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Found 7 results

  1. For details and chances to debunk caveats errors etc, please refer to If you have suggestions, you're welcome Results : Bernie Grundman uses, or at least used to master Joe jackson's Body & Soul in 1984, the Bruel & Kjaer Kevin Gray, at least for his SACD of the above, used straight 0 dB to 1K, downward straight slope from 1 to 20 K, 20 K -3 dB (projected, he lets, as I do, the natural parabolic fallout of his speakers above 10K) Alan Yoshida seems to use the same (Somethin' Else by Cannonball Adderley for HDTracks) Shawn R Britton to master Steamin' by Miles Davis (MFSL SACD) : same as above but -4.25 at 20 K Bob Katz (claims): same as above but -7.26 @24000 Steve Hoffman : oh my ! read in the above quoted thread Please think carefully : are the versions you prefer the absolute very best (could be because of source, transfer...) or the best match to your own in room response ? What if there's no alternative version and you're far off ? I encourage to have at hand several in room responses to match the engineers' Mind that there are several dBs difference that no usb cable or LPS will compensate
  2. As suggested by the title, this is no substitute to much admired @Josh Mound ‘s TBVO. However, it’s a follow-up to comments/questions I appended to TBVO: Can we deduct the in-room response (and/or “ear”) of praised mastering engineers from comparisons? Isn’t our “in room” response (“” meant to include headphones) biasing best version into best fit? TBVO thoroughness can’t be beat and I don’t pretend I address source tape quality, transfer (machine, baking, ADC, etc) nor do I address dynamics, levels, compression, etc, I don’t even compare peaks while the attached comparison goes to show than peak treatments can display bigger differences than averaged power comparisons.
  3. (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
  4. I'm thinking of purchasing a Melco server in the next few months but just had a thought. I currently EQ music via J River and with the Melco, or other servers, you don't go through a computer with J River so is there a way to apply EQ in the realtime runstream? I don't want to record it into files since the EQ is just for headphone listening. I just purchased a Ayre Codex so don't have a option of a headphone amp with EQ. The best solution world be software running inside the Melco runstream.
  5. My system (so was my relative mania) was laid to rest for several years. Since my (once was reference, very expensive) cd player refused to wake up, I turned to computer based audio (using my MBP Retina 15”). I used to hear differences based on the cleaning fluid applied on my vinyls (cleaner grooves are wider grooves; i.e. the bass extension provided by Smart - Premier Pre-Cleaner, even on brand new records, is amazing); I could not resist exploring sonic impacts in the computer audio realm. Here are my findings: Respect endianness (relates to file/processor interaction ; further readings might be : The FLAC Audio Format | Start at Zero and aiff). Apple discretely turned AIFF little endian: it matters. FLAC is big endian while Intel processors are little endian. WAV is the uncluttered little endian file format: convert your files to wav just before playing (store flacs for size & tagging). It takes seconds with XLD (Mac) and it’s free (though we should all donate). Yes FLAC offers lossless compression and all your bits are there; but it doesn’t sound as good as AIFF-C and WAV is even better. Because of endianness. To exit any process that can have a hold on a file, including the Finder, seems best practice. Minimize CPU load Yes, you already know that and quit all unnecessary applications etc, but… Using Audirvana +, I heard horrors when streaming audio files from my Time Capsule, because of longer processing times keeping the processor highly active while the track had started playing. Google “RFI” and you will know why it sounds awful. Now I first import on my ssd and wait for the cpu load to come back to zero or so before I launch a track. I wish Damien Plisson would implement an option allowing us to preprocess and load a whole album in RAM, ahead. I think that the sequential processing from track to track is a major design flaw that defeats part of the “Full memory play” promise; in the “Tracks loading, decoding, format conversion complete before playback to ensure best Sound Quality” claim, there should not be a “s” to (more than a few)tracks. Check your Activity Monitor. For the same reason of almost continuous cpu load and thus, RFI, playing isos was also unbearable to my ears. Take the time to extract dff from your isos with sacd extract. EQ & deess I love my music scary, to involve my whole skin in the listening experience. I can comfortably enjoy the 2 drums + 2 basses interplays in Daft Punk/Moroder’s Georgio and be worried for my things and walls by the deep electronic notes at the very end: I don’t listen through headphones. And thus have room interactions. I enjoyed Amarra very much, over Audirvana, until I could go further with a Audirvana 1.54/iZotope Alloy 2 combo. My eq is based on Amarra’s setting for classical with an emphasis of the deep in the 800 Hz region, maybe because of my room. Amarra’s : My EQ : My deesser setting : the whole idea is to apply a dynamic eq that never interferes with acoustic music (never saw my deesser triggered by any kind of classical music). Do I interfere with the original mix, the artist (engineer) intention etc ? oh yeah ! So do you anyway by the mere simple act of playing back music on your system with its idiosyncratic characteristics. I feel perfectly comfortable, subtly and surgically, smoothing vocals when they are shrieking on a reference grade system because of a mix/mastering for cheap record players years ago or smartphones nowadays. Give it a try: you will crank the volume up and get a much more concert like, realistic, playback. Think twice about DSD Especially if you’re a Mac user. I purchased the TEAC UD 501 partly because of its DSD capabilities. However, I now convert to PCM through Audirvana (cleaner to my ears than Audiogate, though with a loss of bass weight). Mainly, it allows me to benefit from iZotope Alloy 2. With a slightly different eq to compensate for the loss of bass weight : Furthermore, though Audirvana can feed the TEAC via DoP (and bits are preserved, OK) it does not automatically detect the device as DSD native though it is. I wonder (read : I doubt) if the global process is optimum. Anyway, I never liked what I heard feeding the TEAC with DSD from the Mac (DoP mode); I enjoy very much my converted eqed dff files… Choose the right filter There might be a rational for playing back pcm through the TEAC with a sharp filter for redbook, a slow filter for 96 K, and no filter above… Well, now we get bored, don’t we ? I turned the upconverter on and the PCM filter off and am happy… ("Minimal Phase filter", see : Archimago's Musings: MEASUREMENTS: Digital Filters and Impulse Response... (TEAC UD-501)). Cheers , Le Concombre Masqué P.S. : I have long been an analogue enthusiast and record collector. I have been blown away by what Paul Stubblebine did on Waltz for Debby for HDtracks (I own a test pressing of the Acoustic sounds 45 rpm ; always found it much inferior to its SatVV sibling though) or by what Kevin Reeves did on the Velvets for HDtracks (I own an unbrushed Eric Emerson cover very first pressing).
  6. dezfretz

