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Sadly not, HQ listening is done at work and nobody else is allowed in right now!
But skipping quickly between intros is pretty telling.
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Yes dB poweramp secure. Read speed set at 40x, compression set to default, FLAC 44.1/16. Getting through roughly 10/hr, album art and track names take about 20secs to retrieve. I don’t bother re-ripping inaccurate tracks, the ones I’ve tested play back fine, although I’m sure I’ll find some blips and thumps.
The Roon metadata would be nice, but Audirvana is a stop-gap while Im playing direct from the laptop. I’ll be rendering through Volumio once I bring my streamer in which is as basic an interface as they come. Then when I set up some sort of server I’ll probably buy Roon.
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Just re-reading what you've written here, we must be using different versions of secure. On my dbpoweramp it goes to a very very slow re-ripping stage if after 2 passes it doesn't have the correct checksum. Following that if it fails it doesn't write the file and delivers a warning. I can go back and burst rip that file.
This box set comes from the 90's and got played in car stereos, I'll probably have to get out the plastic polish for them.
Last summer I managed 700 in a week using XLD. That was pretty much every waking hour though.
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Yes I think so. I did have that thing where it displayed frames being rewritten at an absolute crawl and I think I disabled it.
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Thank you. I guess start at power sockets and work backwards?
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Like with one of these?
1 Attachment
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I'd just do it by eye. You might see something is obviously loose or feels loose.
Also, if it wasnt doing it before repair, and is now. Then it's something to do with the repair (simplest answer) so check that. The amp is grounded inside, so take the case off (with power unplugged) and check in there too.
If non of those fix it, then needs more investigation.
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Thanks, I'll take a looksee.
CD player plug was not 100% engaged, we'll see if that changes anything.
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Ok cool, yes PCs too, but you’ve entirely proved my (imperfectly argued) point which is that there is a huge amount of juju around the ‘purity’ of digital sources, usb cables and power supplies when applied upstream of the DAC on playback. I’ve even seen claims that the speed you rip a CD at makes a difference 😂
The most important thing for quality is how music is captured, mixed, processes and laid down in to its final recording. If mastering is done on a computer of some sort, and in most cases it is, then why is a computer unsuitable to perform the very simple task of extracting the data and buffering it to a digital stream?
I’m not doubting @StevePeel’s ears but equally I think I understand reasonably well stuff like clock syncing and jitter. When being mastered, most devices in the studio containing an ADC will have their clocks synchronised, but that’s because you’re dealing with multiple different embedded OSs, ADCs and connection methods (optical, USB etc). On playback however, clock syncing / jitter is rarely the cause of poor audio unless you’re dealing with a very poor quality stream converter or DAC. It’s far more likely to be caused by poor galvanic isolation or software drivers.
TL/DR: perfect streaming with Sub -120dB noise is quite achievable on a (properly functioning) computer, windows or Mac.
Sources: http://archimago.blogspot.com/2013/04/measurements-laptop-audio-survey-apple.html
https://www.audiosciencereview.com/forum/index.php?threads/auralic-leo-gx-dac-clock-review.11001/
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No-one sets up their studio with any of that stuff in mind.
Have you ever been in a studio whilst it's in use, or conversed with sound engineers?
Noise floor... it's all about reducing the noise floor. You're going to have a lot of channels mixed together each bringing their own background noise. It doesn't matter so much where in the chain you go from analog to digital, that's a lot about trade-offs on control, tooling, sound... but it's noise floor.
The most important thing for quality is how music is captured, mixed, processes and laid down in to its final recording.
Yes... reduce the noise floor, and then compress it out of the final mix. If the noise floor rises into the final listening range of the composition nothing can help you.
Oh, I should include a picture...
Source: https://libremusicproduction.com/answer/noise-floor.htmlAll that hiss and noise, even when not audible on a single channel, combine to create noise that when you have enough it rises into the audible range. This stuff is like magic, if you think of ghost notes where several other notes are combined and give rise to hearing a note that wasn't played... well this happens with noise too, individual samples and channels may have no perceivable noise, and yet combined in certain ways they give rise to notes that don't exist but can be heard... fighting the noise floor is hard. I've been in a studio with PJ Harvey and Flood and saw them spend the whole day just trying to find which sample was causing one minor symptom of raising the noise floor.
The academic and hypothetical purity of any single component is just irrelevant, it may be relevant for reproduction of audio and processing of single signals (games, VOIP, etc)... but it's just totally irrelevant when it comes to recording, production and mastering... because so long as the word clocks are synchronised everything can be lined up just fine.
The battle in a studio, the concern for those who record, is combining their many channels together without increasing the noise floor.
