In the Bounce to Disk settings of Pro Tools you can select an option called 'convert after bounce'. This makes sure you bounce in the session's sample rate before converting the result of this bounce to a lower sample rate.
you're absolutely 100 percent right about the details. its the same with any art form, attention to detail (without losing overall perspective, of course) is the thing that really shines through with its cumulative effect.
Ok people here is what I recommend as a professional studio engineer. As you record each of your tracks, add your effects, automation, etc. Then, right click the track if in Reaper and select Apply Track/Take FX As New Items. What this does is that it lowers your CPU use so that before you start the project you can pre set the project as 192 instead of 44.1 This turns your track into a .WAV. Remember though that if you do this and you want to edit your tract you can revert by turning the track to the following state called Active again. When you have done all your mixing and you want to render/bounce then just bounce your project as usual that will each be a .WAV and you'll have a great sound rendering it as 192. My favorite mastering plugin at a reasonable price recommendation would be the Abbey Roads Mastering Chain and select the preset called Balanced. Here is one instrumental where I played all the instruments and did what I recommended to you guys. P.S. the guy throughout the video is not me and you can see what my real name is. Cheers. Frank (New York City) ruclips.net/video/D6hZtFMx12o/видео.html
Thank you from the bottom of my heart! This is something that has bugged me forever. I'm just a dude in his bedroom playing around but I want the best quality that all my amazing PC can put out :-)
My first finger-up for you, - hey man, you‘re able to describe important things, incredible : go on with this- your nervous style gives this sweet appearance of reliability :-)
this is the topic where ify pondering and guessing youre alright will set you back years. youll get all invested in gear and work all day, ruining every mix by bouncing wrong. youll even stop checking the mixdowns, the repeated truama of having your vision destroyed will be too much to risk. you start rendering at -6db and hire a mastering engineer eventually, assuming you have no ears and never did.
It's like this I always say people never even finished quotes because they don't even understand too take everything with a grain of salt means if you like the taste season your food with it not just judge everything and be skeptical that's why I appreciate your videos man thank you
"foldback distortion" refers to a kind of waveshaping, not aliasing. check out the free vst Driveshaper for an example of how foldback distortion sounds.
I am so happy you are doing videos that take on this subject. Solving these kinds of issues have been the keys to me finding a clean sound on a DAW. You have certainly aided in my understanding. Thank you.
Hey there, actually heard about that only a handful of producers tempt to organize their projects like that. I was never certain if I should do it either, but this video gave me a clear answer. Thank you for that.
Only reasonable thing to do now is to make your next video a blind test with different audio samples at different sample rates to prove that it matters and actually sounds better. Perhaps you would not be able to tell which is which in some cases. Make examples for us as well. Play A/B parts bounced out at different sample rates, then play them without showing which is A and B, then we can hear for ourselves. I have had so many discussions with audio engineers that claim this and that, then we do blind tests and they make a fool out of themselves. Would make a fun video either way :)
Yes, I agree and I was all was always taught if you can't hear it don't do it. Makes no difference to me anyway as I run on a core duo with 4gb ram, it only just copes with 48000.
J W while i agree 44.1 is fine, there are benefits to going higher at least in the studio and they have to do with pushing aliasing artifacts out of the range of human hearing increasing the quality and authenticity of analog emulations and synth, saturation and distortion effects, also reducing latency,
In Pro Tools, I have always bounced to disk at 24/96 stereo, then brought the resulting file into a 24/44.1 pre-mastering session, letting it down-convert upon import using the best settings. I then pre-master my final to 16/44.1 with medium dithering.
your best to keep all files at the higher recorded rate until after mastering and then convert them for specific mediums like cd via good dithering software .Your approach is wrong totally as downsampling via import is both a substandard quality and also its pointless prior to mastering where the higher quality will allow better subjective evaluation of the music for mastering.You should keep the main master at working rate and secondary masters at lower rates off the back of the finished master.
@@cornwallradiophonic6250 You are absolutely correct. Nathan's approach is so weird and unprofessional that I was wondering whether he was a troll? By the way: use Voxengo's free R8brain for SRC - it's better than most DAWs SRC algos!
@@cornwallradiophonic6250 Those are good suggestions except for the understanding of the sample rate conversion on import, which uses exactly the same set of algorithms (for the various speed vs. quality settings) as if you converted when bouncing, at least in Pro Tools, which is what I have used for years.
Hey thanks , I know this is an old video but it really helped me . I have been struggling to render a finished track and getting it to sound just the way I hear it in the mix . This works ! -cheers from Canada
Wow, thanks! Bringing up the project sample rate to 192 before bounce made it sound a lot better and/but showed in the sound overuse of the soothe plugin, because im taking away harsh sounds that only appear in lower sample rate that im mixing in...
Great video, I love the technical science side of audio and this is very useful information. Sometimes I see people post in forums about when they render a file it sounds different than when they were mixing it and I've always wondered what could be the cause of that and this could be one of the explanations in certain cases. Also what frequency analyzer are you using with all the pretty colors?
Awesome thought provoking video. If only my CPU could keep up with realtime 32x oversampling all all plugs at once. Is there was way to pull this trick off in Ableton?
