thanks for this, I'm a long time EAPO user, but I feel like I never got it dialed in right. when using rephrase and a 5.1 system, are you taking a measurement of each speaker in rew, passing it through rephase, and then using that per channel in EAPO, or taking measurements of say the front 3, and averaging that, and then passing that through rephase, and using the filter for the LCR channels... I had tried fir filters some time ago and it just sounded lifeless the last time
Per channel, but because I use a spatial average type measurement (MMM-moving mic measurement), it uses periodic pink noise, I'll probably make a tutorial at some point. I digress... Because I use that method, it means descrepencies between left and right channels are less. When doing individual measurements, or just a few averaged, differences in the responses are typically exaggerated between the channels. But yes, I have a different filter for each of the speakers.
Typically the lifelessness comes from the EQ itself not being done right. Filters from FIRs compared to just PEQ aren't actually any different when they are both minimum phase, assuming you have enough taps applied of course. Also, I recommend checking out my other tutorial on how to do the loudness correction properly, the audio out built into the program for the calibration isn't the right volume. Also, if you aren't already using loudness correction, I highly recommend it.
You just made my life easier with this software 😍👍. I would like to know more about phase linearity. How that affects sound? Sources of phase shift etc. Thank you 👍.
I saw that you have already seen my video on linear phase, but I'll try and add to that here as best as I can. Sources of phase shift: Normal EQ (minimum phase crossovers and parametric EQ), passive crossovers. The best explanation for how it effects sound I could come up with was in that linear phase Vs minimum phase video, however you could always try both out for yourself, should be most noticeable with strong corrections in low frequency.
What parts about phase shift do you need more details on that weren't in the original video on the topic? It would be useful for me to know so I can make a follow up video clarifying these details.
@@Elliott-Designs First of all thank you for giving such a detailed reply. I would like to know the difference in sound between the minimum phase and the linear phase. It will be great if that could be recorded with a mic while playing white noise. In REW SPL, phase can be seen during the measurement. I would like to know why the phase measured with REW is so messy and should I care about the phase measured with REW? Thank you in advance.
Ah, you shouldn't care about the phase measured in REW at all. That phase is all to do with how your speaker is interacting in your room and is totally natural. If it wasn't there it would actually sound horrible. The reason we don't want it for the EQ is it makes it less effective and adds time smearing, particularly for low frequencies and we don't want it in crossovers because it makes them even more imperfect. A linear phase crossover will integrate drivers much better. The constant phase change in crossovers means that as you subtract gain, more and more phase is changed, it would be the same moving the driver further and further back as the gain reduces in the slope. Think about the effects of physical driver offset, you want your tweeter to be aligned with your woofer. It's that exact same issue. I could probably show you what minimum phase does to the response in REW, but because it's time dependent it has to be done with a sweep, not pink noise.
@@Elliott-Designs Thanks alot 😍. I got it. One small request there are 'audiophiles' crying for tube like sound. If there are any provision to add limited amount of 2nd order harmonics with this software, please do a video on this topic too. Thank you for your patience and time.
I didn't get a Configuration Editor, but instead, the site I got it from told me to navigate to the dir I installed to, and start it there. Aside from that, it all went "swimmingly," as ya'll say.
Thanks for the video! In EQ APO I use convolution wav files, exported by REW. But there is a lot of latency! When I press stop on the player, music keeps playing for some time. Is this a problem for quality music listening? I know it's a problem when I do measurements with REW, because the waterfall graph shows very long decay in high frequencies... Is there a way around that?
If you're talking about what I believe you're talking about, what you're doing is effectively doubling the colouration of your speakers and your room, that initial convolution is of your original measurements and effectively "describes your room", try enabling the convolution whilst listening through headphones, you'll see what I mean. If I'm understanding you correctly that is. You might want to have a look at my series on linear phase correction, to see how to correct for your room and speakers. That's if you want to try and be faithful to the what the post-production audio engineers intended anyways, if you prefer the colouration of your room then keep it 🙂, after all you're listening to it back, it's not like it's critical as if you were a pre/post-production engineer 👍. Oh, as for the long delay, that's because it's quite literally doubled the decay time of your room.
