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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!

I wanted to start a topic that focuses on the parameters of these two upsamplers in A+ and perhaps even get some objective measurements of Fourier transforms (impulse response graphs) from those people who can capture them. My experience comes from these software upsamplers as they are implemented in Audirvana Plus 3.2 - But I don't think there's anything keeping others with different players from chiming in here. To kick things off, I'm using @copy_of_a 's extreme settings for iZotope as shown below. In order to avoid aliasing with his settings, you have to use a software plugin to further lowpass the signal. In my case, I'm using FabFilter ProQ2 steepness: 3 length: 500,000 Cutoff: 1.25 Anti-Aliasing: 50 Prering: 0.36 For SoX, I've chosen the below based solely by ear. No LP filter is needed with an EQ plugin since the filter is much steeper it seems. Admittedly I haven't put near a much time into SoX as I have with iZotope: Bandwidth: 90 samples: 500,000 Anti-Aliasing: 85 Prering: 41 Overall, the biggest difference between these is the inability to finely tune the filter steepness in SoX. I've read some anecdotal things and tend to agree with most of them. For instance, Sox seems to present a great image/stage/separation, but comes at the expense of some more ringing that can be heard in the "shimmering" highs on cymbals. What's interesting though, is that the preringing doesn't seem too bad when I scrutinize the timing of a kick drum - hearing first its batter head which should be followed by its resonate sub bass thump. I find that my setting in iZotope still perform best here - but the SoX isn't nearly as bad in the lows as with the high frequency "smearing/shimmering" I hear. But this could just be my settings, more time will tell which is the better upsampler for me. What else have other's experienced?

JRiver has implemented upsampling using the open source SoX codec for a while now, but it is only implemented as a checkbox "on or off" on the local playback. The full parameter set of SoX enables the user to choose between linear o minimal phase, steep or slow roll off options. This is important to upsample in what one can consider a more musical way, improving the sound reproduction (optional!). In the link below I have compiled a few graphs of different hardware implementations. I invite everyone who thinks this is important to post a +1 on their forums for the feature request. Thanks! https://yabb.jriver.com/interact/index.php/topic,113602.0.html