Ableton 10 automatically renders in higher sample rate and then exports at whatever you choose. If you set it to high sample rate in project and then go to export it tells you that it will render at higher sample rate
Kclip Zero bounces at 8x even though it has no switch. But its weird because it sounds different then what you were working on. Which can be confusing for beginners. I believe Sytrus also does this in FL. Which drove me crazy when I was younger because i really like the aliasing sound for background dubstep effects. But for anyone who is thinking of buying flatline or knock or standard clip, I highly recommend FreeClip. It has 32x oversampling switch and beats out most clippers in sounding analog because of it. It also has 5 different softclip modes. The quintic setting sounds very proffesional when bounced at 32x. Other plugins might seem better because they have a percentage of how hard to clip, but nothing comes close to FreeClip. It just sounds so clean and analog. I stopped using Kclip because of it. Kclip is so good on paper but it sounds digital and almost exactly like Fruity Clipper but with more options. Some guy did a comparison and showed that FreeClip has the least digital artifacts out of all of them, even though it's free. I've never tested flatline myself, but it seems to be snakeoil. Maybe I'm wrong.
100%... Actually you're correctly thought out on everything I've watched of your channel. This video educated me on some questions I've had, great work!
Man, this was the best tip ever! So I have to condition myself to think of setting 192 kHz in the 'project settings' before rendering. While producing/mixing I would leave it at 44.1 because otherwise my RME (or let's say my CPU) can't handle this amount of processing well.
This kind of thing is what makes people record a bounce inside the DAW in real time. Sending Mix Bus to an Audio Track then press Record. Your recorded file is your bounce
It is great to know that it actually makes sense working at higher sample rates when you are processing and harmonics are newly produced (is this right?), and it is a bit weird to see that you need to ask Reaper to convert after exporting, I thought that was a given, Ardour and Mixbus are not as flexible as Reaper but this is not a conversation needed there. - Quick question: assuming you were working at 44.1Khz, how would you avoid these fold back harmonics, does it make sense adding a hard LPF on the master? Buses? Tracks? Nowhere at all? Sorry, this is not the most simple topic 😅
In my opinion and from my experience there is no need to use plugins only in mixing ( it is all about balancing the tracking) .. once you hear everything in a perfect way , your job is done which leads to less is more.. in mastering yeah you need plugins that adds taste to the sound
After reading one book about how Digital processing and how things are converted, it's become a primary concern . Right next to Acoustic Treatment and Ear training.
Cool, another video :) I love your Snake oil vids but these really set a balance and show that you really know your stuff. Very informative and interesting . I use reaper so this is useful for me, thanks.
HQ offline rendering is supported by some plugins, and several of the clipping plugins i use (standard clip, v-clip) offer internal oversampling to counter aliasing. Setting my interface to 192kHz uses a lot of cpu, so I’d rather stay at 44.1kHz and use oversampling where needed (in the plugins themself).
It would be cool if you made a series of process checks that helps illustrate different concepts like this video, with example files and different scenarios to help drive the concepts home.
Thank you so much for your info on aliasing. Being aware of this issue has had me second-guessing my use of saturation plugins to ensure I'm not muddying up my busses. Plus, this episode may help me to figure out why my mix downs, in Reaper, always come out sounding more muddy than the project. I assumed it was the down sampling, but more specifically, it could be the aliasing caused by these settings. Much appreciated. Keep up the good work.
I think that the reaper test tone generator it is not band-limited. With "normal" VST instruments, you shouldn't see that enormous amount of aliasing. I tested it by initializing a Retrologue patch, setting OSC1 to triangle and generating a test tone of ~ 16.4Khz @ 44100hz sample rate: no aliasing. At the same time, I agree with you that with higher SR some virtual effects and instruments sound a lot better! Cheers :-)
Great info!, I feel that maybe is a lot to ask, but many of us that follow you could really benefit if you could do the same experiment of exporting and checking distortion in different DAWS. I feel many of us want to improve our sound, but in regards to sample rate conversion we are a bit lost; plus, the options for exporting are not the same in every DAW. Maybe if you can help the community you have built many of us would be forced (in a good way) to jump to another DAW to mix the project, like e.g producing in Ableton or FL and then exporting the session to Reaper or Logic or whatever has the exporting option that has the less artifacts. Thanks for all the information you are constantly delivering! Greets from Argentina
I've seen a few of your videos, this one made me subscribe. I was not even aware that rendering at the project sample rate wasn't automatic. Keep it up!
Hello! Big Fan! That happened when you bounce, how about when you print and then export the audio in a different sample rate? o when you print and then bounce without any plugin??? it would be nice if you can do a part 2 of this video showing what im saying! THX!!!
Curious to know people's thoughts on Bouncing to disk, or just printing to an audio track within the session, then exporting that newly made audio clip. I can say with 100% confidence, that bouncing to disk will give you a slightly different sound to sending your aux mix bus to an audio track and recording/printing that in session. For those who are skeptical, try both of these methods, then open a new session with both of the audio files on seperate tracks, and then phase invert. In theory, the result should be silence, however it is not. As for which method is better, that is subjective, however I have found to take a liking to the printing method as opposed to 'bouncing to disk'.
Very true. Great video! I completely agree with - I want a bounce that is exactly the same as what I hear. The only way to get that, that works for me is to use my virtual channels on UAD Apollo (and I'm guessing you can do this with other I/Os that have more advanced mixers, RME comes to mind) and record back to DAW ( I use S1). After that I use Ozon standalone to covert it to whatever format I need...
