i'm not sure which mic is suitable for my voice. i don't have a store nearby to go test mics. i sing rock usually. can get sensational, can get high pitched. also i live in a place that isn't soundproof. do you think i should but a large diaphragm condenser like Samson C01 or an AKG? can i use them on acc guitars? i was thinking of getting an sm57, cuz they're pretty good and unbiased on vocals (i've heard) and i've seen you using them on acc guitars. this will be my first and only mic for long. also, does the sm57 need a pop filter?
what do you think about Steinberg audio interfaces? the ones which are in the same price range as Scarlett, PreSonus and Native Interface cheapest models?
Yes, actually it's about the Kali monitors: i've looked at several videos on you tube where it is said there is a loud hiss from these speakers: do you have the same experience with yours? I like the sound quality from and am considering getting a pair.
I want to know if you have had any experience with RME interfaces or other their other products? You seem to have dealt with pretty much everything out there, so I am curious as to your thoughts?
"I placed the mic and got it sounding good, but then I read the manual so I changed it. Now it doesn't sound as good but I guess that's how it's supposed to be. Anyway, enjoy your 'by the book mix'." :-)
I work at 44.100, from start to finish, so the conversion will happen only once. What the mastering guy I am working with suggested, and I trust him, is to export at 32 bits float even if the session is at 24.
Hi ChristianIce thanks for your comment! Which Mastering guy said you should export at a different bit rate to the one you recorded at? All of the Mastering Engineers I know would never recommend changing the bit rate.
@@Producelikeapro Bernie De Bernardi at Eleven mastering studio, Italy. He didn't simply explain why the floating points of the 32 bits allow you to export from a 24 bits session, but he also asked me 2 exports, one at 32 and one at 24. Passing through the same mastering chain, the 32 bits version came out more crispy, with better transients. Technically, a 32 bit float format also uses the same digits and the same 8 bits per point that a 24 does, so no interpolation happens.
Modern DAWs and plugins process at 32 or 64 bit floating point and that's totally regardless of the bit depth of the recorded files. So when you export a final mix that then the audio is in that floating point "world"... so it makes perfect sense to export mixes as 32 bit floating point WAV files. Your DAW will process in floating point regardless if it's set to 24 bit. The session bit depth is basically for all new files that are created within the session (i.e. committing, record and such), so I recommend setting the session to 32 bit floating point so all "work in progress files" stay in floating point. If you're recording from a 24 bit interface then recording to 32 bit floating point doesn't add anything :) ... and recording at 24 bit is then optimal. But even just moving a fader on that recorded 24 bit recording in the DAW will make it 32 or 64 bit floating point. So ... :)
About placing a mic in front of guitar cabs: You needn't run back and forth to the control room. Just put a single tap delay of about a second or so on the signal. Then have the guitarist play a second then pause for a second. Now you can evaluate the mic position on the headphones because you hear the delay of the signal when the guitarist pauses.
Here's the thing with higher sample rates. The higher the sample rate the farther the anti aliasing filters are out of the hearing range. This is all converter dependent. Some higher end converters/interfaces sound better at lower sample rates than cheaper converters/interfaces and higher rates. So to break it down and dirty- it's all about the filter design.
For mixing ITB its great to record in highest possible samplerate because of math. It gives you a lot cleaner tuning result, less sample transitions inside plugin math, reduses linear filtrating (a part of plugin processing when oversampling). So aliasing artefacts goes to not audible freqs.
When I do tracking for ITB software instruments like Omnisphere, etc. There is a strikingly obvious difference in sound between playback on an 88.2 or 96k session to a 44.1/48k session.Especially is the sounds have higher frequency content in them. It has to do with downsampling filters in either the DAW or the virtual instrument when the samples in the plugin are 96k. To avoid over-ringing artifacts, the downsampling filters will roll off just under 20k. Sometimes that takes off just a bit of the top end that is still audible. So, if I'm doing a 48 or 44.1k mix, I'll track the virtual instrument stuff in the higher sample rate, then bounce the tracks out and batch downsample them in iZotope RX7. The downsampling filters in RX7 seem to preserve more of the top end clarity. If I do it that way, then I usually can't tell enough of a difference afterward to make a fuss about it. A
can you unlink the sampling frequency of your daw and d/a? like telling your soundcard to run at fixed 96kHz and do your project in 48? its sound like the filter in the d/a ...
