a moot point since I'm using different software AND microphones to measure ARC vs REW. I'm measuring from the same place from both. I have used the ARC microphone in REW for what it's worth and the graph is roughly the same.

 

[dualazmak]

Yes I know but this is partly a moot point since I'm using different software AND microphones to measure ARC vs REW. I'm measuring from the same place from both. I have used the ARC microphone in REW for what it's worth and the graph is roughly the same.

If you would like to be more technically/scientifically strict in your approaches, you need to establish independent "validation" methods before blindly using/trusting advanced software tool(s).

This is why I use "recorded rich white-noise FFT averaging" (ref. here) for Fq-SPL measurements, and my own "time-shifted tone burst pulse sequence" plus "exact sine-wave shape matching" for time-alignment measurements/tunings (ref. here and here).

You (we) should note that advanced automatic software tool(s) do not always give proper/correct information/results for improvement; just for example, as you can find here and here, @zergxia carefully traced my methods in his audio system using the test tone signals I prepared, and he found/shared that EqualizerAPO does not always give him correct/right tuning information in terms of time alignment over the SP drivers.

 

[Snoochers]

If you would like to be more technically/scientifically strict in your approaches, you need to establish independent "validation" methods before blindly using/trusting advanced software tool(s).

This is why I use "recorded rich white-noise FFT averaging" (ref. here) for Fq-SPL measurements, and my own "time-shifted tone burst pulse sequence" plus "exact sine-wave shape matching" for time-alignment measurements/tunings (ref. here and here).

You (we) should note that advanced automatic software tool(s) do not always give proper/correct information/results for improvement; just for example, as you can find here and here, @zergxia carefully traced my methods in his audio system using the test tone signals I prepared, and he found/shared that EqualizerAPO does not always give him correct/right tuning information in terms of time alignment over the SP drivers.

I’m already using REW with a calibrated microphone with the mic moving method. are you saying that’s not good enough? That I need to take better measurements? I’m not saying my measurements are perfect, but I feel like they’re probably accurate enough to diagnose an issue here. Or could they be so far off as they are?

 

[dualazmak]

are you saying that’s not good enough?

Maybe so, or may not be so...

Since you already have calibrated microphone, you can easily apply "recorded rich white-noise FFT averaging" (ref. here) for Fq-SPL measurements, and you can compare the results to those given by REW method(s).

And the point given by @Keith_W on mic stand/tripod would be also important in terms of possible artifact(s) on the measurements.

 

[boxerfan88]

I don’t know what you’re talking about. Is this something in ARC? I’m not familiar with that setting

Now I realised you’re using ARC automated room correction, not REW room correction.

My suggestion is more easily implemented using REW/manual PEQ adjustment.

Sorry for the bother, kindly ignore my post.

 

[Keith_W]

I’m already using REW with a calibrated microphone with the mic moving method. are you saying that’s not good enough? That I need to take better measurements? I’m not saying my measurements are perfect, but I feel like they’re probably accurate enough to diagnose an issue here. Or could they be so far off as they are?

You can use REW with a MMM. You just can not compare that result to the ARC measurements. Different measurement method, different microphone.

 

[Snoochers]

You can use REW with a MMM. You just can not compare that result to the ARC measurements. Different measurement method, different microphone.

I'm not comparing them to the arc measurement. I'm just looking at the measurement post ARC calibration. I'm not looking at the ARC curves at all. They look perfect lol, and I know they're not supposed to be.

 

[Snoochers]

I have made a lot of progress by using Equalizer APO. I have a question and hope someone has the answer. I saw this on Reddit:

"Keep in mind that if you use an AV receiver, there is a good chance you are bypassing EqualizerAPO anyway. It hooks into Windows Audio API, which is bypassed when bit-streaming audio directly over HDMI or SPDIF."

Is this true ? If so I am screwed because most of what I watch is TrueHD Atmos over HDMI etc.

 

[Keith_W]

It is not necessarily true. And if this is what is happening, you might be able to fix it.

Firstly, some background information. There are 3 standards we commonly use for audio in modern versions of Windows. WASAPI shared, WASAPI exclusive, and ASIO. WASAPI shared is the default sound device and uses the Windows mixer. This means the audio may be resampled, but worst of all, all system sounds share the same device. Meaning you might get notification dings while listening to music. DO NOT USE WASAPI SHARED.

WASAPI Exclusive means that one program has exclusive use of the audio device. It bypasses the Windows mixer. ASIO does the same thing, but ASIO is a third party standard and is the preferred choice of pro audio (and should be your choice if you want audio on Windows). Like WASAPI Exclusive, it gives access to the sound device to one program only. It also has lower latency. The downside is that your DAC or AVR manufacturer has to provide an ASIO driver.

Older Windows standards such as WDM or DirectSound are run as emulation layers in WASAPI Shared.

The first thing you need to do is find out how your AVR is connected to your player software. On your player software, look for the default sound output. If you don't see one, or if it says "Default", it is likely going to WASAPI shared. Solution: set it to output to EqualizerAPO. If it does not let you change the output sound device, use another software.

In EqualizerAPO, set the output to your AVR. I don't use EqAPO so I can't tell you how to do this.

By far the easiest way to check is to go to EqAPO and put in some crazy wonky configuration (e.g. cut all the sound above 100Hz) and see if you can hear a difference. If you hear only bass, then you can be assured that EqAPO is in your signal chain.

 

[Snoochers]

It is not necessarily true. And if this is what is happening, you might be able to fix it.

