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Goal /the incumbent filters: get a faster more “agile” articulate intelligible sound while keeping big bass extension Crisis started when I switched off convolution to compare my setup + original Argo of Jamal at Alambra to the needle drop sold as dsd download (yes yes and to much praise by Acousticsounds ; meaningless comparison since I doubt the needle drop is sourced from the distorting but sometimes gloriously tubey original) : though convolved DSD was a clear winner over non convolved DSD or LP, it sounded too analog and too tamed, fat and slow. Findings Do eQ, REW is donations welcome free : try it (but you need a microphone). Doubt you get comparable results from power usb whathaveyou cables… RR1 (as in https://www.innerfidelity.com/content/warren-tenbrooks-summary-head-measurements-harman) is a mighty good target house curve Mind applied smoothing : a predicted response curve matching Bob Katz’s target (flat to 1 k, 6 dB downward to 20K ) created with “Psyschoacoustic” smoothing in REW resembles RR1 when looked at in VAR or traditional (ie 1/3 octave) smoothing mode (Bob Katz uses Acourate’s Psychoacoustic) Multiply measurements but pick the one that resembles average, not the average summation (ie in REW the Psychoacoustic profiling only applies to single measurements ; as do many options) Avoid messing with frequencies above Minimum Phase zones galore (on the left half of frequencies) (850 Hz max here, different from the theoretical 300). Lower Shelves are great : found out that RR1 paralleled the measured response in my room and that a 5 dB 600 Hz start LS puts frequencies below 600 on the RR1 target Dare switch to Manual. Automatic was not at ease, I took the time to flatten the bumps and, up to 9 dB in Minimum Phase zones, fill. Listen (Stereophile CD3 and best of Chesky 2 have been most helpful, mainly the tracks with moving point sources (acoustic clickers, cowbell…) ). I did not have at hand the mike I created the measurements with, only a iPhone + SPL app, useless below 250 and probably giving a false 3 dB bump reading at 500, but measuring would be great too. In the end listening matters more. Bottom line :girlfriend threatened to quit madman but is as amazed as I am in the end : WOW ! Goal met

Can Bad Recordings/ masterings be fixed and made to sound good? Maybe a little creative EQ for each track? Decompression/Declipping? Maybe a better system does it for you (realizing the potential within) ? Color me warm with sonic sunglasses? This topic was inspired by posts in another thread some of which are quoted below to kick things off. -----------------------------------------------------------------------------

Measurements like statistics in politics can be manipulated or even ill used with best intentions. In this thread I intend to go under the hood showing the options I chosen along the way while creating convolution filters. So maybe I'll expose my goofs and you're welcome to comment them. Or maybe this will enlighten your own choices and help you take better informed decisions.

There are several threads discussing target curves ; please let's keep this one tidy with simple sharing of our findings regarding Matching House Curves and Mastering. So yes there's a postulate that beyond subjectivity and idiosyncrasies of loudspeakers/room fit, there's a rational for using different curves so to get closer to the conditions in which the art was created/approved. The linked paper is an excellent read. I provide a picture of the curves I'm referring to (top to bottom : JBL Synthesis, Harman RR1, Bruel & Kjaer 1974, not theoretical but how I actually achieved them L + R with my room/system) ); please specify if you use different ones but let's not discuss their respective merits and motivations here. Once again, let's focus on matching, simple, straight to the point. However a few words about your criteria might be useful. AFAIC, it's mostly soundstage, how things fold together. The obvious more or less bass and sub bass criteria is not that operational and how vocals are infatuated or not tends to be the go/no go point for RR1 (above the 2 others between 160 and 500 Hz) vs either BK or Synthesis that are equivalent in that region. Also, the extra mids from 800 to 4 K of the Synthesis often act as an appealing reason to go Synthesis even if there's a touch too much bass, because of extra presence and delineation of soundstage, instruments and vocals separation. Next post is an example of how I suggest we populate this thread. If deemed necessary, precisions about decade of mastering or country could br added 17839.pdf

Hi, I have got a Cambridge Audio CXA80 amplifier, an Audiolab M-DAC+ and a pair of KEF LS50 speakers. I'm looking for a high quality graphic equalizer similar to the Behringer Mini FBQ 800. It's price tag (80$) says a lot about its expected sound quality though... I'm not looking for a dual band EQ with as many as 20 bands to adjust for each channel, but a simple one yet with a high sonic quality. What would you recommend me?

Just looking to compare contrast mine with yours. I will show you mine if you show me yours

Hi all, I have a matter of months experience with Audirvana previously using a network audio player via hardware eq to dac and amp. What I am struggling with, I believe, is assessing accurately replay gain for each album I want to listen to. I know replay gain can be tagged in Audirvana but to my ears gives a poor sound. Also I am using the eq plugin and have bluecat gain plugin for some adjustments. What the software does not offer ( I believe) is any visual metering plugin to accurately render gain. I have had a bit of a search on our site and see software eq such as Fabfilter pro q 2, Izotope Ozone 8 amongst others. Is that the way to go? Any further suggestions/advice welcomed.

I bought a very inexpensive, new in the box Pono music player recently. Made a balanced star-quad cable for my HD650s and found that it is a very nice player, but the combo of HD650 and Pono seems to lack a bit in bass and midrange, resulting in an overall too-bright a presentation. I'm already using HQPlayer convolver on the PC to apply EQ and cross-feed, so I came up with the "brilliant" idea of applying DSP impulse response files to music before transferring it over to Pono, hoping to hear a much more balanced output through the headphones. My question is, what's the best way to do this? I have Sox and used it to convolve files before. I assume just convolving the music file with the EQ impulse response file will be sufficient to get a better frequency response, this seems simple enough. The more complex question is how do I do this with the four impulse response files used for cross-feed by HQPlayer? My current setup looks like this in PC HQP: Am I correct that to apply the same processing manually, I would have to convolve the left and right channel music files with L-L.wav and R-L.wav (respectively) and then mix the resulting two files at 50% each to get left channel output? And then do the same with the right channel: convolve L-R.wav and R-R.wav with left/right channel audio, then mix at 50% to get the right channel output file? I'm going to test this, but was curious if anyone can confirm or deny that this is the correct process. Thanks for any help or insight, and happy holidays!

Here are some itching funnies (and ready made for REW targets) I’d like to share. First, Let’s take the same set of Impulse Responses obtained by Sweeps and average* them 3 ways in REW : Frequency Dependent Windowed RMS Average, RMS Average, FDW Vector Average. I add results from Real Time Analysis of Pink Noise measured while Moving Microphone. There have been some changes between MMM and Sweeps times but still there’s a good consistency between RMS Av and MMM results while FDW RMS Av and FDW VA yield radically different results, in my room with my speakers. Hence, you’d better know what you (or the expert and/or commercial software you would trust) are/is to pick. If I am to obtain textbook Impulse and Step Response (thanks to RePhase) I have to pick the FDW VA. If I set Phase with the FDW VA and then set Amplitude on RMS Av or MMM, Impulse and Step remain OK but not as pretty. This raises the question of what but the VA is a commercial software picking if it is to display textbook Phase related results. And that conflicts, unless What You See is Not What You Get, with the choices of targets (neither B&K nor Harman dubbed them such) such as RR1 or B&K. RR1 low end extends B&K that dates back to Pink Noise and RTA time, so no Impulse Response, no FDW, it’s Steady State that is measured… So, if one is to choose a target related to Steady State, one should better apply it on Steady State measurement… *L and R are in different environments in my room and hence measure differently below 300 but for the sake of simplicity they are here added/averaged