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About Me

Found 6 results

  1. 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
  2. Just looking to compare contrast mine with yours. I will show you mine if you show me yours
  3. 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.
  4. 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!
  5. Audiophile Neuroscience

    Can Bad Recordings be Fixed?

    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. -----------------------------------------------------------------------------
  6. 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?
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