been meaning to get into music production, and i'm just getting everything into order before i actually START doing anything (also gathering funds for a better pc jejjj) and i'm just wondering if there's any programs that can accurately emulate the compression/effect that certain digitized recordings from the 70's & 60's have. [example to what i'm yapping about]

(should've clarified that i'm not specifically looking for compression, but rather just things that produce the silly little quirks that old recordings in general have. i look like an even bigger newb then i actually am lul)

This legendary mixing plugin never seems to go away. It’s been replicated in various DAWs, people still aren’t sure what it actually does.

Anyone still use it? Mix bus? Mastering? Individual tracks?

Hello everybody,

I've decided to up my gear game and get my first studio monitor pair, but before I do anything stupid and blow away cash for something I can't use to its full potential, I've decided to consult people with more experience.

On the following link I've uploaded 2 crude drawings of my room with measurements:

https://imgur.com/a/y1EsPX9

With that said, my question is: what is the ideal place to place my computer and the monitors I'm planning to buy?

Since I have a sloped ceiling only on one side, a rule of thumb I've picked up while reading other posts is that I should place the monitors to face away from the shorter wall and have the room "open up" behind me. Another thing that I've concluded is that I should definitely use the "longer side" of my room so the sound has more room to travel to the end of the room before it bounces back.

That being said, the logical choice would be to place the setup on the shorter wall side. One thing that I'm having a dilemma with is the portion of the room that has a green line (0.55m) on the second image. That part of the room is basically comprised of built-in open shelves such as ones on this photo:

https://www.pinterest.com/pin/11-builtin-bookshelves-cococozy--495184921521905760/

Will this create any unwanted sound bouncing if I place a monitor right next to those shelves? If that's the case, what can I do to eliminate it?

If I'm completely wrong with any (or all) conclusions I've made, please correct me and give me your input. Any other input on acoustic treatment of the room and equipment placement is absolutely welcome!

Last thing I want to ask is recommendations for the actual hardware. I have a budget of around $450-$500, which allows me one of these choices (that I'm aware of): Adam D3V, Adam T5V, Adam T7V, Kali LP-UNF and maaaybe Yamaha HS5.
I've been told by some music store employees that my room is way too small for even 5" monitors and that I should look for 4" or even 3" monitors and add a sub later on if I decide I need more low end, but I've discovered the Kali LP-UNF some days ago and everyone's raving about how they're the best thing ever for small room setups, so some input about that would be greatly appreciated.

Much thanks to everyone who takes the time to look into my problem and offer some advice!

Hi everyone,

I have been recording a podcast for friends for some time and decided to go "multitrack" to allow a more flexible post production treatment. The purpose is to be able to apply treatments indepedently for each speaker and make sure all voices are at the exact same volume.

While editing the podcast for the first time in this "multi track mode" I noticed normalization wasn't exactly as easy as I expected, at least in therory. The end result was fine though, luckily.

Here is the issue I may encounter later with normalization:

I noticed some speakers speak a lot less than others. So when applying a -16fb LUFS normalization to each track, I'm almost sure that people who have a lot more silent parts in their track will have their voice louder than people who speak a lot with very few silent parts. Since, correct me if I'm wrong, LUFS normlization perform an average target volume.

So, what would be your recommendation to normliaze each track to make sure all voices are perfectly at the same level? I am using Audacity for editing for information.

Thanks

Vocal are heavily distorted/verby (early black keys) pushed through a guitar amp and neve 1073. Guitars high gain marshall (Early Oasis). Obviously I know the vocals needs to win this battle so I EQ the shit out of the guitars but I still feel like the vocal does not pop out as much as I would like. My opinion is the guitars are way too distorted but they insist on recording the amp live and takes are already done. If I had more control over guitar tone I could shape it but these are driven to the point of a naturally compressed block of a sound wave

Background: my wife is enamoured by those YouTube “smooth jazz” videos where they just continually loop a rather simple and unremarkable but auditorily pleasing jazz soundtrack along with a bucolic animated scene without any people in it - think a warm coffee house, or a cozy library, with snow falling outside the windows, that kind of thing.

I have managed to grab the audio track for a few of these, most of them in the 2-3 hour range, so the music does loop quite a number of times within that track. Cutting out a single loop is NOT what I want to do - I just want to start and end the track at the first and last loop transition that I managed to save.