@@karlarsch7068 It's actually the sample-rate conversion happening inside the DAW (Ableton in this specific case), not at the hardware D/A conversion. If you're running a 48k session in your DAW that uses a VST Instrument with 96k samples, your DAW will downsample the VST playback on the track to match the session sample rate. Depending on the quality of the SR conversion in your DAW, you may lose depth and length in the high frequency content of your VST. I do at least. So, I just do my VST stuff in 88.2. Bounce the tracks out at the higher rate, then use iZotope's higher quality SR conversion to take the audio down to 48 or 44.1 for mixing. Doesn't take that much time and definitely sounds better.
Warren, I am partial to very good electric guitar sounds. While I generally don't have to use headphones while moving a 57 around to find my sound, something you can do if you ever did need to is to simply turn the guitar amp volume down very low while using isolated headphones like drummers have. Look for what you need while at this volume to find your placement, then remove headphones, crank the volume of amp back up, and listen to a pass in the control room to verify satisfaction. While the sound of a tube amp will be different at low volumes, it should not effect your choice of placement as you will know what character you are after regardless. You will end up having a marvelous day. I love your videos!!
The conversion in my shop is handle by a Lucid 88192, absolutely stunning converter with an incredible feature set. The fact that it supports 192, was certainly in the list of reasons I went for it over some other beautiful legacy converters such as the Lucid 8824, and the Apogee AD8000. That being said, I record all my projects at 44.1 and only work on mixes at higher sample rates, but knowing I have the future proofing is exactly why I went the route I did. It's all about options in this day and age.
Hey Warren the tips for recording a choir was very useful. I'm actually recording an orchestra soon in an auditorium. I will experiment with your advised techniques!
Yes, Warren is correct about the UA interfaces. I use two Apollo 16 , tracking a band or drums, I track hybrid style, committing plugs and hardware to the tracking session. UA is just wonderful to use. Tracking from 1 track to 20-21 tracks at a time. Just fantastic.
What do you think about bouncing the mix Online vs. Offline ? For sure there is a difference but is it worth waiting the whole duration of the track for the bounce? How important is it for the final sound quality?
Summary for the benefits of high sample rate: More detail for stretching and time alignment tools. More detail for a handful of plugins to Analog model with. Lower overall latency. No, you can't hear the difference, but after a full mix, you might hear a little one. In film and TV they need this high bandwidth to resculpt many sounds.
Yes, you can hear the difference. However, it probably is not the difference you expect. You won't suddenly hear violins in the background that you could not hear before. It is not that kind of thing. I know this can degenerate into a religious war, but I just wanted to point out that the differences are audible if you know what to pay attention to. The simplest way I could describe what to listen for is very similar to how to listen for improvements in clock jitter, for those who have tested external clocking on converters. The generic and vague description that comes closest in my mind is "ear fatigue". I found that it is easier, when doing an A/B comparison, to pay attention to the tiny spaces between the notes, rather than the notes themselves. For some reason, it takes your attention off of the "fundamental" of the note and allows your mind to absorb more of the space around the note where aliasing, harmonics and ambient background noise lives. However, all of this is very much like an experienced pit crew in the process of tuning a race car. Lots of pieces need to be right to have the resolution available to cross the threshold of perception. Personally, I can not go back below 192k these days. My ears clench up and I am not happy. Totally spoiled now, so beware if you start to spend all of your time up in the stratosphere. Ear Fatigue.
9:50 I think it would've been nice to answer the question more deeply : "Why do Film & TV ask for these higher frame rates ? ". Higher sample rate does not affect playback in fact but it's about having a higher resolution for post processing the sound.
One more benefit of 96 Khz over 48 and 44,1 - lower latency during recording. Buffer size is always in samples, not ms. That means that buffer size of 96 samples in 96 KHz is lower than in 44.1. That's more convenient for guitar recording.
2 inch reel to reel was at 15, or 30, ips, and the best Dolby -S system could only reach 13 bits ! Cassette Tape was 6 bits if you were lucky, so having just 16 bit supersedes the 13 bit other than the analog tape being more full, and thick warm, digital at 24bit well is very good now so 44.1 or 48khz is your best bet ,192 is just not going to make any difference other than huge files !
