Thanks so much! And if there is anything we can do to help, just let us know: info@elysia com

Cool! Thank you very much for the kind comment.

Thank you for the kind comment. You're welcome!

Hey, thank you very the positive comment. We're glad to hear you like your xpressor!

Hello Thomas, he's using several pieces of equipment for different tasks. For more details, we'd advice to get in touch with Simon directly at: www.simon-phillips.com/

It's a BURL Audio B2 Bomber.

The alpha compressor includes a great limiter stage as well, which was just perfect for this session. It does not do the brickwall thing and behaves more like a saturated tape machine - not perfect for everything, but very nice for this track.

Hello! Yes, it actually is the stereo master track. Regarding your question, your idea should certainly work from the technical perspective, but only you will be able to judge if the result is what you had in mind in terms of sound (quality). Let's put it like this: It's certainly worth giving a try, and if you don't like the result, keep experimenting with the analog chain.

Thanks, Would it be fine to take the Master L & Master R (2 tracks) from the Mix Page out of the DAW through the mixer & record back into 2 channels on the interface into a stereo track in the DAW to create the stereo master -or-is it taking the already summed stereo master track (1 track) out of the DAW through the mixer & back into the interface throug 1 track as well (in the example Simon's giving)? I imagine both can work as well as taking several (maybe 4 stems ) out of the DAW into the mixer, out the mixer Stereo L and R, & then back into the interface through 2 channels into a Single Stereo Master Track? Thanks & let me know, please. I may have taken the mixes as far as I can (& possibly seeking some local mastering people to take it further, if possible) but I figure I'd try this mixer as 1 final mixing/summing/mastering possibility myself. Kind of like you said, the proof is in the pudding...@ a point, it's seeming anything I try (taking final mix through a tape machine or compressor) just seems to make it worse (going through extra A/D, D/A conversions etc. comes to mind) & I'm pretty happy w/where I've gotten the mixes myself on my system (through Studio One w/KRK Rokit 5 monitors), so that seems a lot of it, & I just want it to translate properly to as many if not all listening forums as possible, of course...So perhaps the mixes are done & a mastering person can help or perhaps they're done completely for this project & like you said, it would be worth a try to try running through the analog mixer & back. I'm guessing I can just run out my interface into 1, 2 or 4 Line Ins on the mixer & either out the mixer's 2 stereo L & R Master outputs, if not out the mixer's Master Singular Mono output, but I imagine it's better to utilize the Stereo Outs to retain stereo information even if the initial track is coming into the mixer through one track, but it seems it would stand to figuer coming into the mixer w/the Stereo L as well as R coming out of the interface to process the truest stereo mix? Thanks & sorry this message is so long, all this is a little daunting @ 1st & I'm trying to make heads & tails of it, so any & all help would be greatly appreciated. Thanks!

To be honest, you should check out the possible options and judge them with your own ears - as we're talking about a few tracks only, it would not take long to set this up. In the end, mastering can be a complex process and it usually takes lots of experience to get where you want... so one of the right things to do would certainly be to experiment as much as you can. On the other, it's absolutely worth investing time in finding a professional mastering engineer who "sounds" right for you and maybe have this guy master a demo track for you, if it seems affordable. This way, you always have a reference you can compare with your own approach... Sometimes not doing everything by oneself is just the right thing to do - sometimes the opposite is true. Yes, this is not very concrete, but these things are usually so individual that there is no one solution fits all at all.

OK, I'll check it all out & see what's best! Thanks so much for being in touch, Athuai

You're welcome, and enjoy the process - it can be tricky, but it's worth while for sure.

It means 16 bit/44.1 kHz (mp3/aac uses exactly that). You could master it to 24 bit/96 kHz (popular for flac).

​@@jonasnitz7678 But mp3/aac aren't exactly lossless RedBook 16/44.1. :-) Apple does have some software for creating AAC and they prefer labels use 24/96 masters and the end result is much, much closer to RedBook 16/44.1. They use to call it Mastered for iTunes, I believe they are now calling it Apple Digital Masters. So they are originally 24/96 masters that have been run through their MFIT mastering software. I know it reduces to 16 Bit, but the software still utilizes the extra bits from the 24/96 masters. I read an article/interview of Bob Ludwig and he was praising it as a solid method for creating Lossy, even though Lossy isn't Lossless. Then there is the, dare I say it, MQA, takes any format, creates a lossy version, packs it in FLAC or ALAC and the regurgitates it back to original format IF you have proper software and MQA enabled DACs for playback. I know there's a TON of controversy around it, but it's another format for high res streaming. I have not done any listening comparisons, but I was watching a video on someone that's pretty honest about his listening and he actually prefers MQA for most of the recordings he's heard. He only said a few weren't up to snuff, but overall, for newer high res recordings, he actually liked that format. The other is DSD. There are some boutique record labels that actually record in DSD or 24/384, and then convert to PCM. The reason is they claim they sound closer to tape only without any distortion, tape hiss, etc. They are using DSD because they claim it sounds more natural and has higher dynamic range because they aren't using any type of compression or limiting, EQ, etc.. and they are just recording live performances of acoustic instruments and vocals without ANY special effects, etc. Just pure acoustic instruments/voice, so they don't need PCM to alter anything. DSD is the archival format that Sony created for the analog recordings done on tape, and then they sell as SACDs or DSD digital format came much later after they realized there was a market for that. Don't know how well it streams since the file sizes are HUGE. There are a couple of streaming services that are streaming 24 bit Lossless, but not as much content and they aren't getting that many subscribers yet.