The objective of this is to drop it onto an older classic iPod (30-pin) for use around the house where we have docking players set up. That way, she can take the iPod with her, set it to play a random collection of tracks within that group, and she would get a good 12hrs of smooth jazz music with decent transitions between the tracks - it wouldn’t abruptly cut off in the middle of one jazz loop to abruptly start in the middle of another one when the tracks changed.

So I am curious if there is any software that can analyze a track of music and identify where, within that track, one loop ends and another loop starts.

Each loop seems to be in the 5-15 minute range, so I would be hard-pressed to catch this transition manually. Which is why I am looking for software that might be able to do the same thing.

I work in IT, so I can converse on a decently high technical level, I just have zero experience in the audio engineering field, so I have absolutely no clue what software is out there. Similarly, my apologies if I have mangled any technical terms.

I’ve been looking at my first control surface now that I’m actually starting to take music production and engineering as a career, but because I’m a college student, I’m crazy broke. On Reverb.com, I’ve found a bunch of awesome midi control surfaces, but they’re firewire. Would I be fine using a firewire to usb cable and using it as a control surface? Any help is appreciated!

Hey there, I have an audio engineering question. I have a live drum recording, however the drummer was just slightly off time in various spots. Ive tried manually editing the recording to fix it, but its proving to be very difficult. I was wondering if anyone happens to know of any AI tools or online resources where I could upload the recording, and the audio could be quantized to a specific BPM thereby fixing the timing issues and making the drum take usable in my project? Thanks.

I know resonance applies to “low” frequencies and presence applies to “high” but what specific frequency numbers do they encompass?

Hello everyone,

I have an old family history video that a deceased family member made back in 1988 on VHS, and is currently in MPEG-4 format. Most of the video is the family member interviewing other relatives, with the audio of those interviews playing over old photographs. Unfortunately, the whole thing is riddled with awful static sound from a crude transfer process at some point, and it makes it very hard to listen to. At some points, the voices are very hard to make out because of this.

Is this the kind of thing I can clean up using either AI tools or more manual audio editing tools that a novice could figure out? If so, any good recs?

Or would the experts here recommend I take it to a professional audio engineer to see what they can do? If that route, any recs for best services?

Trying to preserve this old family history treasure now 30+ years after the original videographer passed away.

Thanks very much for your help!

Thought I’d post in here for anyone who knows of some good resources for ear training, I can differentiate basic frequencies but I’m looking to practice getting better at ear training geared more towards general mixing. I obviously plan on just practicing mixing stuff regularly and get better that way but I’m looking for some additional help 😅

I’m worried about potential lack of space

Im going for a very lush and full sound

Hey everyone,

I come from a background in electronic music production. I usually produce my tracks at 44.1khz, and when I’m working on film scoring or orchestral/cinematic stuff for Video, I set my sample rate at 48khz..

But for sound design/SFX sample libraries, I noticed a ton of them are at 96kHz. I’m curious , why is that the standard for so many sound design libraries? I’m just genuinely curious about the reason behind this practice, so if someone has knowledge about this and could share, I would be very grateful as it would help me a ton.

I know that even with professional equipment, some post-production is always necessary to achieve high-quality voiceover audio. What tips, plugins, or software can help make the post-production process super straightforward, fast, and easy?

Hello!

This might not be the usual type of post here, but I live in an apartment with a bit of a problem. I can randomly hear this really deep "booming" noise (even at night), which manages to vibrate my floor and my desk a noticeable amount. It's unclear where exactly it's coming from, because it feels like it's simultaneously coming from every direction except the floor.

It's very easily audible to the human ear, but I've tried to capture it with both my phone and a Zoom H2n microphone, and I'm not having much luck; you basically can't hear it in the recording, even without other background noise. I tried EQing it to boost low frequencies, but that doesn't do much either; it seems it's just not being captured adequately in the source recording.

Does anyone have any hardware recommendations that can either specifically target low-frequency recordings, or otherwise provide a great sound profile that includes any "background noises", and mostly just captures what a human would hear?

Thank you very much!

Question in the context of learning and experimenting. I thought modern DAWs, internally working at 32 or 64 bits would let you crank the gain way pass 0 DBFs without any clipping/distortion.

