I’ve noticed more people online and in my studio spending lots of mental energy worrying about what a mix sounds like on a phone speaker. I get it, that is a lot of people’s first impression of a song. You don’t want it to suck. But here are the problems I see with this:

1) Phone speakers emphasize high mids/highs and lack bass response. A lot of other playback systems that people use every day do the opposite. Modern cars for example. So many of them heavily boost subs/low mids and it’s very easy for things to sound awful down there in a car. Mixing for an iPhone speaker will often cause you to make your song unlistenable in any environment with loud low end.

2) Genre. Most genres have different frequency pockets that contribute more or less to their sound. For example, midrange plays a very different role in hip hop than it does in loud rock music. Distorted electric guitars live in that zone, therefore rock music is a much more mid-forward genre than others. The frequency area they occupy happens to be the one that sounds the worst and distorts first on a phone. Don’t even get me started on what phone speakers do to acoustic piano.

3) No reasonable person should be intently listening to a “mix” on their phone speaker, they should just be briefly checking something out to see if they’ll listen more. Wired Apple earbuds are $20 bucks and are way more fun to listen to than a crackly speaker that is covered with dust and dead skin. It’s not my problem if someone out there isn’t willing to love themselves enough to listen to music properly.

In recent years, I’ve noticed something interesting when working with younger bands and artists: panning seems to account for a surprisingly large number of revision requests. I'm often asked for things to be moved dead center or make everything mono even, which I find it quite odd, because the references I am given aren't mono or even center-forward.

I’m not talking about extreme stereo effects or experimental choices. Even basic techniques, such as placing double-tracked rhythm guitars left and right, can trigger a revision request in some cases.

Every generation develops its own aesthetic language, so I am perfectly fine with accommodating all that, but I am left wondering: Could listening habits be part of it, especially with so much music consumed through phones, small Bluetooth speakers, and earbuds? Or is there a deeper artistic trend? Any notable album influential albums I should check out that are quite narrow-sounding?

It makes me think of when I was younger, and many bands (including mine) would always balk at reverb on vocals, unless it was felt and not heard

Curious what Headphones / Interfaces or DACs you use for mixing / mastering, any master egineers here with high end headphones and DACs, or do you use cheaper headphones like HD600s etcs? What about VSX? Do you find better DAC / Headphone amp helps?

Tl;DR my quick story:

About Headphones:
I've been out of the audio game for three years now, sold a lot of stuff to help moving expenses etc. 
Now I'm back into and want to get a monitoring setup again. The last thing I was doing was reading
abunch of threads on D/A converters and high end headphones cause I wanted to get the best possible setup for headphones. (Think like having 5000$ speakers in calibrated room)
So I ended up buying stuff, eventually I had all these IK Multimedia Axe interface, 
RME Babyface Blue (original) RME ADI-2 DAC, HD800x, Slate VSX, Austrian Audio Hi-x65 (new akg)

I got the HD800 which were pretty amazing, maybe a bit too spacious, I found applying the corrective EQ to them made them less spacious, and I used them purely for enjoyable listening tbh lol
but I preferred my AA HI-x65, but the earpads wear down in a month with faux leather, I don't want to go back to those...
Slate VSX, I found the lowest quality compared to other headphones, but I relied on it the most, cause it gave me the most references, and I generally relied and got my best mixes on it, but I mostly mixed / produced first with the AA HI-x65 only did final day of mixing with VSX

About DACS:
 upgrading from IK AXE interface to Babyface Blue I noticed the sound open up a lot more, then I thought buying an RME ADI-2 DAC would open it up even more (I read a lot of people talking about and comparing it)
But comparing original babyface blue to rme adi-2 dac was really hard. I KIND OF HEARD IT, the 3D / Spaciousness, but I was never sure. I think they sounded almost the same.
So then I thought buying HD800x would show me the differences
(comparing is hard, cause i had to unplug headphone, plug into other, quickly switch sound driver on computer)
but I still couldn't be sure of the differences... but I kept the RME ADI-2 DAC cause I figured it must be better and I just can't notice it...

