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Found 18 results

  1. Why are there debates if "bits are bits"? Can you hear a difference in streaming audio software? Bits are Bits! Foobar2000 was lighter on its feet, more accurate tone/timbre, quieter, and had better timing of decays. There are a delicacy and precision about its sound. JRiver had a “fatter” sound with more bleeding edges and smeared textures. It sounded flatter (by a good margin) and more veiled in a thicker, unnatural warmth.
  2. (Not in DSP or other sections -- this doesn't seem to fit) One thing that I have always thought to have an ugly sound -- the Aphex Exciter. I have noticed either the 'Exciter' or other kinds of specialized EQ being used in various material from at least the 1970s. This example seems to work well for Linda Ronstadts recordings, and a modified version seems to help control the overly scrambled 'wall of sound' effect on ABBA. It might not correct all of the 'Exciter' sound, but seems to help with the 'chalk on the chalkboard' graininess. The script below seems to help, but an additional amount of treble boost might also be helpful -- additional treble boost doesn't seem to re-instate the 'grainy'/'makes me itch' kind of sound. It seems to make the 's' sound move to the correct place in time. (mitigates some of the reverse lisp on Linda's recordings.) The additional treble boost -- I have tried +3 dB at 3kHz/Q=0.500, and it seems to help bring the highs back without adding the grain back into the audio. This is just a first iteration of the anti-Exciter, and really does help get rid of the 'grain'. How effective -- on an ad-hoc scale, maybe 2/3? I wrote a more complicated anti-Exciter, but it was done with FIR filters, which was stupid, since the original 'Exciter' used analog style filters (IIR near-equivalents.) Basically, this is an interleved set of filters at logarithmically (almost) even spaces, +3 then -3dB at the same frequency with different Q values. Here is a first cut script: (I verified this exactly -- I included a version that can be tweaked in case someone wants to play with it.) HAVE FUN!!! Q0=0.500 G0=3.0 Q1=0.840 G1=-3.0 DECVARIABLE="treble ${G0} 1.5k ${Q0}q treble ${G1} 1.5k ${Q1}q treble ${G0} 2k ${Q0}q treble ${G1} 2k ${Q1}q treble ${G0} 3k ${Q0}q treble ${G1} 3k ${Q1}q treble ${G0} 4k ${Q0}q treble ${G1} 4k ${Q1}q treble ${G0} 5.2k ${Q0}q treble ${G1} 5.2k ${Q1}q treble ${G0} 6.84k ${Q0}q treble -3 6.84k ${Q1}q treble ${G0} 9k ${Q0}q treble ${G1} 9k ${Q1}q treble ${G0} 10.4k ${Q0}q treble ${G1} 10.4k ${Q1}q treble ${G0} 12k ${Q0}q treble ${G1} 12k ${Q1}q treble ${G0} 15k ${Q0}q treble ${G1} 15k ${Q1}q" DEC="treble 3.0 1.5k 0.500q treble -3.0 1.5k 0.840q treble 3.0 2k 0.500q treble -3.0 2k 0.840q treble 3.0 3k 0.500q treble -3.0 3k 0.840q treble 3.0 4k 0.500q treble -3.0 4k 0.840q treble 3.0 5.2k 0.500q treble -3.0 5.2k 0.840q treble 3.0 6.84k 0.500q treble -3 6.84k 0.840q treble 3.0 9k 0.500q treble -3.0 9k 0.840q treble 3.0 10.4k 0.500q treble -3.0 10.4k 0.840q treble 3.0 12k 0.500q treble -3.0 12k 0.840q treble 3.0 15k 0.500q treble -3.0 15k 0.840q" DECEQ="treble 3 3.0k 0.500q" Use a command line this in 'bash', but something similar in cmd.com or elsewhere will also do the job: > play -q infile.wav gain -3 ${DEC} ${DECEQ} 'DEC' does the 'grain' removal, and DECEQ undoes the HF cut (at least partially.) This does work, and if there is interest, I can upload some examples. It is NOT perfected -- but MIGHT be a good first step. John
  3. I'm trying to get an audiophile sound when playing remux ripped blu-rays on my computer. Getting the right syntax to make afir work in FFmpeg would be a way. At the moment I follow Bob Stern's advice to try Audio Hijack to capture the audio output of my video player app and put an AU in. Allloy 2 works but I know I can beat simple eQ with my filters REW/RePhase designed for HQP. I can't get iZotope Trash nor Fog convolver to simply convolve my own 2 channels FIR. The simple but mighty old LAconvolver works and beats Alloy 2 yet. However it resamples my filters to 44.1 and I'd like to keep things 24/96 at least. Furthermore, there is a several second lag between image and sound with LAConcolver. I'll try to sync via video player app but would definetely prefer a better performing solution Any suggestion?
  4. I'm still an audiophile when I watch TV. And my computer is now by best source for ripped movies too since I am back to the future with a Kuro 500 Plasma TV (that doesn't have the capability to process my Blu-Ray remuxes) I could do with the included in IINA (new os x player based on mpv) eQ but I want better. However I must confess I have not been able to write a correct syntax to get IINA trigger FFmpeg afir that would convolve the IR filters I have designed for HQplayer. Anyone can help?
  5. I was fortunate enough to be invited by David Snyder @dsnyder of the San Francisco Audiophile Society to talk about DSP/DRC.While the video begins at my DSP talk, I encourage folks to rewind to the beginning of the video as David has a wonderful presentation on streaming and using a Raspberry PI as a Roon endpoint.
  6. Is anyone been able to use the software version of Dirac with their Microrendu or similar DLNA network player? If so can you please describe your signal path. What software are you using, Jriver, HQPLayer, Roon etc.