Thanks for the video. Not a whole lot of good how to content on this app, most of what you explained I had to figure out on my own. I was wondering if it is possible to route the output of a DAW like studio one to the APO app before it hits the interface converter outs? Thanks for your time
No worries, as long as you don't mind the extra delay you can configure your playback in your DAW to different output types, DirectSound will likely work, but I'm pretty sure the standards for such interfaces like ASIO are not compatible since it's a different audio chain that operates independently. Other than that, make sure to try different methods of applying equaliser apo to your interface via the configurator and then restart the computer before attempting to use it, because it will need that in order to apply to configuration. Hope that helps. You also mentioned you had to figure a lot out on your own, any chance you could give some input on what I wasn't clear on so I could perhaps make a follow up in the future for others that were also struggling? Thank you.
before I saw your video I had downloaded the app to tune my speakers. It took me sometime to understand what I was doing and though there were some tutorials on REW and APO there seems to be certain functions and features that are not well explained in both programs. For instance in APO the signal chain was not so evident. The way you explained it was fine and understandable but it took me about 3 hours of messing around to understand it. I did not have your video at the time for reference. Right now I have been searching for a way to tune the out put channels of my DAW with AP0 for ATMOS. It seems not possible however maybe you know of a way. Thanks for your time. @@Elliott-Designs
@@metalplatedfacejob Ah, yes, unfortunately not something that can be done with ATMOS currently as it's a tightly controlled ecosystem, that and windows audio only supports up to 8 channels as standard.
I did this but I used smaart v9 in transfer to capture live export to REW save one file for as is untouched and then the other file repaired. Open both files in rephase used linear 4500 export out to eqapo for 5.1 audio via hdmi to avr. IMHO sound way better than Dirac or minidsp. I just ordered iem measuring rigs and next I’m gonna order headset measuring rig. Try it. I had a question how to use convolution files on iPhone to play audio there an app? Or is this why they use daps or dacs? I’d like to take it portable with all my measured gear including AirPods pro.
So glad to hear it worked so well for you! Sorry, I don't use IPhones, on android you can sometimes find some systemwide EQ apps that support convolutions, particularly when rooted. You may find there's some IOS apps that can also do this for you, but unfortunately I couldn't name any specifics. Best of luck!
Thank you, and very true about the delay. You can get a lot done with 10000-11000 taps still, and that delay isn't too bad when the rest of the chain has very little delay. Also, you can lip sync correct either in a video player like VLC, some TVs have it, also most AVRs. You can add delay to display output essentially to resync. Wikipedia says audio can be delayed up to about 140ms before it becomes noticeable for lip sync mismatch, those numbers above should give that. ^those numbers work really well for low frequency corrections, but only really when optimisation is used like in rePhase to get more out of some taps. Also, doesn't matter for music, so you can go nuts with that if you have the processing power
I'm not actually sure. I have a feeling it may just be JSON formatting. If you create some filters using the GUI then save that config, then you'll be able to see how it formats it.
Hello,...does digital equalizers such as eq apo harm the quality of source signal?, I do use it for bookshelf stereo setup primarly but at the moment I do not have good acustics, so cant evaluate if it harm somhow to soudstage, clarity etc, actually I need it because I found my new dac trable expanded and I need to reduce it all the way from 2000 HZ - 3000Hz about 1.5 db and fter that about 1 db, so would it e preferable to change the dac until I can or eq is 99% ok?