It's actually very simple. Record at 44.1 or 48 kHz or whatever YOU like/can. Then if someone says "we need 192 kHz", just lie "yeah, I actually recorded at 192!" then just convert and deliver 192 or whatever they want. There would be zero differences and if someone asks for 192 as a master or mix, they probably don't understand anything about audio anyway.
I’ve been deliberating for years over this as I used to own a Spirit 328 Digital, which I loved then... I found its limitations at 48k in light of the prevalent HD systems of the day... since then, I’ve been strictly in the box (making things sound “Good”, figuring things out, Scratching my head about what‘s been “missing”) until about last Spring when I came upon a Zed mixer on loan from a friend... I have since been conducting many experiments and am now personally convinced that Analog is the way!!! (Because of Physics, Vibe and Energy :) ) And I am back at the Analog-Summing Digital Mixer solution, as I would rather store settings rather than take photos :P ... I have chosen the SQ7 because it operates at 96k... and it Sounds Amazing, as a matter of fact. Very Happy!!! 🤗 The glue I have been missing for all this time since I turned my back on the Analog Process after the Great Digital Distraction of late...
I have found vocal mic placement does indeed work best about 8 inches or so from the singers mouth..sometimes with a couple of pop filters and maybe a windscreen over the capsule of the mic to cut down on breath pops. it gives more "chest" to your track.
The only time I found the higher sample rate (higher than 48 kHz) very helpful - when I would record something at 192 kHz, for example, and then slow it down significantly, like, five or six times. There are still lots of high end and Nyquist limit isn't prominent, like it would be with slowing down 44.1 or 48 kHz audio.
Simple experiment: take a project done at 192k. Render it. Make an equivalent project at 44k or 48k, downsampling as necessary (that is, take all recordings and resample them). Render. Do A-B comparison and null test. And for an intermediate, stick with the 192k recordings, but run the session at 44k or 48k. (I haven't done this, don't know exactly what the results might be, and won't because I'm a hobbyist who's happy with 44k and 48k, but this is the experiment I'd do if I cared.)
Hi Warren, I would love to see a video on recording classical guitar. I play the classical guitar in pretty unorthodox ways, with alternate tunings and such - and I mix it with cello, drums, vocals etc. But I've found that when recording classical, it's easy to get too much of the nasty lower mids which muddy it up. Realising that you probably don't record a lot of classical, I'd still love to see a video on it if you get a chance.
I never understand the 44.1khz, 16 bit guys, CD is pretty much dead, all music videos are usually 48khz,24bit. I run at that only because my CPU/RAM power cant handle higher rates with the amount of processing I am doing (plus samplers and such) and I often use a Behringer ADA8000 for extra inputs which is only 48khz. 96khz does make sense for the future if your system can handle it
higher sampling rate result in more noisy sound. That's generaly not what you want, so for the best possible quality you pick the lowest sampling frequency that still allow you to capture the higher possible notes that human can ear. Taht's actually exactly what the people at sony and philips did almost 40years ago and they ended with a 44.1kHz. More recent study ended up in the same ballpark. As i doubt either physics or human biology will change much in the next century so i don't see any futurproofing there. The only way that is futurproofing is if you have stoke of big audio electronic company, because all this is only a marketing gimmick to makes people buy new gear.
@@lolaa2200 is there any articles exaing why higher sampling rate is noisier? I have never heard this before and from my knowledge doesn't make sense unless you are talking about capturing noise that a nyquist filter would eliminate.
I have been recording and mixing at 96 kHz for over a decade. Recently a musician I recorded back then wanted to re-release their songs on a compilation album. Now that the music streaming services are heading in the high-resolution direction, I am glad I future proofed tracks. For me the higher resolution audio files allow for better EQ particularly in the high shelving area. Along with better pitch correction or pitch tuning plus slightly improved compression characteristics. As others have already stated, there are improved latency figures when recording as well.
You answered my stupid question...lol I was reading an old shure manual from the 60's ,for the first time. It belonged to my father. That's what inspired my question.
Self-producing is so difficult, I totally agree and have the same feelings/doubts as you! However, when I listen back to something a while later, I might kinda like it. Especially when I listen to older recordings (months!) I might actually like it a lot...
Hey man, love your videos. An incredible amount of useful information. Question though: You suggested using 192khz..what about bit rate? Would you as well recommend the highest bit rate? Or a particular rate? I imagine the higher the better, I'm still a greenhorn with all of this.