@@dodo.danciu Hi bud! So, the concept is simple: You need to go from your DAW to UA console's virtual channels and then go back to the DAW by making a new track that has those channels as input (and mute them during recording). Most of the DAWs will automatically assign the available inputs and outputs (if not then add them yourself) and then you only need to link either 1 and 2 or 3 and 4 Virtual channels in UA console for them to work in stereo. I hope I was clear enough :)
This is the message I've been trying to get across to all my friends who are kind of on the fence about how the world is going complacency is the devil whether or not you believe in one or not it is the very thing that destroys Humanity diversity and detail is everything because we are all the same yet so unique! Wake up people!!! Be different, search for YOUR sound👊👊👊👊and keep doing what you doing man I appreciate your videos and what you're doing
I would love to replicate this with Cubase but I don't know how. You cannot change project sample rate without converting every file. Any helps will be much appreciated.
you just made my day with video..we are learning about this rn in my audio course and this rly sums why i love to put high sample rate for projects, and we render at 44.1! :D thanks!
I had to rewatch the part where you say the trick, this looks really good. Btw, what happens if you just literally put a brickwall filter at the end of the mixbuss chain and render like that? Would it eliminate aliasing distortion too? One recommendation for high quality resampling: Use Audacity, it has SoX (one of the best resampling algorythms and for free), so the "resampling distortion" should be almost inexistent. So for this, render the mix from the DAW at its original sample rate, import into Audacity, set the propper quality from Audacity settings and the target sample rate, and export.
I’ve found myself getting ridiculously obsessed with getting rid of aliasing recently (doing crazy things like running at 192 at 16 or 32x oversampling) but then I think.. often it’s only really at a problematic level when you’re really smashing it to destruction, and if you’re just using mild saturation or something, perhaps all that is really achieving is adding needless processor load. I dunno.
It depends, if your doing any higher frequency saturation then you should oversample but really 8x oversampling is enough in almost all cases, a couple instances of aliasing distortion in the odd track or two isn't really an issue it's when you start using tonnes of saturation on busses or your master that it's a problem because it's introducing inharmonic frequency content along with the innevitable intermodulation distortion that occurs with buss processing, if your subtle with it, it's barely an issue but it's certainly something to be aware of along with things like the nyquist shannon theroem and bit depth.
Wow that is incredible. I had no idea that Reaper was doing that in the background. I'm going to bounce at my native resolution from now on and then do the conversion separately.
Thanx for your description. To have it crystal clear: Do I have to tick or untick this option "Use project sample rate for mixing..." ? Any help will be appreciated.
Thank god you didn't assume everyone knew this, no wonder I always thought my down mixes didn't sound the way within the DAW. "but" now questions arise: If I am working in a 192 khz environment, should I set plugins that oversample to that oversampling mode? Or that plugin will work as it should if I am in that 192 khz environment? No wonder I wasn't liking Reaper, if only I knew this tip sooner (I switched to Studio one pro and yes, I love it). Those back harmonics/distortion is so irresponsible from developers... sigh...
Wow. This is actually really interesting. I am working in 48k for Processing power... I think i will try this next time and Render at 192 k. Should work If i understood correct.
ummm... As a mixer with an older computer, I'd rather be able to use more plugins and more tracks. I have to do tons of bouncing, splitting projects into separate pieces, disable all kinds of stuff as it is just so I can keep working and I'm thrilled when I can even barely finish a project on 44. It feels like I performed a slight miracle every time and I deserve a medal or something. Given the choice, I'd rather run a high quality convolution reverb (which I can't do) then run at a higher sample rate.
I use 48kHz as the project sample rate and also bounce the tracks in 48kHz. This is a really interesting topic. I will definitely try out sampling down from 192...
Would a really simple way of doing this on your final mix not just be to bounce out the entire thing in 192 and then open it up in your DAW again and bounce it down to 44.1Hz? So all the plugin processing etc will be done on your first bounce. The downsample conversion will be much cleaner?
NOTE: Aliasing only affects non linear processing. There's no point in doing this for linear processes like EQ (unless decramping near Nyquist is the goal). Oversampling smears small transients in the time domain, so shouldn't be used unless pros outweigh cons. Here's a tip - if you want clean audio, don't put "saturators" everywhere. "Saturation" is a recent sales gimmick used to make people buy more stuff they don't actually need. It has a valid spot in the workflow, but even with mitigation techniques, over-use of digital saturation is a recipe for bad sound , not a panacea for digital vs analogue folklore.
DigitalDiggo 100% with you regarding saturation plugins. I run a hybrid studio that tends to record live instruments. I’ll run some saturation here and there on a channel to help a single note guitar or snare or whatever cut through a mix, but I can’t imagine it on a bus. But, having said that, no rules baby! If it sounds good it is good. So many of these very helpful videos are always gear, gear, and more gear. You need quality equipment but the key to a good mix is a good arrangement.
As a DSP scientist, everything you said is correct. The time domain will get smeared as a result of oversampling. But you missed one small part of this: the time domain ONLY gets smeared as a result of calculations of DA/AD converters. Essentially, the result of bouncing a high project sample rate, and then converting back to 44.1-48khz will indeed result in a cleaner signal after exporting. You can even go from 44.1-48khz to 192khz, bounce, then downsample back to again. The affects of increasing a project sample rate are null. Also, what you say about saturation have never been a gimmick so this is misleading. Digital distortion can sound just as good as analog distortion if not better these days. You just need be careful with oversampling. The higher the internal oversampling, (especially linear phase), will cause continous smearing of transients because of the affects on time domain.