Hello 👋. The EQ process itself should be perfectly fine for audio quality if you stick to these rules: if the total gain from the sum of your filters is positive (red gain value when viewing the summed output at the bottom), you'll want to apply a negative preamp on all your outputs (on all of them so the levels don't become mismatched). No positive gain peaking filters (only cuts), and no too high of a Q (sharpness of the dips, etc). Both of these are to avoid noticeable time domain ringing (think echo). Although if you follow those rules the EQ process itself doesn't hurt sound quality, the way you apply the filters might. You may want to have a look at some of my videos on room correction, or there are plenty others that have done it too which might have some simpler methods if you don't mind changing how the speakers sum (if you want to know what I mean by speakers summing differently from typical methods, I have a video on linear phase filtering explaining that). Is there anything else I can try and help you with 🙂?
I see the issue, those cards have a fake subwoofer channel. What you want to look for is a 5.1 or 7.1 card with multiple analogue outputs, you can then choose to only use a portion of those channels, they're quite inexpensive too, all you'll then require is an amplifier for each of those outputs, one one that covers all that you need. Hope this helps 🤞
miniDSP HD is one, but it's still rather limited in terms of the amount of taps/samples you can use which severely limits low frequency performance with convolutions. Anything else (that I'm aware of) would be in pro audio, and therefore definitely not cheap. It's why I cover it in software, not many good implementations on hardware based DSPs at the moment unfortunately.
@@Elliott-Designs too bad.. I already have t rack dsp, and i dont think this would be upgrade of mine.. I did stumbled on ir loader from tc eletronics, but i dont know if that would work the same since it has impedance made for guitar
@@ivanmaskov Sorry, I'm afraid I have no experience with either of these things and therefore can't help you with this issue. Hopefully you can find someone that can help you. Best of luck 🤞
What is it you are using it on? I might be able to help. There are some systems though that seem to just be completely incompatible. Does it say anything in particular when applying a filter in the configurator? Have you tried the 32bit version (or 64bit if you were running 32bit)?
I don't know if you're still active on youtube, but I need help with rePhase. I've made the cvs file and imported it into it, but whenever I export as a .wav file 32bit iiie, there's information in the file but it doesn't do anything in my convolver
Still active. Just unfortunately very busy and current projects are taking far too long. Nonetheless, still here to help. What convolver are you using?
@Elliott-Designs booger by voxengo, I've fixed the problem with loading it, but now I'm trying to find a way to cut down on latency while keeping the fluctuations negligible :(
@@FriendlyAudio unfortunately that's not really a thing because the way waves work, the most you can do is use the optimisation filter or lose that high resolution variance in those low frequencies. However, if you use "Psychoacoustic" smoothing in REQ (room EQ wizard) you'll see how little definition we can actually detect in such low frequencies anyways. You may find that it's enough 🤞.
Not that I can find I'm afraid. That's the thing about MacOS unfortunately. Not as many things are developed for it. Unfortunately there's not much I can do about that
No, I've not seen that issue before. Have you tried reinstalling it? The next step I'd go for after that is trying the 32bit version or the 64bit version (whichever one is different to the one you were trying to use). It might be that the driver isn't linked in properly with the configurator, so you could try changing which device it's applied to, or try the experimental settings option shown in the video. Give these a go, I'd love to hear if it worked or not, if it didn't, we can find an easier way to communicate, e.g. email or toidsdiyaudio.com, and I can try and help from there
thanks for this, I'm a long time EAPO user, but I feel like I never got it dialed in right. when using rephrase and a 5.1 system, are you taking a measurement of each speaker in rew, passing it through rephase, and then using that per channel in EAPO, or taking measurements of say the front 3, and averaging that, and then passing that through rephase, and using the filter for the LCR channels... I had tried fir filters some time ago and it just sounded lifeless the last time
Per channel, but because I use a spatial average type measurement (MMM-moving mic measurement), it uses periodic pink noise, I'll probably make a tutorial at some point. I digress... Because I use that method, it means descrepencies between left and right channels are less. When doing individual measurements, or just a few averaged, differences in the responses are typically exaggerated between the channels. But yes, I have a different filter for each of the speakers.