Sheesh, even the links you cited mentioned "reproduction" and "playback"... single source, and nothing to do with music creation, production or mastering.
There are some areas I'll defer to you on, but this one where you specifically stated "most music is produced on a Mac mini / iMac"? Nah, you're talking out of your arse quite loudly.
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I'm guessing your studio experience was some time ago - 24bit recording (and the 32/64bit fp audio engines) means there's essentially no noise floor in the digital realm. Obviously if you're capturing analog sources then getting the gain structure right and minimising signal chain and source noise is part of the game (shit in = shit out) - compression raises the noise floor [of an individual recorded track] in most cases.
Other than that you both seem to be arguing the same point - audiophile 'purity' is nonsense.
The only place where really high paper-spec gear is used in music production is in mastering and I suspect most of the magic there is in the people carrying out the work!Edit: and I think you're both right about who is worrying about this - friends in the industry just make music and use their ears, it's dorks on places like SoS who obsess over jitter and speak authoritatively about £30k stereo converters (at least until you realise they've never made music professionally).
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@Velocio
You seem to be strongly agreeing with me in that mastering and production is where audio quality is defined. I understand the concept of noise floor pretty well, thanks. I didn’t include it as i didn’t think anyone was seriously suggesting that converting a stored file to a bitstream introduces actual noise artefacts. For clarity, as @Dramatic_Hammer already said, it doesn’t.Next point. I said, and you replied:
When being mastered, most devices in the studio containing an ADC will have their clocks synchronised
No-one sets up their studio with any of that stuff in mind.
But then you also go on to say
so long as the word clocks are synchronised everything can be lined up just fine.
So we also agree that during production it’s important that clocks are synchronised, yes? Good 😀
Finally:
There are some areas I'll defer to you on, but this one where you specifically stated "most music is produced on a Mac mini / iMac"? Nah, you're talking out of your arse quite loudly.
This is a bit ungracious IMO. I’ve already conceded that in my original post I made my point a bit glibly.
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Put on Mark Ronson's Version album recently, expecting it to sound great. It's awful, really compressed and boxy. The production or mastering or something is rubbish. How come he's a super-producer?
Kiwanuka or Tony Allen / Hugh Masekela on the other hand, beautifully recorded.
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Hah! Fair enough - tech hasn’t moved on massively since then but a whole day chasing ghosts is nuts!
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They'd created a crowd for the background noise on Dollar, Dollar on PJ Harvey's Hope Demolition Project album. The crowd wasn't a signal background track, it's actually about 300 tracks cleaned as best as possible and arranged to construct the hot city background noise. At some point, some blend of those tracks was creeping above the noise floor and affecting how the vocals from PJ Harvey and the wind instruments they were experimenting with were sounding.
Honestly, I couldn't hear it at all. I couldn't hear the flaw, I couldn't hear the fix... I kinda agreed with it sounding different but couldn't have said why.
Flood knew though, he led them through all of the groups of samples methodically, blending them in in different combinations until he'd isolated it. It was a group of samples that were supposed to bring in the heat of the day, and removing those satisfied all of them. By itself it was fine, mixed with a lot of others it was fine, mixed in the final blend it was a no from them. Did it sound better after? I couldn't say, my memory of the before was gone and I'd listened to their reasoning too long. Chasing ghosts is an accurate description for it.
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What I'm picturing happening with streamming transports, jitter and interference pre-dac, and this is not based on any technical experience whatsoever - just trying to piece together all the contradicting arguments I keep reading on the subject is this: Audible interference and distortion is at the extreme end of the spectrum and is not really to be tolerated. Beneath that, the small little details - the decay of a piano note, the room reverb in a recording, those subtle nuances that transform listening from just hearing music to the experience we're in this for, they're the bits getting lost from source along the way by the miniscule timing errors and uncorrected dropped data. The effect is probably small compared to what gets lost in the DAC and in the speakers, but it's a contribution. I may have totally misunderstood the theory, but I have compared digital sources in an otherwise identical system and they sound different, so IME the argument that anything pre-dac can't make a difference doesnt hold up. I'm not going to publish a peer-reviewed paper in Your Mum about it but I've heard it.
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I've heard it
On the grapevine?
Auditory perception is psychosomatic. When the voices in your head tell you to do something, you don't have to obey them 🙂
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Audiophiles hifi appreciation thread old and new
Posted by
@coppiThat
Have you tried Yate yet? I got into it because I'm creating a lot of needle drops (for want of a better term). By a lot I mean over a thousand in the space of a few years (1.2tb of 96/24 audio) and of course tagging these files is more complicated than ripping from a disk as you start with nothing.
It's a long way ahead of anything I've used before, it can also do things like change the folder icon to the album artwork and operations like that can be automated or batched.