@@shinyukomusic Small world lol! Not suprised to see you reacting to this comment but the fact I happen to check this vid for the first time and then see you responded to this comment one day ago… what are the odds? 😂
Some Voxengo Plugins can be set to use maximum oversampling in bounce mode. I don't know how well this works, but since I got plenty of CPU I usually set these to permanent 4x oversampling anyways. Which results in 196khz processing inside these plugins since I usually work in 48khz in Mixbus.
If I’m recording, mixing, and bouncing my tracks in 44.1 will this fold back distortion apply since everything is kept at the same sample rate? Or does this still apply? If so will filtering everything above 20k fix this as you demonstrated? Thanks for the video as it was very insightful.
Thanks for that great explanation indeed! I used to work in Ableton Live before I switched to the much easier to use Acoustica Mixcraft 8 Pro Studio. This is a niche DAW aimed at musicians, not mixers. Well, I now totally understand what foldback distortion (or harmonics, what's in a name) means. I work in 48 kHz because of CPU constraints (the DAW enables 192 kHz) and bounce/render in 48 kHZ before delivering to clients. I have noted that bounces in .FLAC result in significantly less artefacts than .WAV, especially after I upgraded my flawless i7 Windows laptop into 16 GB of RAM. Any thoughts on this (no compression on the .FLAC)?
By the way: the machine is a quad core ASUS i7 at 2.4 GHz with a Native Instruments Komplete 6 audio interface that works flawlessly while running in the project
Really good point i had this problam with most daw that i used to compose. Okay we got fix that on digital daws.what about external hardware do they add same backword harmonix too ?
Are you saying that 44.1 and 192 are interchangeable? I thought the sample rate of 44.1 is the only one that can be used? I am using cakewalk by bandlab. Can this be relegated to this DAW or does this only work in the DAW you mention. I'm a little confused in general with this video but want to learn more about it all. Thank you so much for the great videos.
You'll find a lot of plug ins have built in oversampling, but, the issue is you then rely on the accuracy of their own SRC, so yep, best to work at 96k or above if you can. What he doesn't mention is not all SRC is equal, Izotopes is top notch in tests, ie, RX7. There's a website somewhere that tests all DAW similar to how he has here, can't recall the name but I'm sure you could find it. Btw, most of the plug ins have intelligent auto sampling, so if you run at 96k it will know and either disable or maybe reduce to x2 depending on the plug in.
![Megan Thee Stallion - Roc Steady (feat. Flo Milli) [Official Video]](http://i.ytimg.com/vi/HozPPyqW0dY/mqdefault.jpg)
In the Bounce to Disk settings of Pro Tools you can select an option called 'convert after bounce'. This makes sure you bounce in the session's sample rate before converting the result of this bounce to a lower sample rate.
Do you happen to know how to do this in Cubase?
Thomas Niehof Thanks!
What version of Pro Tools is that? I don't see it in 2020.5.0
Goated Comment
when an engineer says I was playing around in my daw...AND I found a few interesting things..................... You listen!
made my day!
And you really listen when said engineer has a non-American accent!!!
I listened, but clearly I don't know enough to pick up what he's putting down.
Paying close attention to fine details now may make a huge difference 20 years down the road if you believe in your music that much. Thank you
you're absolutely 100 percent right about the details. its the same with any art form, attention to detail (without losing overall perspective, of course) is the thing that really shines through with its cumulative effect.
It's also true for other engineering (physical/electrical etc), architecture etc. Details separate excellence from ordinary.
Well said!!
Ok people here is what I recommend as a professional studio engineer. As you record each of your tracks, add your effects, automation, etc. Then, right click the track if in Reaper and select Apply Track/Take FX As New Items. What this does is that it lowers your CPU use so that before you start the project you can pre set the project as 192 instead of 44.1 This turns your track into a .WAV. Remember though that if you do this and you want to edit your tract you can revert by turning the track to the following state called Active again. When you have done all your mixing and you want to render/bounce then just bounce your project as usual that will each be a .WAV and you'll have a great sound rendering it as 192. My favorite mastering plugin at a reasonable price recommendation would be the Abbey Roads Mastering Chain and select the preset called Balanced. Here is one instrumental where I played all the instruments and did what I recommended to you guys. P.S. the guy throughout the video is not me and you can see what my real name is. Cheers. Frank (New York City) ruclips.net/video/D6hZtFMx12o/видео.html
Thank you from the bottom of my heart! This is something that has bugged me forever. I'm just a dude in his bedroom playing around but I want the best quality that all my amazing PC can put out :-)
Very timely. I've been been thinking about moving to a higher sample rate but wanted to avoid the traps that can make it worse. This was helpful.
Never realized this before. Easily the best tip I have heard in a long time, thank you
Dude. You're one of the most informative RUclips producers. I learn a lot watching your channel. Thanks for the information.
My first finger-up for you, - hey man, you‘re able to describe important things, incredible : go on with this- your nervous style gives this sweet appearance of reliability :-)
"finger-up" shure sounds funny.
this is the topic where ify pondering and guessing youre alright will set you back years. youll get all invested in gear and work all day, ruining every mix by bouncing wrong. youll even stop checking the mixdowns, the repeated truama of having your vision destroyed will be too much to risk. you start rendering at -6db and hire a mastering engineer eventually, assuming you have no ears and never did.