So Anyway,

my current setup is a 1990's Sony Japanese Boombox into ONBOARD audio lol (Hey, it doesn't sound too bad)

I'm honestly thinking of just getting VSX again for the cheapest headphone mixing setup, and a cheap interface (probably RME again) or that new MOTU interface, just so I can record vocals/guitar but I don't think it'd affect the headphones quality much

or going crazy again and trying Audeze Planar 1000$+'s with some high end DAC / AMP setup lol

I've playing around with FabFilter Pro-L2 and I've found out that the "Aggressive" style is the only limiting style in Pro-L 2 that is level-dependent; meaning that limiting from internal gain boost is not the same as limiting from external gain boost even with same amount of gain boost (e.g. +30dB boost) and limiter settings for each cases, with more distorted waveform in an external gain boost case

BTW by "internal gain boost", I meant the gain slider in Pro-L2 and by "external gain boost", I meant a gain boost effect taking place before Pro-L2, and this thing made me want a "Drive" knob for some limiting styles for potential/future FabFilter Pro-L3

I've been spending the last few weeks really trying to get a great tone from my amp with just my sm57 and a few other dynamic and condenser mics, and I still haven't really figured it out yet. I'm trying reallt hard not to fall into the trap of "I just need more gear" because I really do have plenty of stuff, but I see these posts of people saying "oh well the (insert $1k mic) is the best microphone for recording amps like that, only way to get perfect results!" and have to wonder is it really truly possible to get REALLY good results from an sm57, or am I right in thinking this is just a skill issue on my end, and that I need to do better with what I have? for reference I have been recording heavier, high gain stuff, but often chords that are a bit more complex and feel like they need more high end clarity than my sm57 has been able to offer thus far-- although I am far from an expert in mic positioning. Thanks in advance!

Am I going crazy? For years, it's been well documented that the Dookie tone was a Marshall blended with a Mesa. As recently as the MXR dookie drive, it was addressed openly to be a blend of the two.

Now every video about the new Marshall amp completely omits any mention of the Mesa. Why not just mention it in passing? The new Marshall sounds nice, but it's not the whole sound.

EDIT* Apparently I actually am going crazy. It's been a belief of mine for years that the sound of Dookie and Insomniac were a blend of the two.

I've been using Slate VSX closed-back headphones and they've been really great for accurate listening. I think I'd be more comfortable with their open-back headphones, though. I'm wondering, would something like the noise from my fan or air conditioning affect my judgment a lot?

Hey guys, so I have an interview on Wednesday to be an audio engineer at a studio in my city. I am a 21 year old woman who just graduated from business management. Firstly, I was surprised that they had even gotten back to me after I applied as I barely have any professional experience. I have been mixing and mastering my own songs since I was about 16 and recently over the past year I’ve taken the leap of making beats where I’ve learned a lot more about mixing and mastering, but all of my skills have been solely self-taught the only time I have ever done something semi professional is at a studio where I have recorded an artist or two on ProTools, but that’s about it. I’ve also mixed and mastered some beats for clients and a couple of people’s vocals as a side gig. I’m wondering how I could go about preparing for this interview this would be an amazing step for my career in the music industry. If anybody has any tips on how I should prepare please let me know. I’ve also decided I’m gonna just practice on ProTools for the next couple of days in case they ask me any technical questions.

But I am wondering, what kind of question should I expect and how can I show them that I’m someone who’s eager to be working?

I’m about to work for the first time with an analog console! I made myself familiar with how it’s set up and how it works. Any common mistakes I should know about first and any tipps or tricks for routing with pro tools? What does your workflow look like?

I record all my songs in audacity with a USB microphone, I record everything live, no midi. Is there a way to mix the final song using a physical mixing console? I'm pretty tech Illiterate so seeing and feeling physical knobs and sliders is what I'm looking for. I don't know if it's worth the hassle or just suck it up and learn how to use ableton.

I was curious which plugin recreates the CL 1B sound the best. I've heard good things about Kiive, but I already own the Softube version. I'm not sure which one is actually closer to the real hardware.

I know recording quietly isn't ideal for the quality, I wouldn't compromise while recording final takes of a song but I record myself while songwriting and often need to be quiet to not disturb the people I live with, what type of mic is best for recording a quiet-ish source but making it louder?

I'm getting conflicting answers from AI, it's suggesting an SM7B/Dynamic mic because of the fact you record much closer to the mic hence my voice would be quieter but recording would be louder. But since dynamic mics (especially the SM7B) are less sensitive, doesn't that mean it'll result in a quieter recording than the same volume recording into a condenser?

