This has been a game changer for me, thank you! Even with some 50mm thick acoustic panels in my lounge room, the mid range was too forward and busy. Post correction, it has significantly improved the imaging with bigger soundstage and dare I say more "holographic" presentation. It also gives more weight to the low end and allows the sparkly high frequency detail to shine through. Magnificent! For anyone else looking to get started, I followed GIK Acoustics Room EQ Wizard Tutorial video to take the measurements. That leads perfectly into this guide!
@PassionforSound one thing I will add: I made a "mistake" by applying corrections to the signal with 1/6th smoothing instead of psychoacoustic originally. It used all 20 filters but worked to get the final output close enough to the room curve with only two minor boosts. This is what my listening impressions above were based on. After I realised you said it should probably use psychoacoustic smoothing, I went back and re-did it. It used fewer filters and gave me a lower headroom number, but the end result just wasn't quite the same. My conclusion is that it's definitely worth playing around with the smoothing to see if it works for a given measurement since it may give you a more accurate and exciting result!
EQ is by far the greatest thing in audio. Free, highly effective and allows for speaker/headphone designer to not compromise as much between FR and technical performance.
What’s funny is I’ve tried those Oratory settings before on my headphones and everytime I just like the headphones as they are stock and turn them off. Same goes for pad rolling, never once found a pad that beats out what the headphone shipped with.
@@mikeg2491 The Oratory presets are usually personalized for taste, but still my experience is the opposite in that essentially every headphone is better with than without. Same experience with pads though with IEMs some tips have been way better than the stock ones.
It's always going to be personal and I'm starting to think it depends on each listener's priorities. Those who prioritise tone above all else will gravitate towards EQ. I'm in the other camp though where the slight timing, imaging and resolution smearing created by EQ is enough to detract from the listening experience. To be clear, I'm not promoting one side over the other - what matters is that people enjoy the music.
I know nothing about this except that the final result is obviously a better sound tailored to my taste and room. I am wondering if there are any introduction type tutorials along with what you teach here? Sorry but it is all new to me. Thanks for posting this.
Lachlan, thanks for this video. I've made a first pass effort using the rePhase method you describe and have gotten good results, but I need to work on it some more. A suggestion for you is to run REW with and without the convolution filter engaged to see the results for IR, phase and EQ and then explain what one should look for and how to interpret the results. I've done this with the filters I've created, and I see the changes, but I have only a superficial understanding of acoustics. Again, very nice video! 👍
Glad you've started trying it out. Finding the right room curve / target curve can take some fiddling. I actually thought about remeasuring with the filters running on the DMP-A8 just to see the phase shifts created. I can't do it with the rePhase version because that has to go via Roon and I can't measure via Roon when using REW.
@@wc8617 1. In REW, select the Generator icon and Save to File sweeps in .wav format for the Left and Right channels. Use 1M, 48 kHz, 0 to 24000 Hz and add a timing reference of 4 dB. Import these files into Roon 2. Take REW measurements by playing the L and then R sweeps through Roon; select Measure, Use Acoustic Timing Reference, 1M Length, 0-24000 Hz. Make sure you mute the computer speakers. Start the measurement in REW, the sweep will start and stop with the ‘chirp’ of the acoustic timing reference. I do not use any upsampling for the measurement process. I like to first align the L and R impulse responses by moving the UMIK-1 microphone as appropriate. If desired, take multiple sweeps and dB average in REW. 3. Create the Convolution filters using REW and rePhase as described in Lachlan’s video tutorial. Load the filter into the Roon DSP engine. 4. Retake L and R sweeps as described above with and without the Convolution filter engaged. You should see the frequencies EQ’d, and improvements in the IR and phase traces. If done correctly, the music will sound A LOT better. Good luck!
I played also a lot with lots of method in REW alone or with rePhase. I managed to fully iron out EQ and phase, or just use simple EQ, and tried everything in between, including room mode dips with clever phase filtering. Neumann KH 310 with dedicated studio converter… these are very good and powerful Class AB amp, which is important as they have naturally really good flat phase response despite being 3-way speakers, plus EQ correction does not push them to distort even with 6dB boost that requires double effort from amp to perform. At the end of the day all failed and were much more useless then the powerful algorythm of my Sonarworks SoundID, but likely Dirac and IK ARC are safer too. Note: I still measured the SoundID correction with REW and applied further fine tuning as it is not flawless, but I got much tighter response, and coherent response without manual phase correction. Because the problem is always IR caused delay and pre-ringing, or sometimes post-ringing side effect. :(( This hasn’t been mentioned or presented in your vid. Listening test is always the most important beyone freq response or phase response smoothing. Michael Jackson’s - Beat it… the kick part right after the intro will always reveal if you went too far with phase linearization, and always always check distortion ratio and group delay, which is much more important in our brain’s process when we are listening to music.
