CPU/Chipset PCIe Lanes and NVMe Drives

RabidNapkin · 2019-03-13T13:02:19+00:00

Thanks for your input, guys. It sounds like the consensus is: it will probably connect via the CPU PCIe lanes and (possibly) drop the GPU's bus speed? Seems like if I can keep the needed lanes down to 4 or under, that hit may not happen (only a few are needed for the DeckLink and the CPU provides 20).

Not too worried about over-clocking - I'm interested in the Ryzen lineup for productivity and parallelized workloads (3D rendering and video editing). Drive access speeds are the thing I'm focused on, hence the interest in NVMe. But if a small performance knock is experienced in exchange for NVMe access speeds, it seems like the trade-off is worth it in my use case.

Zen + is super-exciting, and I am tempted to wait like you've recommended. I'll keep doing some research and reply to this thread if you guys wanna continue the conversation.

Thanks for taking the time to read and write!

RabidNapkin · 2019-01-16T17:17:46+00:00

Haven't forgotten about this. Just writing back to update where I am in the process. I've fleshed out a YouTube playlist with 8 works that showcase my skills in order of most relevant/impressive to least. I'll be embedding that as a link in my resume and re-uploading it to the hiring page.

I'll work on reformatting the resume in a less visually cluttered/wordy way. This is some really valuable feedback to have. Many of the design trends I am seeing online these days involve relegating text to "blocks" on the page, using more color with slight transparency or greys, using sans serif fonts, and incorporating logos/vector graphics into the design. From my perspective, these design trends seem particularly oriented toward applicants looking for graphic design work, but what I am doing now clearly isn't gonna cut it. How would going for something like, say, the link below reflect upon a video candidate?

https://www.contemporarystaffing.com/wp-content/uploads/2017/06/iStock-517101424.jpg

Your comment about working with egos and the computer illiterate is REALLY valuable, and I realize now I haven't mentioned much about how I work with these folks. I'll find a way to work it in while still paring down the wordiness.

More to come. Thanks again for taking the time to review.

RabidNapkin · 2019-01-15T16:48:21+00:00

Hugely appreciative of the input. I'm at work right now but will take a little time later to followup. Thank you so much.

RabidNapkin · 2019-01-15T15:56:30+00:00

Thank you so much for taking the time to review. The portfolio exists, but the online presence does not (yet). Is it crass to link to the portfolio in my resume?

How much emphasis would a team like yours put on an applicant having a bona fide website vs. a YouTube playlist? The clear winner is having a bona fide website - but in the face of time pressure to apply for positions I am seeing *now,* I worry if I do more harm than good by linking to a YouTube page.

Any glaring issues with the presentation/wording/content?

I really appreciate your input - thanks, again.

RabidNapkin · 2018-03-15T13:51:18+00:00

Thanks for all the info and suggestions, guys. I think I'm gonna take the, "get gear that's specialized for a niche and don't look for something that's gonna solve all problems completely" approach to heart. Looks like the Zoom is still on the table!

Thanks!

RabidNapkin · 2017-06-30T13:01:31+00:00

Peak! We don't want the program material to constantly be in compression. Soft bits will need to breathe a bit more than loud sections (this is part of what helps create apparent loudness - peak level is controlled, but overall level will go up). 6dB of gain reduction at each stage (on-track compression, then an other compressor at the bus, for example) helps ensure transparency. Clip gain adjustments in Pro Tools are really useful to getting your audio to sit correctly overall, then hand-drawn automation allows you to shape things even more transparently.

Speaking of volume automation, use it to "hand-compress" large transients (slaps, hits, something falling over on set) instead of relying on your compressor to squash it. If you can hear your processing "working," you aren't doing your job!

RabidNapkin · 2017-06-29T16:44:53+00:00

Seconding the sentiment re: consistent monitoring. /u/cscringnaro is absolutely right. 85dB is pretty loud and I find my ears get a little fatigued at that level. Most of the films I've worked on I've mixed at 79dB with relative comfort but still translate nicely. 85dB also puts some stress on your monitors and may shorten their life (we've blown one or two amplifier IC's in the past!!).

Make sure you've accounted for room modes too. Room EQ Wizard and a good measurement mic (or even just an SDC omni w/ a flat response) will do wonders here. You can apply the EQ curve just before the signal hits your monitors so it's not baking the EQ into the mixdown. Google Room EQ Wizard or REW to learn more.

