Jump to content
Computer Audiophile

Search the Community

Showing results for tags 'equalization'.

More search options

  • Search By Tags

    Type tags separated by commas.
  • Search By Author

Content Type


  • Reviews
  • CA Academy
  • Audio Shows
  • Bits and Bytes
  • Digital Vinyl
  • The Music In Me


There are no results to display.

There are no results to display.


  • Downloads
  • CA Sample Club's Files


  • Equipment
    • General Forum
    • Music Servers
    • DAC - Digital to Analog Conversion
    • Disk Storage / Music Library Storage
    • Networking, Networked Audio, and Streaming
    • Headphones & Speakers
    • Software
    • DSP, Room Correction, and Multi Channel Audio
    • iTunes and Everything Apple
    • Article Comments
  • Music
    • Music Downloads & Streaming
    • Music in General
    • Music Analysis - Objective & Subjective
    • In Memoriam
  • Sponsored Forums
    • Sonore (Sponsored)
    • HDtracks (Sponsored)
    • UpTone Audio (Sponsored)
    • Highend-AudioPC (Sponsored)
    • Abbingdon Music Research / iFi audio (Sponsored)
    • Klipsch (Sponsored)
    • Superphonica
  • CA All Access
    • Buy & Sell Audio and Computer Components
  • Allo's Allo Topics
  • CA Sample Club's Topics
  • CA Sample Club's Q & A

Found 4 results

  1. 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.
  2. 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).
  3. 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.
  4. 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.