Typically the lifelessness comes from the EQ itself not being done right. Filters from FIRs compared to just PEQ aren't actually any different when they are both minimum phase, assuming you have enough taps applied of course. Also, I recommend checking out my other tutorial on how to do the loudness correction properly, the audio out built into the program for the calibration isn't the right volume. Also, if you aren't already using loudness correction, I highly recommend it.
You just made my life easier with this software 😍👍. I would like to know more about phase linearity. How that affects sound? Sources of phase shift etc. Thank you 👍.
I saw that you have already seen my video on linear phase, but I'll try and add to that here as best as I can. Sources of phase shift: Normal EQ (minimum phase crossovers and parametric EQ), passive crossovers. The best explanation for how it effects sound I could come up with was in that linear phase Vs minimum phase video, however you could always try both out for yourself, should be most noticeable with strong corrections in low frequency.
What parts about phase shift do you need more details on that weren't in the original video on the topic? It would be useful for me to know so I can make a follow up video clarifying these details.
@@Elliott-Designs First of all thank you for giving such a detailed reply. I would like to know the difference in sound between the minimum phase and the linear phase. It will be great if that could be recorded with a mic while playing white noise. In REW SPL, phase can be seen during the measurement. I would like to know why the phase measured with REW is so messy and should I care about the phase measured with REW? Thank you in advance.
Ah, you shouldn't care about the phase measured in REW at all. That phase is all to do with how your speaker is interacting in your room and is totally natural. If it wasn't there it would actually sound horrible. The reason we don't want it for the EQ is it makes it less effective and adds time smearing, particularly for low frequencies and we don't want it in crossovers because it makes them even more imperfect. A linear phase crossover will integrate drivers much better. The constant phase change in crossovers means that as you subtract gain, more and more phase is changed, it would be the same moving the driver further and further back as the gain reduces in the slope. Think about the effects of physical driver offset, you want your tweeter to be aligned with your woofer. It's that exact same issue. I could probably show you what minimum phase does to the response in REW, but because it's time dependent it has to be done with a sweep, not pink noise.
@@Elliott-Designs Thanks alot 😍. I got it. One small request there are 'audiophiles' crying for tube like sound. If there are any provision to add limited amount of 2nd order harmonics with this software, please do a video on this topic too. Thank you for your patience and time.
I didn't get a Configuration Editor, but instead, the site I got it from told me to navigate to the dir I installed to, and start it there. Aside from that, it all went "swimmingly," as ya'll say.
Thanks for the video! In EQ APO I use convolution wav files, exported by REW. But there is a lot of latency! When I press stop on the player, music keeps playing for some time. Is this a problem for quality music listening? I know it's a problem when I do measurements with REW, because the waterfall graph shows very long decay in high frequencies... Is there a way around that?
If you're talking about what I believe you're talking about, what you're doing is effectively doubling the colouration of your speakers and your room, that initial convolution is of your original measurements and effectively "describes your room", try enabling the convolution whilst listening through headphones, you'll see what I mean. If I'm understanding you correctly that is. You might want to have a look at my series on linear phase correction, to see how to correct for your room and speakers. That's if you want to try and be faithful to the what the post-production audio engineers intended anyways, if you prefer the colouration of your room then keep it 🙂, after all you're listening to it back, it's not like it's critical as if you were a pre/post-production engineer 👍. Oh, as for the long delay, that's because it's quite literally doubled the decay time of your room.
Thanks for this, is there a way to apply EQ to an app running in exclusive mode? Thanks
I'm afraid there is not, unless it is done between your final output and your computer in terms of the audio chain.
Found you through Toids DIY Audio
Awesome!
Thanks for the video. Not a whole lot of good how to content on this app, most of what you explained I had to figure out on my own. I was wondering if it is possible to route the output of a DAW like studio one to the APO app before it hits the interface converter outs? Thanks for your time
No worries, as long as you don't mind the extra delay you can configure your playback in your DAW to different output types, DirectSound will likely work, but I'm pretty sure the standards for such interfaces like ASIO are not compatible since it's a different audio chain that operates independently. Other than that, make sure to try different methods of applying equaliser apo to your interface via the configurator and then restart the computer before attempting to use it, because it will need that in order to apply to configuration. Hope that helps.