Well, middle finger is still a finger 😂
It's like this I always say people never even finished quotes because they don't even understand too take everything with a grain of salt means if you like the taste season your food with it not just judge everything and be skeptical that's why I appreciate your videos man thank you
"foldback distortion" refers to a kind of waveshaping, not aliasing. check out the free vst Driveshaper for an example of how foldback distortion sounds.
Excellent video. Now show the results on a mix! show us the "sandy" texture, versus clean or minimal aliasing. That is something to behold.
you'll barely notice a difference on RUclips, cause you can export audio in a decent quality
Did you not get the point? Try it yourself if you absolutely need to verify if it works.
I am so happy you are doing videos that take on this subject. Solving these kinds of issues have been the keys to me finding a clean sound on a DAW. You have certainly aided in my understanding. Thank you.
Absolutely great to know this-- making the change in my settings now-- thanks!
Hey there, actually heard about that only a handful of producers tempt to organize their projects like that.
I was never certain if I should do it either, but this video gave me a clear answer. Thank you for that.
I love how you consider the human side of things and not just the technical details.
I like all tips keep them coming, except mushroom tips.
Only reasonable thing to do now is to make your next video a blind test with different audio samples at different sample rates to prove that it matters and actually sounds better. Perhaps you would not be able to tell which is which in some cases. Make examples for us as well. Play A/B parts bounced out at different sample rates, then play them without showing which is A and B, then we can hear for ourselves. I have had so many discussions with audio engineers that claim this and that, then we do blind tests and they make a fool out of themselves. Would make a fun video either way :)
RIchard Stylez just try tidal and go back and forth from normal and master modes. Easy to hear the difference
Yes, I agree and I was all was always taught if you can't hear it don't do it. Makes no difference to me anyway as I run on a core duo with 4gb ram, it only just copes with 48000.
@@WillJukedTheBox "Difference" doesn't mean better. 44.1 is fine, there's really no benefit to going higher
J W while i agree 44.1 is fine, there are benefits to going higher at least in the studio and they have to do with pushing aliasing artifacts out of the range of human hearing increasing the quality and authenticity of analog emulations and synth, saturation and distortion effects, also reducing latency,
In Pro Tools, I have always bounced to disk at 24/96 stereo, then brought the resulting file into a 24/44.1 pre-mastering session, letting it down-convert upon import using the best settings. I then pre-master my final to 16/44.1 with medium dithering.
Are you serious? I really can't tell.
@@dissdad8744 Are you a troll? I can tell.
your best to keep all files at the higher recorded rate until after mastering and then convert them for specific mediums like cd via good dithering software .Your approach is wrong totally as downsampling via import is both a substandard quality and also its pointless prior to mastering where the higher quality will allow better subjective evaluation of the music for mastering.You should keep the main master at working rate and secondary masters at lower rates off the back of the finished master.
@@cornwallradiophonic6250 You are absolutely correct. Nathan's approach is so weird and unprofessional that I was wondering whether he was a troll? By the way: use Voxengo's free R8brain for SRC - it's better than most DAWs SRC algos!
@@cornwallradiophonic6250 Those are good suggestions except for the understanding of the sample rate conversion on import, which uses exactly the same set of algorithms (for the various speed vs. quality settings) as if you converted when bouncing, at least in Pro Tools, which is what I have used for years.
Great video! Can you quickly explain me how to deal with this trouble in Fl Studio?
Hey thanks , I know this is an old video but it really helped me . I have been struggling to render a finished track and getting it to sound just the way I hear it in the mix . This works !
-cheers from Canada
Wow, thanks! Bringing up the project sample rate to 192 before bounce made it sound a lot better and/but showed in the sound overuse of the soothe plugin, because im taking away harsh sounds that only appear in lower sample rate that im mixing in...
its hard to argue with a waveform... great video !
www.image-line.com/fl-studio-learning/fl-studio-online-manual/html/app_audio.htm
Great video, I love the technical science side of audio and this is very useful information. Sometimes I see people post in forums about when they render a file it sounds different than when they were mixing it and I've always wondered what could be the cause of that and this could be one of the explanations in certain cases. Also what frequency analyzer are you using with all the pretty colors?
hey you might wanna check out this topic
www.image-line.com/fl-studio-learning/fl-studio-online-manual/html/app_audio.htm
Awesome thought provoking video. If only my CPU could keep up with realtime 32x oversampling all all plugs at once.
Is there was way to pull this trick off in Ableton?
Ableton 10 automatically renders in higher sample rate and then exports at whatever you choose. If you set it to high sample rate in project and then go to export it tells you that it will render at higher sample rate
Kclip Zero bounces at 8x even though it has no switch. But its weird because it sounds different then what you were working on. Which can be confusing for beginners. I believe Sytrus also does this in FL. Which drove me crazy when I was younger because i really like the aliasing sound for background dubstep effects. But for anyone who is thinking of buying flatline or knock or standard clip, I highly recommend FreeClip. It has 32x oversampling switch and beats out most clippers in sounding analog because of it. It also has 5 different softclip modes. The quintic setting sounds very proffesional when bounced at 32x. Other plugins might seem better because they have a percentage of how hard to clip, but nothing comes close to FreeClip. It just sounds so clean and analog. I stopped using Kclip because of it. Kclip is so good on paper but it sounds digital and almost exactly like Fruity Clipper but with more options. Some guy did a comparison and showed that FreeClip has the least digital artifacts out of all of them, even though it's free. I've never tested flatline myself, but it seems to be snakeoil. Maybe I'm wrong.
Brilliant as always! I would have had no idea that was happening if you hadn't pointed it out.