All samples, instruments and synth would be at 192khz 32 bit (I hope I’m not wrong), but reverb might be explainable… (This starts strong but then I fall asleep at the end!). Let’s compare looking at an image quality with hearing sound quality. If you were to draw the same size “circle” on a grid with squares and all the squares were either 1x1 OR 2x2 but you had to at all times combine the 1x1 squares to the size of the 2x2, then the “circle” will have the same dimensions either way. Think of the circle quality as a sound or song bit rate and the squares as the sound affects, now: using only 1x1 squares, you can fit more affects into a 2x2 area because the higher bit rate removes the margin of error, a possible value of either 1 or 0 (bits are binary) in a 2x2 square becomes half as random with double the information. I assume the quality difference between the DAC and speakers are made up for with aliasing, like a circle on a TV because I am understanding that for every two sounds a driver makes, a third sound is made in the transition, an event that becomes half as random but twice as common when doubling the driver bit rate. So to answer your question: a synth, etc; to a producer could be PLACED more accurately at higher bit rates if the sound were slowed down by 100s of times(higher bits means more changes can be heard or better circle accuracy), if it effected the quality then there wouldn’t be a market and you wouldn’t have downloaded the VST (with exceptions to bit-crushers for LoFi music). If you tried reverbing a reverb then maybe that would sound like shit but then you wouldn’t want that in your song anyway. I honestly received my caffeine supplements in the mail yesterday and had no clue about bitrates/depths 24 hours ago. Just take LSD and DXM with Mary and you will become the sine wave looking into the future.
48/24 is all you need. In the future there will be converters who will solve everything you can and cannot imagine today. When it comes to mix placement it’s all about the power, register and flavour in the voice. Everyone is unique with different sweetspot.
Warren there is software that can convert your sample rate from 44.1 to 192 in your daw or whatever sample rate ... and back again. Alot of flim work here in Georgia and I just give it to them in the format they request. This does not mean I recorded at that rate. No different than mp3, wav. flac or AAC I use 48k to record myself but can convert the whole song or just the mastered file to 192k in seconds.
If one has fold back problems, yeah, why not? Theoretically yes, it would fix a lot of aliasing introduced by a lower sampling rate. We can t hear the difference really, yes between the sampling rates, of course. But these are 2 different things ;)
It’s funny, not so long ago I was recording and producing at 48k always have for around 13 years. Looked into the mad science of it all and came to the conclusion that for me personally there is no point in going higher as the final format will be at 48. I was soooo stubborn about it. Then I tried 98k and.... everything sounds better my piano sounds way better, distortion sounds better, the little grit you put on things sound more natural. I looked into it and it’s something to do with harmonic distortion and the way upper inaudible frequency bounces back and effects the audible range. Forgot exactly what it was. Plugins, samplers everything works better together in the protect at 96 over 48 and that sounds better down mixed to 48. It just does, hard hitting piano for example has a kind of harsh distortion in 48 in 96 it’s gone and sound natural even when down mixed to 48. Plus I get way lower latency at 96. There is no reason to go back to 48 for me. I have the computing power to handle it well. Most modern computers do. It actually teaches me to think more about cpu usage and efficiency in the mix and not be lazy by wacking pluggins on every single track and to use buses more. It seems less about lows and highs and clarity and more to do with the way your mix interacts with each other and artefacts when things are pushed hard.
@ReaktorLeak no, your not understanding what I mean, the way digital plugins work with sample rates. nothing to do with converters or interface. its on the software side.
Do you generate tab when you are writing or to provide to musicians that you hire? If so, what software do you use? Guitar Pro 7.5 looks interesting. Is maintaining tab or scores more useful for copywrite purposes than DAW work product?
I was just at a workshop and I learned that tracking in 44.1 or 48. I learned that it is easier to up sample the audio for mixing than it would to go the other way.
I was at the Recording Summit at Welcome to 1979 Studio. Anyways I was attending a panel on the topic of "FAQ" and on the panel was Maggie Luthar, Mark Needham, Pete Lyman, Chris Shaw and Chris Mara. A questions came up about samples rates and Pete and Chris talk about tracking at lower rate is ok and then up sampling to edit. They went into some technical talk and explained some examples how it works. I will be honest some of the technical stuff was over my head hahahaha, the good thing is now I have a new rabbit hole to venture down and learn more about it for myself.