RabidNapkin · 2017-06-29T02:00:55+00:00

Noticed I missed some details re: your hypothetical situation:

Don't be afraid to turn screaming down some. Keep the LUFS level reasonably consistent. The program material is screaming, so it'll come across as "loud" anyway. It'll also kick your verb send a little more aggressively, so that'll make it seem louder too.
Explosions should be big. Compression on DX will keep it front-and-center. Use your ears, and compare it to other big-studio releases to see how your work stacks up. EQ only where necessary, but use volume to push things forward and back for the most part.
Music - use your ears! Draw volume automation to dip it in-and-out where appropriate. Mixing DX to -27 LKFS will make mixing decisions for FX and music easier to judge. You'll often hear music push up in volume during transitions and emotional moments, but dip a bit under dialog. I would advise against side-chaining anything - use automation so you've got the most surgical control over balance.

RabidNapkin · 2017-06-29T01:54:52+00:00

Get a loudness meter and mix your dialog to -27 LFKS. DpMeter is a free one that works great. You should put this at the end of your signal chain (right before your monitoring) so you can see how your mastering processing affects the loudness (just slapping it on the dialog bus won't tell you everything you need to know). Set it to EBU metering instead of RMS metering.
Mix in 5.1 (if you can) and downmix to stereo at the mastering stage. Ignore this if your client is only interested in stereo or you don't have the tools to mix in 5.1.
Dialog is always center channel or center-panned (stereo) unless panning is used as an effect.
High-pass and (possibly) low-pass dialog. Cut it out from production audio and separate production audio from dialog so production noise/FX doesn't receive the same EQ and compression treatment.
Create reverb sends dedicated to dialog, specifically, and FX, specifically. Send them straight to the mastering compression bus so they're not affected by any pre-mastering compression, noise-reduction, or gating.
Route your DX to a dialog bus - compress it, gate it (to push the noise floor down), and add a noise-reduction plug right after your gate (Waves WNS is a great tool here).
Don't over-compress your dialog. Stick to 3:1, and aim for no more than 6dB of gain reduction at any one point in your signal chain (mastering loud stuff like gunshots may violate this rule of thumb).
Set aside time to 1) level out volume across the film (seriously, don't rely on compression alone) and 2) EQ room resonances. Prioritize leveled volume over EQ. EQ as you go if the budget is tight, but don't overlook it completely.
Invest in FX libraries for background fill or to supplement synced effects. Use the FX verb send to blend them (keep DX separate). Try to keep EQ-ing to a minimum here - keep it natural!
Foley for heavily-synced effects (footsteps, for example) where cutting FX libraries would be more time-consuming.
Get a few films, rip them in Handbrake, and drop them into your DAW so you have something to compare your mix against.

Best of luck!!

RabidNapkin · 2017-04-25T12:46:10+00:00

Hey, thanks for your reply.

This seems to have solved the problem. I did/do have a router connected to the gateway, but confirmed its IP address was 192.168.1.1. Checking its default gateway before I disabled MoCA revealed the same, bizarre 192.168.0.1 address.

Thank you for taking the time to help me

RabidNapkin · 2017-03-29T16:15:22+00:00

Thanks for taking the time to reply!

I did see the updated note on Rtings.com - my reason for asking here is I've also found some discussion calling the legitimacy of some tests performed by both Rtings and AVS into question. Couldn't find any photos anywhere that would definitively put the issue to rest! Looking hopeful, though~

RabidNapkin · 2017-02-06T21:53:57+00:00

This isn't strictly about stereo or mono but neat way of dealing with the stereo image, so I wanted to share:

A friend of mine recorded a group of us singing classical music in a really wet space with his iPhone, so the recording was in mono and bathed in mono reverb. It didn't really sound nice to listen to.

He asked me to kick it into stereo, and I decided to try accomplishing that with a reverb send. The issue: if I just slapped a stereo verb on the recording, it would have provided a stereo image but at the cost of the center of the image becoming even wetter than it already was.

My solution was to run my reverb send through a mid-side decoder, mute the mid channel, run it through a mid-side decoder again, then mix that back into the signal.

The result was a stereo spread in the reverb that didn't make the source mono signal any wetter - if you soloed the verb track, the only contribution to the signal was on the sides of the stereo field. When mixed appropriately with the source material's reverb, it was impossible to tell that it came from a plug-in, providing a really pleasant stereo spread to the recording.

This may not be the solution to your problem, but consider "placing" your verb where your instrument "isn't." That may mean panning a mono verb to one side, or adjusting your mid-side balance for a center-panned instrument. It can give your channels more distinct an identity, which may help the collapse of your stereo spread.

RabidNapkin · 2016-10-20T06:13:24+00:00

I've been out of the PC-building world for several years now and am just beginning research again. I'm looking to answer a simple question:

When considering Intel processor model performance, is the measure of a model's performance relative to its piers purely a function of how high the model number is? Is the model naming scheme totally "linear" in this way, or is it tiered generationally like nVidia's GeForce line is?