You also mentioned you had to figure a lot out on your own, any chance you could give some input on what I wasn't clear on so I could perhaps make a follow up in the future for others that were also struggling? Thank you.
before I saw your video I had downloaded the app to tune my speakers. It took me sometime to understand what I was doing and though there were some tutorials on REW and APO there seems to be certain functions and features that are not well explained in both programs. For instance in APO the signal chain was not so evident. The way you explained it was fine and understandable but it took me about 3 hours of messing around to understand it. I did not have your video at the time for reference. Right now I have been searching for a way to tune the out put channels of my DAW with AP0 for ATMOS. It seems not possible however maybe you know of a way. Thanks for your time.
@@Elliott-Designs
@@metalplatedfacejob Ah, yes, unfortunately not something that can be done with ATMOS currently as it's a tightly controlled ecosystem, that and windows audio only supports up to 8 channels as standard.
yeah it seems like it should be easy however it is not.@@Elliott-Designs
I did this but I used smaart v9 in transfer to capture live export to REW save one file for as is untouched and then the other file repaired. Open both files in rephase used linear 4500 export out to eqapo for 5.1 audio via hdmi to avr. IMHO sound way better than Dirac or minidsp. I just ordered iem measuring rigs and next I’m gonna order headset measuring rig. Try it. I had a question how to use convolution files on iPhone to play audio there an app? Or is this why they use daps or dacs? I’d like to take it portable with all my measured gear including AirPods pro.
So glad to hear it worked so well for you! Sorry, I don't use IPhones, on android you can sometimes find some systemwide EQ apps that support convolutions, particularly when rooted. You may find there's some IOS apps that can also do this for you, but unfortunately I couldn't name any specifics. Best of luck!
Good series of video, but over 300ms of audio delay from the filter is extreme. That's very noticeable when video is involved.
Thank you, and very true about the delay. You can get a lot done with 10000-11000 taps still, and that delay isn't too bad when the rest of the chain has very little delay. Also, you can lip sync correct either in a video player like VLC, some TVs have it, also most AVRs. You can add delay to display output essentially to resync. Wikipedia says audio can be delayed up to about 140ms before it becomes noticeable for lip sync mismatch, those numbers above should give that.
^those numbers work really well for low frequency corrections, but only really when optimisation is used like in rePhase to get more out of some taps. Also, doesn't matter for music, so you can go nuts with that if you have the processing power
which language is used in "text config files"
matlab or cpp etc.
I'm not actually sure. I have a feeling it may just be JSON formatting. If you create some filters using the GUI then save that config, then you'll be able to see how it formats it.
Hello,...does digital equalizers such as eq apo harm the quality of source signal?, I do use it for bookshelf stereo setup primarly but at the moment I do not have good acustics, so cant evaluate if it harm somhow to soudstage, clarity etc, actually I need it because I found my new dac trable expanded and I need to reduce it all the way from 2000 HZ - 3000Hz about 1.5 db and fter that about 1 db, so would it e preferable to change the dac until I can or eq is 99% ok?
Hello 👋. The EQ process itself should be perfectly fine for audio quality if you stick to these rules: if the total gain from the sum of your filters is positive (red gain value when viewing the summed output at the bottom), you'll want to apply a negative preamp on all your outputs (on all of them so the levels don't become mismatched).
No positive gain peaking filters (only cuts), and no too high of a Q (sharpness of the dips, etc). Both of these are to avoid noticeable time domain ringing (think echo).
Although if you follow those rules the EQ process itself doesn't hurt sound quality, the way you apply the filters might.
