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About Me

Found 8 results

  1. I have spent quite some time assessing PGGB, with success restricted to a tiny subset of early digital recordings. I have been happy to be confirmed by most respected people here that I’m not the only one to have my bad days/moments when I can’t enjoy music. I have stressed that our little games of comparing things is a factor that induces me in such bad moments. Now it’s time I stay away from AS for a while, things to do... But I wish to extend interrogations I started in my Deaf or Dumb thread about volume/SPL. I’ll be happy to read well thought comments. As a starter, I recommend @Confused's, the latest on Deaf or Dumb as I start this thread, with discussion of complex topics in mind rather than anathema or empty contradiction. Le Phénix des hôtes de ces bois wrote : « What's the goal? If it's accuracy, then we must know the volume the artists were playing at when the recording was made? Plus, if one's hearing isn't perfect, a volume adjustment could bring the level more in line with that live performance. What about Dark Side of the Moon, where there isn't a live performance? How should volume be set? It's a fool's errand and it makes no sense to specify a correct volume for listening.” As of classical music, Bob Katz wrote : “The late Gabe Wiener produced a series of classical recordings noting in the liner notes the SPL of a short (test) passage. He encouraged listeners to adjust their monitor gains to reproduce the “natural” SPL which arrived at the recording microphone. The author used to second-guess Wiener by first adjusting monitor gain by ear, and then measuring the SPL with Wiener’s test passage. Each time, the author’s monitor was within 1 dB of Wiener’s recommendation. Thus demonstrating that for classical music, the natural SPL is desirable for attentive, foreground listeners. » I have to say I’m with Bob Katz and believe I’m quite good at reproducing the natural SPL. Of course I heard the greatest pianist in large venues, not ideal, and it makes sense to have a recording of Richter slightly louder than in the concert hall. Jazz is tricky, I will call it mix bag for the moment. I certainly would not call Bob Katz a fool and his K system a fool's errand and yet he made efforts to specify a correct volume for listening and published a list of recordings ( https://www.digido.com/honor-roll/ ) mentioning his monitor position with reference to 0, 0 being for 500-2K PN -20 RMS (available on his site) yielding 83 dB/monitor at Listening Position. I own a 8 amplifiers, 1200 W total capable system, yet setting HQP at -3, I have to crank my preamp, at 63, a mere .5 dB from maximum volume and that is with +7 dB gain at the DAC’s entry ! @Miska I guess this is optimum gain with lowering in HQP ? or may I set 57 for it feels strange to be at max volume ? I also have the possibility to push entry gain to 20 but 7/8 sounds best. So we need amp power and speakers that might handle it… 500-2K PN -20 RMS says Bob Katz, it avoids bass debate but my 8K sure is above Katz’s target and it’s going to matter for pleasure… It happens that Doug Sax’s mastering of Dark Side of the Moon is part of https://www.digido.com/honor-roll/ And it’s 1 point for Chris. I mean I couldn’t listen long at Katz’s suggested volume, even after I created a set of filters rolling-off my treble to be closer to KATZ’s. 98 dB C peak is too much for me! I then played my favorite digital version, 5.1 MCH downmixed 2.0 and enjoyed it much more. It peaked at 94.5, very enjoyable, but I did not feel compelled to raise and up the pre gain to match the 98 reached with Doug Sax’s mastering. I preferred it with no HF roll-off and would have rather turned bass a bit down. However, success playing at Bob’s recommended volumes with Shirley Horn and Rolling Stones though for the Stones I had to use the filters rolling off HF. And certainly others this afternoon So, so far this a reminder that money should go first to basics : power and power handling capabilities. Then, I could agree that whatever pleases someone is the right volume for that someone. But this is also true of the mastering engineer : it sounded right, tonality, image, etc, for him with a certain frequency response and at a given SPL. When mastering has been done at enormous SPL we will have a bass shy reproduction at lower level, etc. Thank you Bob Katz for publishing reference SPL, probably enjoyed by you with monitors set to your target. I certainly would not qualify your tentative a fool’s errand even though I rather let my speakers’ HF FR be rather than make them follow your target and even if I have to sometimes turn down volume, because there are many other factors, damping, reverb time etc that most certainly make my room lesser than yours. So, I respect that enjoyment might be at different volumes but I certainly would not call Bob Katz a fool and his K system a fool's errand. The 90 dB at forte of most classical music can certainly be reproduced at home and Bob Katz’s suggestion of a Mix/Mastering system allowing clever compression to put pop music’s forte, when and where there isn't a live performance as reference, in the ball park of classical music’s, very welcome. Publishing Frequency Response and SPL at LP should be mandatory when publishing a review