I thought i had done it already in the past but rn I'm opening a simple piano sound in Cubase, cranking the channel fader (not touching anything in the VST so the plugin is properly gain staged), and cranking the master channel and it gets horribly distorted.

Not sure if I'm doing something wrong or if i was mistaken from the beginning with my assumption. I even changed Cubase internal processing precision to 64 bits but still get the same result.

I don't have physical WAH pedal but I know that it manipulates EQ somehow. Can I simulate WAH effect by creating bell/shelf EQ? Which frequencies should I cut/boost?

I'm trying to return a Jaspers 170-4-120B I purchased online from Thomann. I live in San Francisco, CA.

The issue I'm running into is that every delivery service I've looked into so far (UPS, DHL etc.) is quoting me between $400-$900 to return it. The dimensions / weight are:

10.5 in x 10 in x 66 in

29.54 lbs

Has anyone here had success returning something this large / heavy to Thomann from the US before?

Any advice or suggestions would be appreciated, thanks!

I'm a beginner when it comes to Audio and yesterday I had my first studio session recording drums for an assignment. only just realised now that we forgot to check for phase issues using the desks phase flip button. I know I can do it in the mixing process but I was wondering if it was something that needed to be done in the recording process. Sorry if this is a stupid question I'm new to audio engineering and I'm trying my best to learn and not make mistakes like this in the future.

Hello! I’m thinking about selling audio that I recorded and edited to animators or even a audio only movie and I was just wondering where can I? Thank you!!

For those that have had ear pinning surgery aka pinnaplasty or otoplasty, has it affected your hearing at all?

How quickly did you return to mixing?
Were headphones a struggle?
etc.?

Any insight is much appreciated, thanks a lot!

Hi! I’ve seen pretty often that some musicians are able to change their voice from male to female in their music when they don’t sound like that irl. And it sounds really good and realistic. Is it possible to do that but the opposite? I can’t find any info on it. Are there any plugins or devices to change a female sounding voice to a male sounding one realistically??

Hey everyone,

I’m going into the studio in a few weeks and I’m researching what the band have asked for in terms of references and they’re pretty amazing:

Recollections - Miles Davis Mortgage On My Soul - Keith Jarrett Sivad (live at the cellar door) - Miles Davis Black Sabbath - Black Sabbath Ince Ince Bir Kar Yagar - Selda Bagcan

So all are 70s jazz/rock/groove vibes. Close Miked drums, quite dry but with a lot of tone. I’ve tried looking for session photos from the albums but no luck with any of the yet so does anyone have any info on how the drums were miked/which mics/which drums?

Would love to know your thoughts

Speaking about live mainly, as most other daws have VCAs. I used to always pull down track faders. Saturation got too hot, pull down the fader. Now I've gotten comfortable pulling down output nobs, either every plug in on the track or simply the last one in the chain. This inevitably ends up leaving me with a lot of tracks with their faders around 0. So I guess what I'm asking is we all know level matching is good, but to what extent? Does anyone else do what I speak of here? I find especially in ableton where VCA tracks are a complete foreign concept and absolutely non existent, its easier to do this especially if you have a tun of group processing, rather than turn down faders and fuck your processing levels up. Is this a bad habit to get into? It seems to work for me, but I like understanding the science of things too, rather than simply "If it sounds good its good", which is why I'm posting here lol. And yes I know the utility plug in has a handy gain nob, but I feel like thats an extra step thats not always needed, since so many plugs have dedicated output nobs.

I am trying so hard to get into recording. I recently decided I was gonna record my band ( live drums etc ) because to be completely honest the economy hasn't been treating me well so I have very little disposable income to give someone to mix. I have been extremely hard on myself and feel like I am spending so much time nit picking everything. I am scared I am gonna do a terrible job and spend all this time releasing something I will not be happy with ( because of my lack of knowledge ).

https://drive.google.com/file/d/1f6Dh6lNL8vfLvXhRyXxfFfx1OVdSOE-m/view?usp=sharing

I am looking for any help I could possibly get. I am also willing to compensate in graphic work or a small amount of money for any help. I also wouldn't mind learning techniques etc. I really appreciate everyone who reads this and lends any feedback. I just really want to do my band members justice and I did not realize the undertaking I would be taking on doing all of this. Much love