100%... Actually you're correctly thought out on everything I've watched of your channel. This video educated me on some questions I've had, great work!
Man, this was the best tip ever! So I have to condition myself to think of setting 192 kHz in the 'project settings' before rendering. While producing/mixing I would leave it at 44.1 because otherwise my RME (or let's say my CPU) can't handle this amount of processing well.
This kind of thing is what makes people record a bounce inside the DAW in real time. Sending Mix Bus to an Audio Track then press Record. Your recorded file is your bounce
Cool!
Thank you for another helpful video.
Please, what colorfull spectrum analyzer is that?
Love this video and your honest view.. follow you and watch every video now for the last 9 months. Keep up the good work!
It is great to know that it actually makes sense working at higher sample rates when you are processing and harmonics are newly produced (is this right?), and it is a bit weird to see that you need to ask Reaper to convert after exporting, I thought that was a given, Ardour and Mixbus are not as flexible as Reaper but this is not a conversation needed there. - Quick question: assuming you were working at 44.1Khz, how would you avoid these fold back harmonics, does it make sense adding a hard LPF on the master? Buses? Tracks? Nowhere at all? Sorry, this is not the most simple topic 😅
Foldback distortion is actually when the amp axis is folded back - harmonics passing Nyquist and folding is just called aliasing/alias distortion :)
In my opinion and from my experience there is no need to use plugins only in mixing ( it is all about balancing the tracking) .. once you hear everything in a perfect way , your job is done which leads to less is more.. in mastering yeah you need plugins that adds taste to the sound
Btw keep those very technical scientific videos that nobody gives a shit. Some of us, in minority, do care.
And those are usually the next Grammy winners ;-)
After reading one book about how Digital processing and how things are converted, it's become a primary concern . Right next to Acoustic Treatment and Ear training.
And these are often the subjects that are not covered. So yeah, much appreciated.
ZpurpZ what book
MIchael Wiggler I’m wondering too!
Cool, another video :) I love your Snake oil vids but these really set a balance and show that you really know your stuff. Very informative and interesting . I use reaper so this is useful for me, thanks.
HQ offline rendering is supported by some plugins, and several of the clipping plugins i use (standard clip, v-clip) offer internal oversampling to counter aliasing. Setting my interface to 192kHz uses a lot of cpu, so I’d rather stay at 44.1kHz and use oversampling where needed (in the plugins themself).
We need another in #depth Q&A with Chris on this one. (:
Yes!
@@Whiteseastudio No need. You got it right :)
@@applejinx7172 yay!!!!
This is great. Really useful. Thank you!
It would be cool if you made a series of process checks that helps illustrate different concepts like this video, with example files and different scenarios to help drive the concepts home.
Thank you so much for your info on aliasing. Being aware of this issue has had me second-guessing my use of saturation plugins to ensure I'm not muddying up my busses. Plus, this episode may help me to figure out why my mix downs, in Reaper, always come out sounding more muddy than the project. I assumed it was the down sampling, but more specifically, it could be the aliasing caused by these settings. Much appreciated. Keep up the good work.
Just replicated this, blew my mind.
Exactly how did you do it? Im using cubase.. Trying to follow
I think that the reaper test tone generator it is not band-limited. With "normal" VST instruments, you shouldn't see that enormous amount of aliasing. I tested it by initializing a Retrologue patch, setting OSC1 to triangle and generating a test tone of ~ 16.4Khz @ 44100hz sample rate: no aliasing. At the same time, I agree with you that with higher SR some virtual effects and instruments sound a lot better! Cheers :-)
Great info!, I feel that maybe is a lot to ask, but many of us that follow you could really benefit if you could do the same experiment of exporting and checking distortion in different DAWS. I feel many of us want to improve our sound, but in regards to sample rate conversion we are a bit lost; plus, the options for exporting are not the same in every DAW. Maybe if you can help the community you have built many of us would be forced (in a good way) to jump to another DAW to mix the project, like e.g producing in Ableton or FL and then exporting the session to Reaper or Logic or whatever has the exporting option that has the less artifacts. Thanks for all the information you are constantly delivering! Greets from Argentina
gushH¡ buy him the daw and he will check it for you
Do the test and find out?
@@MetiCudi You can export in several DAWS without having to buy DAW itself...
@@MetiCudi what is with a Trial Version? I would like to see how it works for example from Ableton to Reaper
google src infinitewave. they made a ton of comparisons of SRC in different daws and even video editing programs
I've seen a few of your videos, this one made me subscribe. I was not even aware that rendering at the project sample rate wasn't automatic. Keep it up!
Simple? Oh my I have so much to learn
I DEFINITELY a difference!!! Thank you oh so much!
Very nice and concise demonstration. Thanks a lot.
Hello! Big Fan! That happened when you bounce, how about when you print and then export the audio in a different sample rate? o when you print and then bounce without any plugin??? it would be nice if you can do a part 2 of this video showing what im saying! THX!!!
Curious to know people's thoughts on Bouncing to disk, or just printing to an audio track within the session, then exporting that newly made audio clip.
I can say with 100% confidence, that bouncing to disk will give you a slightly different sound to sending your aux mix bus to an audio track and recording/printing that in session. For those who are skeptical, try both of these methods, then open a new session with both of the audio files on seperate tracks, and then phase invert. In theory, the result should be silence, however it is not.
As for which method is better, that is subjective, however I have found to take a liking to the printing method as opposed to 'bouncing to disk'.