  2. Hi, i've been diving into digital audio lately and came to this problem. All of my music is replaygained, meaning the RMS values are -18dB (foobar 2000). I have it set up in gain + clipping prevention mode. If i play a song thats louder without RG, well, it sounds louder. This means both my dac and my amp have to have some volume (voltage) overhead (at the same knob position in the amp case). My gear is Schiit Magni 2U and Modi 2U. The DAC can put out 1.5Vrms max. Is there any way to atleast approximately calculate what RMS voltage numbers should i expect from my dac and then from the amp with -18dB replaygained music? I have done some small calculations, using this calculator and some basic equations. -18dB RMS signal should be 0.125 voltage gain ratio, so 0.125 times 1.5Vrms is 0.1875Vrms. If i watch peaks in foobar, most songs peak like 5-7dB under 0, which corresponds to 0.45-0.55 voltage gain ratio. Again i multiply that by the dac's max Vrms and continue down the chain with my amps and whatnots. Am i doing it right? I know there is a lot of relative subjects in this, but i am trying to get atleast approximate (+-1Vrms) number at the end of the amplification chain. Thanks a lot Dave
  3. Hi Last week I bought a new CD. Ripping it, I ended up with FLAC files with considerable lower volume as files I ripped from other CDs I have. I guess, this is perfectly normal. But I hate it, as this means I have to adjust the volume if I listen to some other album or if the music is part of a playlist. Is there a way to change the volume of the ripped files without losing quality or dynamics? Currently, I'm using a Mac with fre:ac to create FLAC. I'm happy to rip the CD again with some parameter or other tool or to use some utility to change the volume post-ripping.
  4. Hello everybody, I'm new on this forum, I hope this topic is in the good section... I would like to control an external volume attenuator via my MPD clients volume control function. This analog attenuator is placed after the DAC and is controlled by 4-bit input lines (for 16 positions). I need : - to manage volume value in MPD (to hold the current value and to change it with MPD client command) - to keep digital flow sent to USB DAC at max level (0dB) - to outside the volume value from MPD to 4 discrete output lines of RaspPI3 How can I do that ? Jacques
  5. I've downloaded a dozen or so High Def albums from HD Tracks and I notice that I have to turn the volume up on my Cambridge PreAmp almost to the max to get the sound at the same level as standard files 44/16 at -20db. Any thoughts as to what might cause this?
  6. I have an SPL meter in my listening room and an Apps on my iOS devices. Periodically when listening I check the SPL to see how loud it really is. I am curious to see if anyone else does this as well, or maybe I am just neurotic.
  7. Hello folks, this is my first post around here. I am new to computer audio. Recently I put together this desktop system: - Macbook Pro Retina (Late 2013) >> Audirvana Plus 2 >> AudioQuest Pearl USB cable >> DAC TEAC UD-301 >> XLR cables >> Active Speakers JBL LSR305. I am pretty happy with it, since I used to listen to music in cheap earbuds, and now, with less than 600 bucks, my experience have changed so much! I have a few questions for the community: 1) I am using my DAC to control the volume, but in the transport area on Audirvana I can't see the volume knob representation. Is this normal with the TEAC? 2) I use macOS Sierra 10.12.1 in my MBP, and when I try to enable Direct Mode plus Integer Mode, there is an error when trying to playback music. Just Integer Mode enabled is fine, no problem. Is this normal? 3) Will I have an audible improvement by upsampling Tidal 16/44.1Khz music to 176.4Khz using the Teac feature? Or should I do it within the Audirvana Plus software? Or not at all? 4) Lastly, what setting tweaks do you recommend on A+ for my hardware configuration? Thank you for all the useful info I got from this community already! Diego
  8. Hello members. I am a new and glad to be here. Hopefully someone can help me out I have gathered up some entry level gear which suits me well, as I travel a fair bit. Currently I run am running Foobar2000 to an ARCAM rPAC DAC, to a pair of Grado SR60’s. My source consists of a Lenovo Thinkpad, Core i3 processor with, 4GB ram. All M4a’s are on an external HDD, connected via USB. For entry level the dynamics of the setup are great (have to love the SR60’s). However I find the integrated headphone amp in the DAC lacks the ability to drive the SR60’s to high volume - the amp seems to run out of steam. The volume does get loud but is far from the “blistering” level that I have been told it should reach, (not that I would listen to blistering volume). So far I have: 1. Checked all of the volume settings (I know of) with no result. 2. Turned off the power management on all USB Hubs, no effect. 3. Attempted to bypass Windows KMixer by loading an ASIO driver into Foobar, hoping this will remove any volume restriction in Windows 7, (which I read has volume output issues). Foobar2000, output now set to ASIO, has mapped the left and right channels (as I can hear them) but the rPAC DAC is still showing a standard MS generic USB audio driver. This would lead me to believe that the music is still being streamed through KMixer. And the volume issue remains. Any advice would be appreciated. Thanks
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