Upsampling from 48 to 96 or even 192 is pretty easy and it's looseless (useless also but hey if the one who pay want you to do stupid useless things... life is a circus isn't it ?). However upsampling from 44.1 to 96 will be less straightforward and might end up as some quality loose (minor one but still there) As of the bit/sample you can add bit without problem, if you do it with dithering it can even somehow improve the perceived quality of the sound but certainly not the fidelity, so ideally it should only be done at the very end of the process, not before mixing. Ho and by the way that's what is done inside most of your ADC chips (they are not realy 24bit inside, but shuuuush, that's a secret ;) ) Just my 2cts...
Mr. Huart. In a hypothetical scenario in which you're working at 48K (you recorded the vocals and guitars), but someone else recorded the drums, bass, etc. etc... but the other person recorded at 44.1K and now you have to mix and master everything, will you up sample or down sample ? or, will will you just keep you're system at 48K and working like that ignoring the fact that you have files at two different resolutions?
Depends on clock jitter characteristics of the interface. Some work better in 44.1 or 48kHz than at the higher rates. If jitter is an issue, then a high sample rate will likely sound worse than at the lower rates.
good call on the mic placement...go the distance to check the sound in the control room - kinda reminds me of the advice to put the pallet 20 ft away from the canvas, also to take a sec to look at the effect of your "move" on the bigger picture (what does the change in mic placement do within the mix?)
If you can’t hear the difference between 96/24 and 44.1/16 your hearing resolution is insufficient to be skilled at adjusting EQ or compression. Work on your hearing resolution. It takes practice.
ChristianIce I didn’t mention anything about hearing beyond 20 kHz. Do you know what jitter is? Any amount of it causes the Nyqyist Limit to be variable per the program. This causes intermittent phase distortion anywhere above 1/4 the sample rate. Meaning 80 kHz is the minimum sample rate to reliably playback 20 kHz audio.
@@audionmusic3628 The real difference happens when you convert the sound from analog to digital. If the conversion sucks, a converted audio at 96K will suck. If the conversion is good, it's good also at 44.100. Higher quality makes for a better manipulation. The final result will be at 44.1, so unless the standard will change any opinion about it is irrelevant.
@@audionmusic3628 Whatever, kid. I am just sorry you will have to suffer the "intermittent phase distortion" in all the music that exists. Listen to any CD must be a torture to you :D
Hi Warren, Thanks for tackling the high sample rate debate again. Always good to hear some clarity on the subject! * *ducks* * 👀👍🏻 What are your thoughts on the Sonnox Dynamic EQ AAX-DSP vs fabfliter EQ or Ozone variant? Have a favorite? Marvelous video.
Great video as always! Just wonder why nobody talks about the metric halo converters. They’ve been rock solid for me and sound great! They also do the same as the Apollo units (different plug ins that aren’t as sexy) as far as tracking with the plugins on the mixer engaged. No latency issues at all. Just spit-balling I guess, but worth a look for anyone looking to get a converter. I don’t work for them or anything, just seems like another good option. Cheers!
Produce Like A Pro all I know is there based out of the US, and they sell a wide variety of converters (that come with the mixer software, downloadable from their site for free) from 2 channel interfaces to 8 channel and they’re all expandable to taste. I’ve had mine for 8 years now and have never had a problem, their tech support is great in case there happens to be one. I’m pretty sure their units work with just about any daw (unsure about pro tools, been a while since I worked on that, but since they’ve switched to the subscription service, I think it’s usable on there as well) and the mixer software and the hardware are bullet proof.
That cut me off, lol, sausage fingers over here. Definitely worth a look if you’ve got a moment. Like I said, I’ve never heard any channel talking about this stuff. Cheers!
The amount of experience you share with us is invaluable. Thank you! As a follow up question regarding recording with highest sample rate, I would like to know one thing. Wouldn't it be enough to simply use sample rate conversion from e.g. 48 kHz to 192 kHz and then send it to whoever is interested ? I mean situation when track is already finished and you still want people to be able to use it.
the higher the better....going to higher rates means less ringing of the digital brickwall filter...even running digital eq in the daw sounds better when run higher.
Once I time aligned 3 mic tracks to the 0.00whatdoIknow1 ms, then - not content with the result - put different delays on them just to end where I basically started with the raw tracks. So I guess, it is really easy to overthink this topic.