Firstly, some background information. There are 3 standards we commonly use for audio in modern versions of Windows. WASAPI shared, WASAPI exclusive, and ASIO. WASAPI shared is the default sound device and uses the Windows mixer. This means the audio may be resampled, but worst of all, all system sounds share the same device. Meaning you might get notification dings while listening to music. DO NOT USE WASAPI SHARED.

WASAPI Exclusive means that one program has exclusive use of the audio device. It bypasses the Windows mixer. ASIO does the same thing, but ASIO is a third party standard and is the preferred choice of pro audio (and should be your choice if you want audio on Windows). Like WASAPI Exclusive, it gives access to the sound device to one program only. It also has lower latency. The downside is that your DAC or AVR manufacturer has to provide an ASIO driver.

Older Windows standards such as WDM or DirectSound are run as emulation layers in WASAPI Shared.

The first thing you need to do is find out how your AVR is connected to your player software. On your player software, look for the default sound output. If you don't see one, or if it says "Default", it is likely going to WASAPI shared. Solution: set it to output to EqualizerAPO. If it does not let you change the output sound device, use another software.

In EqualizerAPO, set the output to your AVR. I don't use EqAPO so I can't tell you how to do this.

By far the easiest way to check is to go to EqAPO and put in some crazy wonky configuration (e.g. cut all the sound above 100Hz) and see if you can hear a difference. If you hear only bass, then you can be assured that EqAPO is in your signal chain.

Thank you for this ! I have MPC-HC and in the audio renderer dropdown menu I don't see EPO. I do see my receiver and I put directsound to that. EPO is not working with MPC-HC because I am lowering by 20db and it isn't doing anything. I thought it was system wide? What happens if I use Netflix or something?

EDIT: It works out of the box with Spotify, but not Netflix or MPC-HC. How do I make these do ASIO? I also game all the time and change games every week. Am I going to have to fiddle all the time?

 

[Keith_W]

I am far from an EqAPO expert because I do not use it. I suggest you find an EqAPO thread on ASR, or find out if they have a support forum, and ask there.

You will need to provide much more detail than you are providing here.

 

[Snoochers]

I am far from an EqAPO expert because I do not use it. I suggest you find an EqAPO thread on ASR, or find out if they have a support forum, and ask there.

You will need to provide much more detail than you are providing here.

Ok ignoring EqAPO for a moment. If I want to use anything at all in this manner with Netflix or video games or whatever, how is it done? How can Netflix be used with something other than shared WASAPI?

 

[eycatcher]

I would recommend you bring the gain on sub channels up at least 3db on the sub1 and at least 6 db up on sub2 on the amp, probably just 1 and 2 clicks. You want the initial target level to be closer or just above to 70db target at 50hz before you measure them with the Anthem. ARG Genesis does a much better job cutting it's eq rather than boosting it. To do this, use the Anthem software to help you. You can do a "quick measure" on each sub to help first to help calibrate the sub amp gain to the target level. The other channels all looked good so then remeasure just the sub channels and upload the settings. Also before you do so, make sure any eq, phase, crossover setting on the subwoofers themselves is disabled. This should also help the phase alignment.

 

[Snoochers]

I would recommend you bring the gain on sub channels up at least 3db on the sub1 and at least 6 db up on sub2 on the amp, probably just 1 and 2 clicks. You want the initial target level to be closer or just above to 70db target at 50hz before you measure them with the Anthem. ARG Genesis does a much better job cutting it's eq rather than boosting it. To do this, use the Anthem software to help you. You can do a "quick measure" on each sub to help first to help calibrate the sub amp gain to the target level. The other channels all looked good so then remeasure just the sub channels and upload the settings. Also before you do so, make sure any eq, phase, crossover setting on the subwoofers themselves is disabled. This should also help the phase alignment.

Thank you for this. I’m actually happy with the EQ now that I have equalizer apo. Now I just need to get it to work with everything

 

[eycatcher]

OK no worries. I still recommend to level match the sub channels first as it looks like you missed that initial step. Genesis is usually really very good at bass management. Two subs are tricky on opposing walls, they may need to be 180 degrees out of phase from each other. It may work better if you rotate them 90 degrees from each other or offset them on the wall. Also mixing ported and sealed can create phasing issues as well. I would expect you should have killer bass with two 21" subs. Not familiar with APO, However It doesnt matter what EQ system you use, generally best practice is to level match then you want to minimize boosting frequencies, the software does automatically, it only boosts about 6db max to protect the speakers and amps( for +3db you need twice the power) however it can cut much more of an amount. Another tip is to increase the gain on the sub so that the level at -3db freq point near 20hz is at the target level. The quick measure is a good tool also to get the best placement in your room. Moving two big subs may not be an option but if it is you can use REW or an spl meter with some sweeps to measure where the mic would get the best room gain and then try to put your subs in that position. Good luck.

 

[boxerfan88]

I have made a lot of progress by using Equalizer APO. I have a question and hope someone has the answer. I saw this on Reddit:

"Keep in mind that if you use an AV receiver, there is a good chance you are bypassing EqualizerAPO anyway. It hooks into Windows Audio API, which is bypassed when bit-streaming audio directly over HDMI or SPDIF."

Is this true ? If so I am screwed because most of what I watch is TrueHD Atmos over HDMI etc.

Yes that “paragraph” is correct. If your MPC-HC is sending the audio stream as a bitstream (undecoded) via HDMI/SPDIF to the AVR for decoding, then the bitstream audio does not go through EqAPO. This is normal.

If you want EqAPO to take effect, your video player must decode the audio bitstream into multi-channel PCM, and then push the PCM audio through Windows OS.