Great Video. Does this phenomenon also happen in Cubase and Studio one?
Man I actually really appreciate your content its been helping me with production for awhile
Very true. Great video!
I completely agree with - I want a bounce that is exactly the same as what I hear. The only way to get that, that works for me is to use my virtual channels on UAD Apollo (and I'm guessing you can do this with other I/Os that have more advanced mixers, RME comes to mind) and record back to DAW ( I use S1). After that I use Ozon standalone to covert it to whatever format I need...
Drasko Popovic that’s interesting. May I ask how do you export from Apollo’s virtual channels? Thank you!
@@dodo.danciu Hi bud! So, the concept is simple: You need to go from your DAW to UA console's virtual channels and then go back to the DAW by making a new track that has those channels as input (and mute them during recording). Most of the DAWs will automatically assign the available inputs and outputs (if not then add them yourself) and then you only need to link either 1 and 2 or 3 and 4 Virtual channels in UA console for them to work in stereo.
I hope I was clear enough :)
And this is a wonderful experience of how to solve aliasing problems. Great video!
Nice 👍 Why aren’t all daws set up like this by default if it yields a more accurate and clearer depiction ?
Thanks a lot, greetings from Utrecht!
This is the message I've been trying to get across to all my friends who are kind of on the fence about how the world is going complacency is the devil whether or not you believe in one or not it is the very thing that destroys Humanity diversity and detail is everything because we are all the same yet so unique! Wake up people!!! Be different, search for YOUR sound👊👊👊👊and keep doing what you doing man I appreciate your videos and what you're doing
Thank you Mr. White!
Great demonstration and useful. Thanks
I would love to replicate this with Cubase but I don't know how. You cannot change project sample rate without converting every file. Any helps will be much appreciated.
Great Video! i'll have to learn how to check and make sure i'm mixing down properly in Cubase.
Very good my friend, 100% right, thank YOU!!!
Love that technical videos, keep giving us more :P cheers from Mexico!
you just made my day with video..we are learning about this rn in my audio course and this rly sums why i love to put high sample rate for projects, and we render at 44.1! :D thanks!
I had to rewatch the part where you say the trick, this looks really good. Btw, what happens if you just literally put a brickwall filter at the end of the mixbuss chain and render like that? Would it eliminate aliasing distortion too?
One recommendation for high quality resampling: Use Audacity, it has SoX (one of the best resampling algorythms and for free), so the "resampling distortion" should be almost inexistent. So for this, render the mix from the DAW at its original sample rate, import into Audacity, set the propper quality from Audacity settings and the target sample rate, and export.
That's not what the video was about at all.....
I’ve found myself getting ridiculously obsessed with getting rid of aliasing recently (doing crazy things like running at 192 at 16 or 32x oversampling) but then I think.. often it’s only really at a problematic level when you’re really smashing it to destruction, and if you’re just using mild saturation or something, perhaps all that is really achieving is adding needless processor load. I dunno.
It depends, if your doing any higher frequency saturation then you should oversample but really 8x oversampling is enough in almost all cases, a couple instances of aliasing distortion in the odd track or two isn't really an issue it's when you start using tonnes of saturation on busses or your master that it's a problem because it's introducing inharmonic frequency content along with the innevitable intermodulation distortion that occurs with buss processing, if your subtle with it, it's barely an issue but it's certainly something to be aware of along with things like the nyquist shannon theroem and bit depth.
Is that possible in cubase too?
have u ever reviewed/explained clipping plugins??? thx mitch
Wow that is incredible. I had no idea that Reaper was doing that in the background. I'm going to bounce at my native resolution from now on and then do the conversion separately.
Yes! Great stuff. Now I have to check how this works in Pro Tools.
Thanx for your description. To have it crystal clear: Do I have to tick or untick this option "Use project sample rate for mixing..." ? Any help will be appreciated.
Awesome as usual. Btw what are tech specs of your pc/mac to run big projects on 192khz ?
Alex Faizulin for real. I suspect he does most of his processing with analog gear though to avoid setting his cpu on fire
@@WillJukedTheBox definetely analog helps ) But it's still interesing what setup does he have.
@@AlexeyFayzulin has to be a maxed out trash can mac pro
Thank god you didn't assume everyone knew this, no wonder I always thought my down mixes didn't sound the way within the DAW. "but" now questions arise: If I am working in a 192 khz environment, should I set plugins that oversample to that oversampling mode? Or that plugin will work as it should if I am in that 192 khz environment? No wonder I wasn't liking Reaper, if only I knew this tip sooner (I switched to Studio one pro and yes, I love it). Those back harmonics/distortion is so irresponsible from developers... sigh...
Wow. This is actually really interesting. I am working in 48k for Processing power... I think i will try this next time and Render at 192 k. Should work If i understood correct.
any tips on this for Pro Tools? Thanks for making these videos
ummm... As a mixer with an older computer, I'd rather be able to use more plugins and more tracks. I have to do tons of bouncing, splitting projects into separate pieces, disable all kinds of stuff as it is just so I can keep working and I'm thrilled when I can even barely finish a project on 44. It feels like I performed a slight miracle every time and I deserve a medal or something. Given the choice, I'd rather run a high quality convolution reverb (which I can't do) then run at a higher sample rate.
I use 48kHz as the project sample rate and also bounce the tracks in 48kHz. This is a really interesting topic. I will definitely try out sampling down from 192...