Something I've not seen anyone do is some micing demos for Resonator guitars and basses. Wondering what your preferred method would be and if perhaps you could do a mic location comparison like you did with acoustic guitars.
Question for those of us working mostly alone. What advice would you give a performer who is also their own engineer? On occasion I lose my inspiration by the time I get things setup and checked. I have learned to leave my amp mic'd up and a vocal mic connected and ready. Any other tips?
I would say make sure you have a basic understanding of your DAW, use track templates and know how your gear (especially if working with some outboard) is setup and routed first. You want to make sure you can open your DAW, select the track, plug in your instrument and hit record. Write down your good signal levels and other settings for reference so you can go to them if needed. Record a few takes if needed then move on to the next layer until it starts to get where you are either satisfied or just making noise and walk away, because that means you have reached the end of your creative stream for that moment. Stay away for at least several hours or a day then return.
Its a "capture" issue. If youre capturing (recording) at a higher sample rate youre simply gathering more information (better sound) during the "RECORDING SESSION" =) . THAT is what then gets dithered down to 44.1 (or..if at all these days). Same as "sh*t in-sh*t-out" really. Source is king. Love your vids man. Keep Rockin'
I think music should be produced / mixed at a higher sampling rate than what it's released at. One reason is that it allows your plugins to render at a higher rate. In particular, analog saturation plugins can create gritty digital artefacts at 48kHz and below.
I use the pinky out sipping tea method, or thumb in mouth end of trumpet (pinky tip) facing mic giving perfect distance and elavation. thats just my op for vocals :)
Unless you failure was to send +48v phantom power to your brand new 2000$ passive ribbon mic, then reading the manual won't save you the 2000$ you just throwed away, it would only tell you you shouldn't have done that. 'oupsy'
I would recommend using minimal compression on your master bus. Leave room for the mastering engineer, plus, you don't want to realize too late that you over-compressed your mix and after it's done, you can't undo that. You can always add more compression later, you can never remove compression (expansion never sounds right). Warren, would you agree?
Greetings and apologies if this is a dumb question. But I am trying to decide between a guitar multi fx pedal that records at 44.1 kHz and one that records at 48 kHz. Which should I get. Presumably, I might need to upsample in the first case if I am doing video work, and downsample in the latter case for music work(?). Assuming I have understood this correctly, which pedal should I get. Of course, the option of recording an amp via a mic is always there, in which case I presume the audio interface’s sample rate kicks in. But I do live in a noisy area so yeah well. Any advice would be welcome, thanks for reading 😊
I have a question 🙋🏻♂️ I'm not a professional and I'm making electronic music, I have external synths but mostly I use VST plugins, I do learn a lot listening guys like you and reading your fallowers on the comments, thank you so much to everyone. If I'm working only with VST plugins everything inside the daw not recording anything, For reproducing the sound from my daw to my monitors, do I need an audio interface over 48khz to have a better reference/real audio or a decent average audio interface like NI komplete Audio 6 mk1 or my #TascamModel12 is enough? Thank you again.
Excited about the upcoming UAD episode! As the owner of some heirloom quality hardware from Neve, API, UA, Avedis, Kush etc, it allows for some easy comparisons to the UAD emulation. The UAD stuff is amazing.
Great question! No, recording at a higher resolution is a great idea! The only thing holding people back is their interfaces and very often the power of their computer which may not handle large sessions at 192 with tons of plug ins and track count
@@Producelikeapro You mentioned about using audient a lot in your projects, does the 96khz rate on the audient enough for futureproofing? I really like their newest line but its not 192 yet
My only commend is you are the very best at what you do and as successful as you are, it hasn't gone to your head. And what amazes me most is how you have not lost "the love" for what you do after all of this time. Keep it up, you're an inspiration.
This is probably a stupid question but... If I'm using a lot of samples in my music, all of which are 44, but my daw is set to 88, with that change the way my samples sound or d0 anything undesirable? Thanks for the video
Thanks to producing like a pro man. I come back once I a while for some new content and have to say most of the time I learn something new ! thank mate for the effort and free stuff. One more thing all people have to invest in acoustic treatment first is my general advice otherwise you can't even hear compressor and eq work. I made 4-panel bas absorber and Jezus sounds like heaven in hear! good day, all.