    Amarra EQ

    Playing files of a "decade" of test tones from the Stereophile CD and measuring SPL with a Radio Shack sound level meter, I have a persistent 10 dB peak at 200 Hz at the listening position. In Amarra, I selected "Bass Reduce" and three sliders appear. I typed in 200 Hz for one freq and zeroed the attenuation on the other two. Q is defined as center freq divided by bandwidth. Bandwidth is defined as the 3 dB-down freqs on either side of center. I previously measured nearly flat at 160 Hz so I chose 180 Hz for the 3 dB-down point and 220 Hz on the other side, so bandwidth is 40 Hz. Q, then, is 200/40 = 5 which I typed-in, setting attenuation for the 200 Hz center freq at -10 dB. At the listening position the SPL now measures flat for the 200 Hz tone -- and the system sounds WAY better on music: Voice articulation and imaging are dramatically improved, and overall bass response sounds "right." I reported my experience to Sonic Studio Support and they validated this approach. They noted that the other sliders could be used to simultaneously deal with similar peaks (or troughs) at other frequencies.
  7. I've assembled a modest entry level desktop listening station (Synology 212j NAS, ALAC files, MacBook, iTunes, BitPerfect, Dragonfly, Sennheiser HD-650) and re-ripped much of my CD collection to a lossless format using XLD ensuring good rips. Now I've discovered that several of those CDs sound terrible at satisfying volume, probably due to compression, as I encounter this on indie and classic rock (although I may find it in other genres, too). I acknowledge the principle of high fidelity to the source, but the sound in some instances is sufficiently unpleasant to justify violating this principle. Some of these bad recordings are of my favorite music, with no superior recorded alternatives to the CDs I own, and I want to enjoy them as much as I can. What can I do? Insert an equalizer into the stream? Where? Hardware or software? How can I easily toggle it off for the good recordings? Although I enjoy tweaking and customization as much as the next guy, I actually wouldn't mind a default setting that addressed the presumably common harshness I'm encountering. The goal here is to enjoy the music, not add another component to obsess over. Not that I mind obsessing over components.
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