The deeper dips in the response are *probably* there because of a cancellation. These are often from a floor reflection (or ceiling) that is a path that is about half the wavelength longer than the direct path (from the driver to your ears). And these dips are from the specific mic position. OR it is from phase cancellation between the drivers because they are at different positions on the speaker. In the crossover region Which are going to happen differently at different listening positions - so compensating for the room/set up is problematic, in my opinion. I would measure the speakers at 1 meter on the listening axis - this removes the floor/ceiling cancellations. You need to use the listening axis that is midway between the center of the tweeter and midrange/bass driver; so the distance is the same to the two drivers. And I would average the left and right speakers. Then you are not trying to compensated for "physical" things that are "outside" the speakers themselves. These "physical" things are going to vary for all the other possible listening positions - so the changes are *not* correcting the sound for any other listening position. This process can only go so far (as you mention) and so it is best to use it to "fix" the frequency response deviations of the speakers. Speaker placement and room treatment is the best way to get their response to match.
Thanks for the info, Neil. I can't recall now which set of measurements I used for the examples in this video so I'm not sure where the mic was placed or if I used a real time measurement across a small range of movement with the mic.
Room correction is essential in unprepared living rooms. I currently use Roon with 2 convolution filters in succession. First is for two channel correction. Second for full response including subwoofer.
Sorry, I just came across this video recently. As a novice, I would greatly appreciate both of your feedback. Using your excellent tutorial, I did create 2 filters similar to what @siamxiii suggested. Just to clarify, for the 2nd filter, this is based on your measurement of running both your 2 main stereo speakers and the subwoofer concurrently. Is that correct? Thanks so much.
@@wc8617 Exactly. However I changed coorrection a little. In first convolution filter I correct speakers only below 500hz. And then for full setup I correct spikes with parametrical EQ. Like result much more.
@@siamxiii Excuse my ignorance. Why would 2 filters be better than just the single full setup? Secondly, for the full setup, why wouldn't you also just correct below 500hz? Much thanks from a newbie!
@@wc8617 Because in first step you correct each speaker, which in my room (pretty assimetrical) is crucial. I also correct below 500hz in both steps. I guess corrections above 500hz mess with phase and i don’t like effect of it. In my case purpose of correction is to solve bass problems, which are very prominent. I don’t need flat flat response in full frequency spectrum.
Brillant video! It works even with the DMP A6 . If i understand well, REW EQ filters recommend me 6.3 db headroom for my 10 filters. Does it mean i have to reduce the EQ Gain by 6.3db in the DMP A6, config 1 menu, EQ gain as well? Fortunately, my 300W has enough juice for it but i must listen at 64% instead of 45% volume for the same 85db output.
Another nice one! Saved to watch properly another day. Am looking for a streamer. Eversolo was top on the list cos it works with Apple Music but I don’t like the DAC inside it.
Hello and thanks for the very useful video. A question: why not focus on frequencies whose length does not depend on the movements on the most common sofa and movements of the listening position, so we are talking up to about 500 Hz (about 70cm). Do you see any critical issues in applying your method however limited up to 500 Hz? Does linear phase work with RePhase cause any mess if the file you are working on is limited?
You can absolutely focus on any frequencies you want to. I don't see any reason not to address 20-20,000Hz if we have the opportunity to though. As for using rePhase, it will work happily with any range of frequencies so go for it! Have a play and see what sounds best to your ears.
@@PassionforSound Because there are differences after the Schroeder frequency, the 180, 200, 300 Hz due to reflections etc with curves that rise and fall depending on small movements and therefore I find little sense in having a theoretical flat curve for a head that must be stationary as in a recording studio as at work. So I'm talking about passionate listening, yes, but for pleasure and perhaps to share with other people. Of course, during the measurement phase, I could create a cloud of points around the head within 50 cm and take an average. And here comes another question, the best average to do is the RMS avg + Phase avr tool? Always in case I then have to make a linear phase filter in RePhase. Thank you very much for the attention
I have deliberately not delved too deep into this because there are SO many factors to consider. If you're working with a larger listening area, taking a real time measurement in REW is a good idea to even out the FR across multiple locations.
Im a little slow at this, but would you use correction on the internal devices eq/peq as well as roons eq/peq? I would think it would be one or the other but I don't know..
Hi, please rate the Topping l70 both as an amplifier and as a preamplifier? And especially the opinion in relation to the Topping A30 pro and the Topping L50? Thank you
It's shocking to me machine learning algorithms are not more widespread in the digital signal processing arena to facilitate detail retrieval. Theoretically, a versatile algorithm properly trained on a catalogue of sounds and music should be able to upscale any signal to say for example PCM64bit/768kHz or any arbitrary (lossless) encoding that is desired. Like (Google's) VP9 codec for audio. Like an artist drawing an ideal waveform on a canvas. All solid state electronics would become obsolete. Digital signal processing algorithms are so useful and powerful to use before converting the signal to analog.
That's a very interesting idea! I wonder if they could provide the accuracy needed down to a timing accuracy of 1-2 micro seconds to achieve a sound that's completely natural to our brains.
I don't know the Qrange plugin and what it's compatible with, but there are quite a lot of export options from both REW and also the rePhase software too.