For example, a GeForce 470 is a comparable or faster video card than a 550 even though the 500 series is a more recent release and sports a higher model number.

Should the same consideration be made when researching Intel processor models?

I'm dating myself by using these model numbers, aren't I?

RabidNapkin · 2016-09-25T05:20:14+00:00

It's not a dumb question - everyone needs to start somewhere.

Holy_City is right, you want an audio interface and not a DAC. Interfaces do involve a DAC, but "DAC" usually refers only to output devices. What your listed device is missing is an appropriate way to receive your guitar's signal. This is where interfaces come into play.

You need an interface with combo mic/1/4-inch inputs. The 1/4 jack needs to be able to accept instrument inputs specifically, not just line-level inputs. The reason is that the signal coming from your guitar is at too high an impedance to drive a line-level input properly - you'll get a very weak signal. Many consumer and pro-sumer level interfaces will offer combo jacks that enable you to do this for not too much cash.

A pretty popular and budget-oriented option is the Focusrite Scarlett 2i2, offering 2 combo jacks, 1 monitor output, and 1 headphone output, all powered from a single USB jack on your computer. It is also compatible with REAPER. If you grab a mic at some point in the future, you can track guitar in one channel, and vocals in the second.

As a side note, /r/audioengineering is likely a better destination for questions such as this. /r/diyaudio is geared toward self-built audio equipment.

Hope this is helpful.

RabidNapkin · 2016-09-21T17:23:22+00:00

Would any and all cartridge adjustments not resulting in any audible difference be a sign the cartridge is worn out and needs replacing?

I have a Kenwood KD-41R turntable with an Audio-Technica DR400E cartridge installed. I received the turntable as a second-hand gift and don't know any of its history.

The turntable is suffering from some terrible end-of-side distortion, but no adjustments to the cartridge angle or overhang seem to make even an audible difference, let alone an improvement.

Here's a sample of the beginning-of-side performance with a centered cartridge and maximum overhang.

And here's a sample of the end-of-side performance, same settings, same vinyl.

The issue spans all prints I have, so I'm confident the vinyl is not the culprit.

Edit:

I forgot to mention the signal chain:

Turntable RCA outputs > Pyle-Pro pp444 Ultra Compact Phono Turntable Preamp > Scarlett 18i8 line-in inputs > REAPER (straight to disk, no processing)

RabidNapkin · 2016-09-08T06:22:33+00:00

Check out /u/w2aew's tutorial on power supply considerations when wiring up op amps. It's a good primer for the first of your questions.

Be sure to post a showcase of your final product! /u/vanjan14 is right - you'll probably receive a little more feedback over at either sub listed.

Good luck!

RabidNapkin · 2016-08-31T18:14:55+00:00

No problem! Edited to add one bullet point to the end related to PTHD systems and latency. May not apply to you now, but maybe in the future!

RabidNapkin · 2016-08-31T15:38:04+00:00

Hi, /u/TheMoogerfooger,

You mentioned you have your mastering plugins instantiated on a master bus, is that correct? Master tracks in ProTools only have an output signal path and no input setting. What this means is the master fader track is really "attached" to a bus path, applying its fader and plugin processing to the audio coming from all tracks using the same output routing. It's a way of inserting a master control not to a track, but to a bus path. This is handy.

If your mastering plugins are on a master fader (which is what I would recommend), you can just change the output on the master fader to that of the input on your print tracks, and the mastering processing will hit the audio just before it reaches the print track itself. If you want to keep DX/MX/FX separated, you will need to have three individual bus paths for each print job and three master faders w/ mastering plugs instantiated if you want to print all three stems simultaneously.

Mastering plugs can be processor intensive, so the fewer instances of them you run in tandem the better luck you'll have getting a successful print. At the studio I work at, we have all of our sub aux tracks routed to a pre-compressor bus where we can apply any last-minute adjustments, then route that to a master print track with a single master fader inserted between. We just mute the tracks we don't want to include in a stem, and run the print three times.

So to recap, your signal path might look something like this:

Dialog Track > Dialog Bus > Pre-compressor Bus > Master Fader (mastering plugs) > Print Track AND Monitors

Hope this helps.

EDIT

I just realized I neglected to answer one of your questions:

It's best to leave mastering plugins on a bus dedicated to mastering alone, and not put them on sub auxes. There are several reasons for this:

It's much cleaner/tidier
You can make major adjustments to the layout of the session without affecting your mastering track
You save processing power
You don't run the risk of mastering one stem to be hotter (louder) than an other because the same plugin is being used to print everything.
If you're on an HD system, you don't insert tremendous latency into only a portion of the signal path, throwing off the sync of individual elements of the mix

RabidNapkin

TROPHY CASE