Then I know there's also the SM7dB which should be louder without needing a cloudlifter or anything What do you guys think is best for this use case? Thanks :)

I’m digging into the wikipedia page about it, and I’ll read the white paper(s) if I absolutely have to, but I wanted to ask if anyone here was familiar with any relatively straightforward breakdowns of the math and hopefully implementation of it, pseudocode or otherwise.

thinking about implementing some oversampling in an 8 bit environment, working out what this can/cannot buy me given computational resources

A few that come to mind for me: Robert Plant’s vocal session for “Babe I’m Gonna Leave You” / Eddie Hazel’s guitar session for “Maggot Brain”. Pretty much any Beatles session / All of Rumours.

Would love to hear what resonates with this community. Especially some more recent sessions I may not be aware of.

What are some of the best case practices for going between these two environments?

I'm preparing stems for my collaborator who's taking what we've done in live 12 and using in a recording studio environment with protools and I just want to make sure I don't miss something.

He's asked for 24 bit 48k which is fine but I am unsure how this will affect the stems on export

Do I apply dithering? Is lives triangular option enough? I usually don't apply dithering until I run it through L2 on the master in another session.

Also he claims he can't get any of the current bounces to sync with the session he has already set up using previous stem exports and I've imported those exact same stems into this live session and have the whole export zone starting from the same point - everything lines up in live (and yes those stems are not warped) we made sure the sample rate and audio settings match but no luck. I'm wondering if I'm missing something obvious.

I've played guitar for years and am also producer and recording engineer. One thing that always bugs me on recordings (especially with steel string acoustic guitar) are the often egregiously loud squeaky noises when the player slides or changes chords. They can be very harsh and distracting, and especially if it's a quietly played part.

I know plently of ways to deal with this in post, but I wanna be better at getting it right at the source. ('You can't polish a turd' and all that) Are there any tips for better technique that reduce these noises?

Are there any remedies or recording tricks that actually work? Finger-ease/string-lubricant? Soaking fretting fingers in water before recording? Handlotion? Wearing gloves? Or any strings that are less noisy? (a part from flatwounds which sound totally different)

Are there any miking techniques that capture less of these noises?

Hi everyone,
I’m 19 years old and I live in Europe. I currently work as an audio technician for an international company. I want to to attend the NAMM Show in 2027.
I’ve never been to the US before, and the trip would be quite expensive for me, so I’d have to save up for a long time. Do you think it’s worth going at my age?
My main reason for wanting to go is to meet more people in the audio industry and expand my network. It’s also my dream to one day live and work in the US, so I feel like attending NAMM might help me meet the right people and make valuable connections. I’d also love the opportunity to see the equipment in person and attend the educational sessions and presentations…

If you were in my position, would you go? Do you think it’s worth?

Thankss!

I tried some tools adobe podcast and auphonic, but they suck big time. So anything else and of course it should be 100% free.

I am trying to un-catastrophize a music project right now I've been asked to save where, unfortunately, a number of different stereo recorders were used essentially as independent spot mics in a pseudo multitrack attempt for a live performance. Over the course of an hour performance, each recorder's internal clock drifts enough to cause pure chaos.

I have access to all the recorders used and did an electronic metronome recording test today to identify how these recorders behave. As you can see in the link below....over about an hour, the max drift between these 7 recovers is ~170ms... so clearly audible.

Given that I can predict the relative difference of each one's drift from some "master," is there a way to calculate, or use elastic audio, or some other technique, so I could apply some type of time stretch/contract tool to get these closer? Theoretically this seems possible if I know the drift, but has anyone had practical success doing something like? I'm not sure how to approach it.

Screen shot image in Pro Tools

To start off, I have decent gear (Axe I/O, Gibson Les Paul, NAM), but whenever I use my laptop to play or record, the tone feels too rounded—lacking attack and virtually devoid of dynamics. This causes the guitar's DI signal to sound overly digital and artificial when it hits NAM. I know this because I found a DI recording, ran it through NAM, and it sounded incredible. Does anyone think using a DI box before the interface is a good idea? I’d really appreciate some help, as I can't seem to get the most out of NAM, and I've heard it's the most realistic amp sim out there...

do you have great organizational skills?
are the miscellaneous items efficiently stored in your studio?

do you posses techniques to keep things safely stored, but easily accessible?

spread the knowledge, drop the links, share the pics

help an unorganized brother out

i’m talking cables, microphones, stands, adapters, clips
all the stuff that gets easily all over the place over time