We Are The Music Makers (Musician's Thread)

O snap I almost forgot about this thread with all the election insanity.

I want to say “I hate closed back headphones so they’re all gonna be equally shitty” but that’s probably not what you want to hear.

I’m also an AKG fanboy so I’m also inclined to say “I’m also an AKG fanboy so they’re all gonna be equally shitty” which uhhh.

Hot damn this is starting off as a gloriously unhelpful answer.

Of those the Beyers are the obv choice, the Senns obv last place, though I’m not familiar with the Sonys.

Here’s the AKGs I had in mind:

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Are these both real-time digital plugins?

Convolution reverb is the best but a big problem has always been that it’s near impossible to do in real time, and a bigger problem is that some hardware and most plugins advertising ‘real time’ weren’t actually doing true convolution reverb. You’d still get a great sound and could usually only tell the difference in an A/B, but still.

I’m loading them in this Linux guitar amp simulator called Guitarix and hearing it back with like 6 to 12 ms latency. This is what it’s using to run all of the convolution:

Jconvolver is a Convolution Engine for JACK, based on FFT convolution and using non-uniform partition sizes: small ones at the start of the IR and building up to the most efficient size further on. It can perform zero-delay processing with moderate CPU load. Jconvolver uses the convolution engine designed for Aella, a convolution application for reverberation processing (to be announced later). This distributes the calculation over up to five threads, one for each partition size, running at priorities just below the the one of JACK’s processing thread. This engine is a separate library that will be documented as soon as I can find the time.

Main features:

  • Any matrix of convolutions between up to up 64 inputs and 64 outputs, as long as your CPU(s) can handle it.
  • Allows trading off CPU load to processing delay, and remains efficient even when configured for zero delay.
  • Sparse and diagonal matrices are handled as efficiently as dense ones. No CPU cycles or memory resources are wasted on empty cells in the matrix, nor on empty partitions if IRs are of different length.
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@6ix

This guy is like the Steve Kornacki of headphones. Agree with him on Senn HD6xx for reference on basically every point so inclined to believe him overall. He’s got nothing but HYPE for the K371. I’ve owned some AKGs in the past and they never seem to fit snug enough is what I’m concerned about. Love the fit of Senn 5xx/6xx more than anything.

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How does composing for large ensemble usually work? I know nothing about classical other than doing these counterpoint and voice leading exercises when I took AP music theory twenty years ago. That was all like four voice SATB though. When I’m listening to a full symphony orchestra (which is almost never), what is going on there in terms of parts and counterpoint? Are all 17 violin chairs usually playing in unison or is there weird stuff going on within instrument groups?

Grunching from here but if this is not already in production then surely the UP Xmas Album is motivation enough

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Typically you’ll have 1st/2nd/3rd parts for most instruments you have multiples of, google the score of a mozart symphony or something, tons of orchestral music is public domain. As to how it works out with parts and counterpoint, listen to stuff. the rules of baroque voice leading are still there but become treated more like guidelines as time goes by.

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ES-8 replacement arrived finally. Dude said he couldn’t replicate it crashing but sent me one to get it dealt with. New one arrived and works perfectly and been running signals through it since early afternoon without any issue. Didn’t even reinstall the driver. Fucking computers.

Worth it just because of the 80s. Just as good as info as there ever was also.

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Basic DAW questions:

I tracked about 2 minutes of guitar improv hoping I’d get a usable riff or two from it. So assuming there’s 1 or 2 things worth keeping in there, what’s the best way to excise those parts with good break points and bolt them to the grid? I assume there are tools that make this easy but I’m a dinosaur that wants to do it manually.

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I do that in Audacity or Ableton. Usually Ableton. If you played it at a good solid tempo then just make sure you zoom in and get the clip to start right on the first wave of your strum or whatever. There are tutorials all over about doing it in Ableton, not sure about other DAWs exactly.

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So it comes in about a half note pickup (slide) before the bar. I’m assuming most DAWs have a way to MARK part of this wav that I wanna lock to first beat of a measure or something? Maybe what I’m imagining doesn’t exist but seems like it would also have some kind of transient finder that could cue these points of the wav up so I don’t have to do a 10,000 scale factor and edit it manually?

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Maybe this is just #Linux.problems but it’s been such a gigantic pita just being able to reliably get any sound at all patched through without breaking something that it all seems very Microsoft Flight Simulator to me (I could never get the plane off the ground).

In Ableton you can pick markers with a pickup and a 0:00 marker and even warp it if your timing is off (sort of like quantizing) and adjust marker points and stuff to lock it in further. I’ve really never messed with it a lot but know the basics. DJs use it a lot to blend tempos when mixing songs and stuff I think.

Time changes get tricky though then. I have an Ableton file of Friends by Led Zeppelin with each of the intro time changes busted into separate samples and have been meaning to remix it somehow but haven’t yet.

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I’m likely horrible at explaining this stuff also. I assume people know what I’m talking about when I make it short as possible and don’t explain things like I should compared to people who do tutorials and I suppose I also assume people know my fragmented statements and stoner English.

I think it’s more like I haven’t really fooled with this stuff much other than for rigging up some guitar backing tracks in 20 years. I don’t really find tutorials to be too helpful at all. I wasn’t kidding about MS Flight Simulator. Another example is 3D modeling software which is what I’m basing all of my assumptions on. I taught myself Blender from scratch and it’s such a giant AIDS ball. I would literally complete one step like “extrude a cylinder” and then have to stop and google the next step. What I learned is there was always some shortcut to do what I wanted but they were like 8 finger hotkey combos you had to memorize.

I’d just say that you can zoom in a long ways and get it clipped right when your first noise is made. I’m not sure about marking for pickups though in Audacity or exactly what you want to do after. In Ableton, you could like program midi drums or whatever to the tempo. Sorry if I’m misunderstanding.

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This guy has a condensed version based on the video.

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I think I can articulate my point better now: watching this dude do all of that mouse dragging looks really inefficient. In other complicated software I’ve used, there’s shortcuts for almost everything. Seems like a pro could trim that without touching the mouse? Maybe I’m wrong.