Also, idk what music school people went to... But I've never tracked a vocal 18" away lmfao. I was always taught do what's needed. If you need more warmth, move up, if you need less, move back. If you're clipping trying to get the sound you want, built in and in-line pads. For me, at 18", that's like an Omni in a room for BGVs or something lol, not a lead vocal.
2:03 I agree with Warren, the UA interfaces sounds great and the system gives some good advantages. The plugins also sounds great but is the actual difference “huge”? No. It's not if you use a few of them. But it might be a bigger difference when using a lot of them. For instance the API modelling plugs sounds clearer on the UA plugs compared to Waves' versions of the same. In the end having many tracks and many plugins probably makes a bigger difference. Also worth having in mind is that UA plugs are very very processor intensive. Some are putting ridiculously high load on the processor. Some do it less, but all of them needs more than any average high quality plugins from e.g. Softube or Brainworx. No strain on the CPU though, but worth having in mind you will get a lot of latency added if you are used to mix in many plugs to shape the sound while tracking and producing when using the UA plugs. I think the best way is to sprinkle with UA plugins and use other plugins for the most common tasks and mostly when mixing and avoid them when still producing and tracking. Unless you track with the plugins inside the Console with near zero latency. At the end, would you really hear the difference between a mix made with UA interfaces and plugins and another high quality interface with other plugins. I'm sure some might promise you they will. Personally that's pure BS. A good mixing engineer will get good results anyway I'm sure the upcoming video from Warren will cover most of this and show good examples in the tracking session.
There is a commercial reason in what UA is doing. They design their systems to be as closed as possible, like Apple does. In that way, they try to 'force' you further into their system and can ask ridiculous prices. UAD plugins only work on UAD platforms. And the power of their SharC processors isn't high, so they sell Satellites which are just SharC processors, but at inflated prices. UAD is also customer unfriendly (like Apple) with their limitations on device and system limits. Their products are really good though. I just don't like their closed system business model. Now they have released the Heritage line, which is the exact same product as their Apollo line, but with 2 or 3 extra plugins and inflated the prices of their interfaces again without improving anything. One day, they will face a point where customers will say 'enough is enough'.
After watching your video I decided to start recording at 192khz. My pc can handle it so why not ( i7 9700kf, Msi Meg Z390 Ace, samsung 970 plus, etc.. Clarett 4 pre usb interface.)The question is can all plugins work well in this resolution, can microphones and all hardware work well? Are they made for such high resolutions? Thank you 😊
Some plug-ins can proces 192kHz. Unfortunately, most of the plugins can’t. I guess this is the point of return where hardware and hybrid mixing comes in place. That being said, it’s just a matter of time when plug-ins are coded 192kHZ ready.
Thanks Kevin :) I started my first project at 192 kHz and everything runs smoothly. The great think about this resolution is that I don’t have latency problems at all :)
That implicates that you had latency before which is strange because if your system can deal with 192kHz, it surely can deal with 48kHz. Anyway, glad that you took the step to high-resolution audio! Have fun!
lola a If it works for you, then thats fine but if I’ve found that if I have to do heavy pitch manipulation to a sample there will be more artifacts the lower the sample rate.
How much we really compromise with audio quality by downgrading sample rate to 44khz 16bit from any higher sample rate just because music distributors want you to do that to distribute on different music platforms even though we knew that we aren't living in the era of CDs anymore. Even after you master your track on your own at higher sample rate and then you have to hire the mastering engineer just to compensate that lag or quality issue in the audio by downgrading your track? Not all musicians have that budget to hire the audio engineer. what's your thought?