  7. Let's assume cables can be proven beyond a reasonable doubt to sound different. Even in the unlikely event that we all would agree on that, what is the justification for using expensive cables, rather than digital signal processing (room correction, parametric equalization, etc.), whose properties can be readily controlled and whose results are easily measurable, to alter what the music sounds like?
  8. 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!
  9. Hi, I wonder if someone could provide a step-by-step guide to achieving room correction using REW and HQ Player. I tried to find a tutorial but the information is scarce and spread all over the web... It's been years since I last used REW to measure the frequency response at the listening spot. I have a mic, REW and I'm ready to go. Thanks, Ricardo
  10. Does anyone on this board have any experience/familiarity with Edifier’s products, especially their powered/active speakers? They appear to be attractive technologically and definitely price-wise. The company seems to have relatively recently entered the US market (via Amazon) and appears to be popular in Asia (based in China) and Europe. They seem to be well liked among posters on HeadFi. In addition the blogger Digital Stereophony http://digitalstereophony.blogspot.com/ has performed some interesting speaker comparisons where Edifier speakers came out well. Based on his recent test (vs. JBL LSR305, Klipsch R15 - PM) of Edifier’s S2000 Pro (incl. DAC/DSP) this seems like a serious performer for the price. Not reviewed is the lower priced S1000DB (w/o DAC). However, except for good reviews on Amazon I’m reluctant to take the plunge despite what otherwise seems to be the real deal. Other speakers in the running: Vanatoo T0 (w/a sub) & T1; Audioengine HD6; Peachtree M24; JBL LSR305; Klipsch R15-PM. Any insight would be appreciated.
  11. Given the loudness wars, I cannot imagine it is impossible to restore some of the dynamic range in a compressed recording. I understand that when music clips, or when you're using the top 2 MSBs in the 16bit word, you're not going to be able to recover resolution as you expand DR, but something better than nothing is surely possible. I would pay good money for such a filter.
  12. Mitch Barnett of Accurate Sound tells us about his journey to become a leading expert in Digital Sound Processing (DSP). Accurate Sound is a remote service using Audiolense, Acourate and Room Eq Wizard (REW).
  13. Hi everyone, after having invested in heavily appreciated "fanboy" gear like REGEN USB , CURIOUS CABLE and a KLEIN DAC (See more here) in my first year with the CA community, I have had a read here on the "biggest bangs for buck" threat, which convinced me, that doing something on the acoustic treatment of my listening room should be the next step before investing even more on hi-end equipment. Maybe investment on useful hardware like the minidsp UMIC or an antimode processor still does make sense, even a bass enhancement with a good sub could be adjusted to the plan. I have two B&W 805s in a "golden" triangle position with my armchair and with the speaker positioning I am quite happy so far (in detail dist. to sw 61cm dist. to rw 131cm, dist. to other Speaker 181cm, dist. to listening position 181cm (157cm from the straight line between the speaker). What needs to be improved - in my opinion- , is a noticeable hiss from the left with sharp high tones , i.e. the harp sound in "Postcard Blues" /Cowboy Junkies/Trinity Session or some of Jan Garbareks sax notes are sounding kind of unwelcome. Plus, even I had looked for a n symmetric and equilibrate approach to the room, I sometimes get the impression that the left speaker is more present than the right, could that be related to the different surface of the sidewalls ? And yes, having exchanged the speakers from left to right gives me the same phenomena. ;-) In addition my speaker have some limits in presenting the deep lows as their frequency response is 49Hz - 22kHz at +/- 3dB and the frequency range should be -6db at 42Hz and 50kHz. For starting this project on acoustic treatment , I'd like to ask: - Is it inevitable to commence with a measurement of the room nodes or are some first steps to improve rather obvious for some expert eyes? - Which would be the software of choice for the measurement ? - Does a Android app like "Audio Tool" on a Sony Z1 could do the job, first hand? My first ideas had been about having some canvas or molleton curtains for the last 150cm on the sidewalls, and maybe some absorber hanging from the ceiling, allowing me to reshape parts the ceiling in a more symmetric way, but only on the right side (WAF), as I would not like to cut too much the luminosity in the room. That said, would a linen drapery on the glass front of the window have some effect as well beside it translucent looks ? - Is there anything I could improve on the first reflection points ? I welcome your thoughts and inspiration on the subject Merci Tom
  14. Hi Guys, here is the video guide about the Devialet SAM and how to. I actually I didn't know exactly how the "uncle" SAM is working but this is giving some very nice details and step by step explanation how to. Of course it's the short version but you can imagine the rest;)) Any thoughts on this topic?
  15. What if your speakers have a digital input? does these make your DAC’s analog output redundant? Do you then just need a streamer or roon device with a digital out? in other words what if you send them purely digital signal? then all you need is a volume control right? not a DAC. How much do we know about the DAC stages in these new models? it seems odd to send a DAC signal down a balanced cable to another ADC - DAC system. thoughts?
  16. Any suggestions of a good parametric equalizer DSP component for foobar for free? Looking for one that allows adding as many bands as I want, with different EQ curves options (high pass, low pass, high shelf, low shelf, peak), Q setting, etc. Thanks.
  17. Looking for a MiniDSP SHD (with A/ D and D/A) in good working order. Please send me a PM if you have one you no longer need. Don't need the mike.
  18. * NO LONGER AVAILABLE * Description Mint Condition. Opened and inspected, but not used. This version has a hard-shell carrying & storage case (something apparently no longer included by miniDSP). The UMIK-1 is an omni-directional USB measurement microphone providing Plug & Play acoustic measurement. For use with REW Room EQ Wizard or other similar software.
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