You may want to have a look at some of my videos on room correction, or there are plenty others that have done it too which might have some simpler methods if you don't mind changing how the speakers sum (if you want to know what I mean by speakers summing differently from typical methods, I have a video on linear phase filtering explaining that).
Is there anything else I can try and help you with 🙂?
@@Elliott-Designs For now I am absolutly satisfied with your unsver and before asking I have to deep in a bit. Thank you very much :)
@@janulik9535 No worries, and whenever you have any more questions, please don't hesitate to ask 🙂
@@Elliott-Designs Thank you
which external audio interface should i buy for multiple output filters ?
Hello there. How many different audio inputs and outputs do you need on the device with the filters?
@@Elliott-Designs i need actually just 2.1 , but the problem is , regular 2.1 sound card is recgonized by windows only as 2 chanells .
I see the issue, those cards have a fake subwoofer channel. What you want to look for is a 5.1 or 7.1 card with multiple analogue outputs, you can then choose to only use a portion of those channels, they're quite inexpensive too, all you'll then require is an amplifier for each of those outputs, one one that covers all that you need. Hope this helps 🤞
Is there any hardware unit dsp that can read convulation instead of normal parametric and graphic eq?
miniDSP HD is one, but it's still rather limited in terms of the amount of taps/samples you can use which severely limits low frequency performance with convolutions. Anything else (that I'm aware of) would be in pro audio, and therefore definitely not cheap. It's why I cover it in software, not many good implementations on hardware based DSPs at the moment unfortunately.
@@Elliott-Designs too bad.. I already have t rack dsp, and i dont think this would be upgrade of mine.. I did stumbled on ir loader from tc eletronics, but i dont know if that would work the same since it has impedance made for guitar
@@ivanmaskov Sorry, I'm afraid I have no experience with either of these things and therefore can't help you with this issue. Hopefully you can find someone that can help you. Best of luck 🤞
I cant get mine to work at all... Ive tried all the "fixes" and it simply doesnt change my sound.
What is it you are using it on? I might be able to help. There are some systems though that seem to just be completely incompatible. Does it say anything in particular when applying a filter in the configurator? Have you tried the 32bit version (or 64bit if you were running 32bit)?
I don't know if you're still active on youtube, but I need help with rePhase. I've made the cvs file and imported it into it, but whenever I export as a .wav file 32bit iiie, there's information in the file but it doesn't do anything in my convolver
Still active. Just unfortunately very busy and current projects are taking far too long. Nonetheless, still here to help. What convolver are you using?
@Elliott-Designs booger by voxengo, I've fixed the problem with loading it, but now I'm trying to find a way to cut down on latency while keeping the fluctuations negligible :(
@@FriendlyAudio unfortunately that's not really a thing because the way waves work, the most you can do is use the optimisation filter or lose that high resolution variance in those low frequencies. However, if you use "Psychoacoustic" smoothing in REQ (room EQ wizard) you'll see how little definition we can actually detect in such low frequencies anyways. You may find that it's enough 🤞.
@@Elliott-Designs I haven't messed with rew that much, I need to watch more videos on how to use it
Windows only! :(
Is there a mac version of this?
Not that I can find I'm afraid. That's the thing about MacOS unfortunately. Not as many things are developed for it. Unfortunately there's not much I can do about that
@@Elliott-Designs Thank you for your response!
When im downloading it the configurator doesnt show up do yk why?
No, I've not seen that issue before. Have you tried reinstalling it? The next step I'd go for after that is trying the 32bit version or the 64bit version (whichever one is different to the one you were trying to use). It might be that the driver isn't linked in properly with the configurator, so you could try changing which device it's applied to, or try the experimental settings option shown in the video. Give these a go, I'd love to hear if it worked or not, if it didn't, we can find an easier way to communicate, e.g. email or toidsdiyaudio.com, and I can try and help from there
@@Elliott-Designs oh and also when i try to open configurator editor it doesnt open do yk why?
@@flyinq_baconz9695 That sound like a bad install/initial setup. Try reinstalling