Nice one, things like these get overseen by many !!
Would a really simple way of doing this on your final mix not just be to bounce out the entire thing in 192 and then open it up in your DAW again and bounce it down to 44.1Hz? So all the plugin processing etc will be done on your first bounce. The downsample conversion will be much cleaner?
NOTE: Aliasing only affects non linear processing. There's no point in doing this for linear processes like EQ (unless decramping near Nyquist is the goal). Oversampling smears small transients in the time domain, so shouldn't be used unless pros outweigh cons. Here's a tip - if you want clean audio, don't put "saturators" everywhere. "Saturation" is a recent sales gimmick used to make people buy more stuff they don't actually need. It has a valid spot in the workflow, but even with mitigation techniques, over-use of digital saturation is a recipe for bad sound , not a panacea for digital vs analogue folklore.
DigitalDiggo 100% with you regarding saturation plugins. I run a hybrid studio that tends to record live instruments. I’ll run some saturation here and there on a channel to help a single note guitar or snare or whatever cut through a mix, but I can’t imagine it on a bus. But, having said that, no rules baby! If it sounds good it is good. So many of these very helpful videos are always gear, gear, and more gear. You need quality equipment but the key to a good mix is a good arrangement.
As a DSP scientist, everything you said is correct. The time domain will get smeared as a result of oversampling. But you missed one small part of this: the time domain ONLY gets smeared as a result of calculations of DA/AD converters. Essentially, the result of bouncing a high project sample rate, and then converting back to 44.1-48khz will indeed result in a cleaner signal after exporting. You can even go from 44.1-48khz to 192khz, bounce, then downsample back to again. The affects of increasing a project sample rate are null. Also, what you say about saturation have never been a gimmick so this is misleading. Digital distortion can sound just as good as analog distortion if not better these days. You just need be careful with oversampling. The higher the internal oversampling, (especially linear phase), will cause continous smearing of transients because of the affects on time domain.
@@shinyukomusic Small world lol! Not suprised to see you reacting to this comment but the fact I happen to check this vid for the first time and then see you responded to this comment one day ago… what are the odds? 😂
Incredibly helpful information. Thanks!
Some Voxengo Plugins can be set to use maximum oversampling in bounce mode. I don't know how well this works, but since I got plenty of CPU I usually set these to permanent 4x oversampling anyways. Which results in 196khz processing inside these plugins since I usually work in 48khz in Mixbus.
You could also bounce or mix down at the session sample rate and then downsample during mastering.
If I’m recording, mixing, and bouncing my tracks in 44.1 will this fold back distortion apply since everything is kept at the same sample rate? Or does this still apply? If so will filtering everything above 20k fix this as you demonstrated? Thanks for the video as it was very insightful.
www.image-line.com/fl-studio-learning/fl-studio-online-manual/html/app_audio.htm
Have you ever tried Focusrite LiquidMix ? imho it sounded extremely close to real hardware, better than any plugin
Thanks for that great explanation indeed! I used to work in Ableton Live before I switched to the much easier to use Acoustica Mixcraft 8 Pro Studio. This is a niche DAW aimed at musicians, not mixers. Well, I now totally understand what foldback distortion (or harmonics, what's in a name) means. I work in 48 kHz because of CPU constraints (the DAW enables 192 kHz) and bounce/render in 48 kHZ before delivering to clients. I have noted that bounces in .FLAC result in significantly less artefacts than .WAV, especially after I upgraded my flawless i7 Windows laptop into 16 GB of RAM. Any thoughts on this (no compression on the .FLAC)?
By the way: the machine is a quad core ASUS i7 at 2.4 GHz with a Native Instruments Komplete 6 audio interface that works flawlessly while running in the project
excelente. lo tuve que ver varias veces para entender y menos mal que tu ingles es nitido.
Great!! This explains the horrible sound of bounces in ProTools. Thank you!!
How does this explain that? Pro Tools applies SRC AFTER processing!
Hmm... no ones complaining about my horrible PT bounces.
Very informative, but very interesting as well to see how much is actually audible to people listening to your song, if at all!
Oh God...Really?
Really good point i had this problam with most daw that i used to compose. Okay we got fix that on digital daws.what about external hardware do they add same backword harmonix too ?
You should do a review of IK T-Rack's Tape Machines, which operate at like 384kHz internally to prevent aliasing.
Yes please.
So if I have recording as 32/44.1 I should upsample it for editing? Will definitely try that in my next mix I now have to make!!
Very good.
This stuff is what I like to learn about.
Are you saying that 44.1 and 192 are interchangeable? I thought the sample rate of 44.1 is the only one that can be used? I am using cakewalk by bandlab. Can this be relegated to this DAW or does this only work in the DAW you mention. I'm a little confused in general with this video but want to learn more about it all. Thank you so much for the great videos.
"Not the sandstorm, the sand feel" xD
Does this apply to Edison from FL studio?
You'll find a lot of plug ins have built in oversampling, but, the issue is you then rely on the accuracy of their own SRC, so yep, best to work at 96k or above if you can. What he doesn't mention is not all SRC is equal, Izotopes is top notch in tests, ie, RX7. There's a website somewhere that tests all DAW similar to how he has here, can't recall the name but I'm sure you could find it.
Btw, most of the plug ins have intelligent auto sampling, so if you run at 96k it will know and either disable or maybe reduce to x2 depending on the plug in.
If I already started a project at 44.1, is it too late for me change it?
Would changing it do harm to my project?