@@Oneness100 Well I never said that AAC or mp3 was lossless or even close. I just stated that AAC and mp3 uses 16 bit/44.1 kHz. It's like ripping a CD and make mp3's out of it.

Kenny Earl because his shelving at 11KHz was too much...it brought too much hi freq. so the snare sounded thinner.

Thats subjective...

@@3star2nr Perhaps, but he's not wrong, it is harsh. And does not come across as a clean master. He almost ruined the track with the decisions he made.

@@Gamervidsman2000 ok... So dont use him on your track...

@@3star2nr It was a quick demo. I doubt he's bad at engineering if he actually took his time and made the right decisions. My issue was the demonstration. It's a "Look at what happens when I do this and this", without giving us the proper procedure, while getting a good result. A good result is important. You'd want the people watching to know and trust that you're giving them the right information by showcasing a proper mix that translates as something better than before (subjective yes, but no matter how unique a sound might be, I think it can always sound "good").

Chris Choir Yes just been to his website

There are no pro studios that record in FLAC. Take that crap somewhere else. It's either PCM or DSD or DXD which is PCM.

@@380stroker DAW's record in WAV. FLAC decompresses to an identical copy of the original audio data (no loss). Then it's up to the player to deliver a good reprecentation of it. Same goes for CD-players. Some can sound harsh (cheap stuff) and some can make the music sound even better than an original coming directly from the DAW due to oversampling.

@@jonasnitz7678 Let's do an SRC shootout. You up for it?

No thanks. Buy good rifles and scopes with that money.

His mastering equipment is analog, so he's mastering at 20-20,000 Hz (or whatever his D/A converter is capable of) and then that's sampled at 44.1/16.

Dude: he's recording at 96KHz, which also doesn't make sense if the final product will be at 44.1 KHz. The analog bandwidth for mastering equipment is usually beyond 35 KHz FYI

felipousismix 96KHz makes _exactly_ as much sense as 88.2 (or 48). I don't think you've grasped the audio path here. He has 96kHz/24bit digital audio which he feeds into his D/A converter, turning it into 20Hz - 20kHz (or whatever specs the DAC has - probably much better) analog audio. He uses the analog compressor and EQ to master it and then feeds that analog sound into his A/D converter, which turns it into the 44.1kHz, 16bit digital audio that gets recorded. As long as his D/A and A/D converters are good the conversion won't degrade the sound. Had he stayed in the digital domain all the way through, 96kHz would have made exactly as much sense as 88.2 as well - modern samplerate conversions are asynchronous, so you don't lose fidelity in the conversion process. In other words you lose more fidelity by "only" mixing at 88.2kHz than you do by converting 96kHz into 44.1 in the final stage.

Being doing mixing and mastering for years, I insist it's better to use 88.2 instead of 96. 96 and 48 were designed for video sync. Cheers.

felipousismix If you prefer 88.2 you should definitely stay with it. But 96 won't sound any worse... And if all you're doing is mixing the average rock band 48kHz wouldn't sound any worse either.

I'm pretty sure that "dithering" is not anything close to as he described it. :S

I'm pretty pretty pretty pretty sure you're pretty pretty pretty pretty right.

It's always about minimizing your losses and preserving low level detail of the source as much as possible. Dither is simply random, low level noise which prevents the otherwise build up of truncation distortion when going from higher to lower bit depth. And this noise accumulates at a far lower level than would cumulative truncation distortion (as it does with any digital processing, even gain, fades, etc). It's also the least most important aspect of mastering. Just do it last. Although, generally capturing at 24 bit (not 16 bit) for this very reason, ie: to allow for any further digital processing and/or lossy encoding.

I like it.

That was Blunt, Frank

That was Frank AND Blunt

That's Frank Blunt for ya