Linear Phase EQ - Room Correction: Part 2 - rePhase
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- เผยแพร่เมื่อ 16 ก.ค. 2024
- In part 2 of my linear phase room correction series, I show you how to interpret the rePhase user interface and turn your EQ files generated in Part 1 into linear phase FIR filters and impulse files for use in convolution, one method of which we will explore in Part 3 of this series.
As always, if you have any questions, just type them into the comments and I will answer them for you! Enjoy!
rePhase download: rephase.org/ - วิทยาศาสตร์และเทคโนโลยี
Something I want to mention that may or may not be important to you. Is that the delays mentioned with the associated taps do mean a delay of that length through playback of audio when using the filter. Luckily total delays up to about 140ms are quite imperceptible to us, but if you wanted to sync display output to audio output, you could always add delay to your visual signal. Typically this can be done with most TVs/Projectors
I would love a follow up to this series! These videos are very helpful to learn what's possible from free tools
Thank you! I'd love to do a follow up at some point!
Currently trying to complete a long term project first- my concrete casted counterpart to Nick (from ToidsDIYAudio) and I's RuleBreaker speaker. Unfortunately due to long setting times and the many iterations required to get it right, it has been taking far too long to complete.
I'd also love to hear what you want from a new series if you have any ideas? I have plenty of my own, but it's nice to see what my audience wants too 😁.
@@Elliott-Designs Wow that sounds like the dream team of DIY audio on TH-cam! If that's the case, I would love to see y'all build a DIY speaker at some certain price point start to finish (sounds like you're doing that already) and then use these correction tools on that same speaker to produce a great house curve. It would be awesome to see that whole process in a digestible format, like your videos are. Also to expand on that, I would love to know if it is possible to use this convolution process in a Digital Audio Workstation, for mixing and mastering purposes.
@@dedik8SKB I'm so going to make a video on that. There's plenty of free tools available. Thank you
Wow. great information!
Glad you liked it! Next one we'll actually be able to use the files through a neat program I've found that interrupts the windows audio output and uses your computer as the DSP!
You helped me a lot! Thank you, thank you, thank you So much! Brother, you have no idea how cool you are, I take off my hat! you have revealed very useful information about Super-high-quality sound! I'm absolutely serious, you can move the big bosses of the audio industry!
Thank you very much for such high praise!
Anything in particular you might like to see from me?
@@Elliott-Designs Thank you for the offer. I don't even know what to ask for yet... Your phase correction and frequency response technology was the most understandable among others. I was able to repeat this on two different pairs of my speakers (there are two more pairs of speakers from my collection left without correction, they are waiting for their finest hour) In all cases, I was very pleased with the result. I heard a lot of new sounds in my favorite music recordings, I was able to hear a very clear transparent space between the instruments, and even sounds from behind... and in general, any music has become more pleasant to sound!
@@syelchin so glad it's been useful to you 🙂
Today I came up with the idea of whether it is possible to get rid of thd harmonic distortion of the speaker in a similar way? The speaker introduces parasitic harmonics in the sound, but if you generate reverse (antiphase parasitic harmonics) that will correct the distortion at the output? Do you have any guesses how to do this using the same programs?
@@syelchin Very interesting idea, theoretically this should be possible, however the very high Q peaks that represent the harmonics will likely cause noticeable pre-ringing, however this is something you would have to test. Even so, you would need accurate measurements of the driver distortion so your results aren't skewed. One way I'd imagine doing it if you don't have an anechoic chamber would be to make a low distortion battery powered amplifier, a laptop, and a calibrated measurement microphone all out to the middle of a field, place the back of the speaker on the ground and rig the microphone to take vector averaged measurements of the driver. You could try other methods but I wouldn't be so sure of the success rate of those. I'd very much be interested in hearing how your testing goes. If you wanted to you could post about it on ToidsDIYAudio.com under the DSP section 👍
Thanks!!!! From 0 knowledge to 100%, excellent video ^_^
Great 👍
Thank you!
Hi Elliott, thank you for sharing
i just discovered APO a few days ago just waiting on a mic to try REW and this channel today so perfect timing.
Are you aware of any issues using linear phase eq with Fidelizer ?
i ask because you mention processing power so im not sure if it just takes power to make the change or to maintain it ?
also since playing with apo on realtime my dac says only 48khz instead of the normal 192, maybe i missed something during install ? i did check and windows set to 24/192
thank you for your time
Processing power is only in comparison to audio processors like minidsp or other typical hardware based processors. Any slightly modern PC should be able to do just fine. A raspberry pi for example can handle processing 8 channels, each with more taps than you'll ever use.
As for the 192khz issue, head over to toidsdiyaudio.com, there's a forum section called DSP. I'll make a topic for EqualiserAPO, if you could post a few screenshots of it all there, I might be able to help 🙂👍
@@Elliott-Designs thank you for your reply.
I uninstalled apo then changed backed to 192 np, I don't remember changing it to 48 so was just curious if thats the max using app.
I'll check out that link as well and hopefully get informed by the time I'm ready to apply all 3 of these, again thank you very much.
I'm so tech challenged, sometimes I think I was born in the wrong era and would have faired better when the most advanced tech was the telegraph... 😂
@@phillipmorris9847 Nope, it doesn't max out, there's something strange going here, as I said, I'd be happy to help on the forum toidsdiyaudio.com/forums/discussion/equalizer-apo-troubleshooting/
There's even people on forum pages (not the Toid's one) that have used up to 384khz/32bit no issue
Likelihood is that it's a communication error between the driver and eqAPO, there's a few things we can try to fix that!
For speaker box correction, should I use port tuning frequency or the dip on the woofer? ( Near field measurement
That's odd, they should be one and the same, the reason that dip occurs is because that is where the port "takes over" in terms of loading. If you go onto Nick's website ToidsDIYAudio, you can start a thread and tag me in if you'd like, that way I can see your measurements and try and help a bit better.
@@Elliott-Designs yeah, the ideal condition would be that. But the box tuning frequency and port tuning frequency can be different sometimes. My box for woofer is a bit small for its VAS
@@user-dh7lt4we2t In that case I'd recommend applying an IR window to "remove your room" as much as you can from your nearfield measurement and then analysing your measured phase response manually to determine how you want to phase correct your port region.
@@user-dh7lt4we2t You could also just try both and then compare the phase responses of the measurements with either correction applied, without an anechoic chamber or at least a portable amplifier and a large field to act as a ground plane, these iterative methods are sometimes the best you can do.
Given how small the average lounge is in the UK and the small likelihood that people will treat the room. Can the rooms handle anything much lower than 50hz?
Totally, in fact, a small room can help get more (although less accurate) bass. This is called room gain, it's the same with car audio, but there it is called cabin gain. I run down to 30Hz flat in my UK bedroom which is much smaller than your average living room. If I had a larger driver in my subwoofer, or more subwoofers then I'd be able to get even lower.
@@Elliott-Designs I guess I worded it wrong. I meant with room gain do we need speakers that can output much lower than 50hz? Do things get messy without room correction that most people don't do?
Love what you're doing here. I'm a carpenter looking to start building speakers. I'm trying to learn all I can about designing speakers but I may need to get help from a designer in the end.
@@bellrobert1978 Not at all, in fact produced music is designed to be played back with room gain. If it's too much bass for you, you can always build a sealed subwoofer (no port, closed box) and utilise its shallow roll off. My content doesn't go into much of this stuff yet, so if you want to know more of things like this, I highly recommend my friend Nick's channel. Toid's DIY Audio. Also, thank you for the high praise! The best place to start in speaker design/building would be to start with a kit/build plan, it's a great introduction into the hobby that way.