I used to track at home in 48k but just dropped to 44.1 recently I even did the “save copy in” bit to re-save previous work in lower res, I find fewer tech problems with native, fewer cpu error messages I feel most home studios are sketchbook territory and with many budgets not allowing for HD, (which would at least let me use my ram cache I got 64 gigs) Uad2 plugs and Apollo quads and octs , manley and lexicon hardware and Neumanns..that if the material is strong enough you will be most likely tracking in a pro shop with someone like yourself at the helm, I haven’t heard anything that can’t be reproduced and most mixing and mastering not done in a reputable house is usually a mess anyway, Izotope just won’t cut it, it’s this thinking I feel that opened the door for daws like studio one, lots of toys bells whistles that kinda work for people with budgets with no studio experience and usually not much music theory so they don’t have to know how a aux or vca actually works it’s just created for them and they can use chord finder or auto scale junk inside many keyboard vst apps since they most likely are dropping loops or whatever and will probably never see a physical patch bay, what’s interesting to me is the bizarre circle of silly-I started in 69 in north philly (John Coltrane house just 7 blocks from my own) as a session and gigging guitarist and tracking was direct press vinyl or reel/reel 4 track then a mix of adat vcr tapes and cassette portables, when this whole digital thing shook down to be affordable the super clean articulate was what everyone wanted imagine practically unlimited tracks no bouncing,,fast forward to the soulless sterile presses and masters,,then digital remasters of some of the greatest music ever made”fast forward to today’s “analog” plugs to make our new tech digital sound like the stuff we were using in the first place and way before me being around but all of it being spit out of a smart phone or computer so right back to zeros and ones and now I’m seeing new vinyl at hipster shops almost as a novelty, some weird shit bro..
So as a lead guitarist self producing yourself would mean you have a little angel on one shoulder whispering "play for the song" and a little devil on the other shoulder going "what would Yngwie play'?
I couldn't remember why I had recorded at 192 starting probably more than tens years ago, but this thing about movies rings a bell. I didn't read microphone instructions either. However, with some of those newer mics having a front and back maybe I will see if I can scan the instructions with my phone. Hee!
What do you think of the presonus family off products in a professional setting. Are the X Max pre amps comparable to what you might use inn a professional studio? I'm looking to start my studio in my basement and am wondering if sticking with this product line is good long term
Long answer short: Unless you are a dog or a bat, you don’t. 44.1 kHz contains all the frequency range perfectly recreated, any human can hear. Sampling rates higher than that won’t result in better audio quality. 😅 Nyquist is your friend. Recording in a higher sampling rate can actually result in a worse audio quality. Why? Because building ADCs capable of higher sampling rates is difficult and expensive. Especially, if you are using a prosumer device or something on the lower end of audio devices, you can expect distortions in the ultrasound frequencies that can bleed down to the audible frequency range. And the way sampling rates work, I guess using a decent upsampling algorithm will yield results indistinguishable from „genuine“ higher sampling rate recordings.
You glazed over mic'ing a guitar and the little you said sounded fascinating. Please make a mic'ing vid on all your wealth of experience. Or multiple vids on different mic'ing situations.
Other reasons to use high sample rates: 1) avoid aliasing with plugins that don't have built-in oversampling 2) if you pitch down a sample as a creative effect you still have high frequency content (without an unnatural cut off at the top end)
for the 2) that's because above niquist/shanon's frequency you have nothing, pichting down nothing will result in .. well nothing if your pitch down plugin editor is not to stupid. If you listen to that track solowed in headphone you will here nothing on the higher end and that's probably what you refer at when you are talking about "unatural cut off...". So what about with a higher sampling rate to start with ? Then on the high end you have noise, those noise that are inaudible will eventually become audible if you pitch down, so if you solo the track in headphone you will notice a difference. Ok but what about in a mix ? If you pitch down that's to make that track occupy a lower portion of the spectrum, you will for sur place some other instrument/ track to occupy that top end. So now not only you don't have this "unatural sounding cut off" anymore but you have less noise ! That's a good thing !!!!! Not to mention that would only be a thing for extrem pitch down of super high frequency sound, let's face it that will almost never append. But if it does and if it's still a problem for you, just add a white noise track and eq it and the pitched down track so it blends together, you will have your "natural sounding" effect in your solo headphone listening. Well of course your mix will be lesser quality but maybe that's not what matters here. Let's not even comment on the 1).
I am not a professional producer, but I am a decent musician. I ask what is the 'downside' of using a higher sample rate? Is it a question of resources or cost of equipment? Frankly, while it is desirable to record with the best mic possible and converters, I think the quality of the talent being recorded absolutely dwarfs any perceptible difference in sample rate with current technology. This is just my opinion and admittedly I don't have experience beyond home recording, so perhaps professional producers could educate me about the issues.
I heard that If I record at 48, and then I want to put it on CD or for streaming, would I have to convert the music down to 44.1 , would that introduce noise into the result? is that right?
Personally I don’t use one because I mix hybrid through a console. However if you need one in order to access hardware compressors etc I completely support it! However, if you aren’t using outboard I would personally stay in the box
