That will be a short video: anyone below a experienced pro level probably shouldn’t worry too much, the biggest problem of your mixes most likely isn’t aliasing. The end ;)
Haha! Another great video David! thanks so much for clarifying this. as always there's little gems i didn't know about things every time you explain something. thank you so much for striving to make things as clear as possible and dispelling myths. you're always a great source of accurate information.
he has great points. however his test at the end did not show him using distortion and other plugins that would cause foldback. in other words all he did was up-sample and dither with no processing. The aliasing only occurs when using low sample rates with a subar plugin not optimized well. nontheless if in 44.1 one should oversample the distortion plugins. if it has no option to oversample, then use a plugin like metaplugin.
Which is not the subject of the video. And oversampling is not a free ride either and there are other ways to avoid aliasing, which is not an issue in 2021
David, one thing should be clarified. Yes, when you're printing tracks with outboard gear, you want highest sample rate possible. But ... To do tha when you're running 44.1 files through the gear you need to capture them into the second DAW (with higher sample rate set). Otherwise your DAW automatically downsample new tracks back into 44.1 to fit the session. So... When you have 44.1k tracks and run 48k session for example (and want to capture your tracks back with 48 into the same DAW), you need to upsample all tracks to 48k b fore even starting the session (even that upsampling doesn't add anything new, just plain zeros). Am I right?
What if I am only ITB and want to record/export/bounce a VST synthesizer with no plugins on the insert? I believe since no ADC involved in the internal exporting and it is happening at 32 bit floating point; I can and I should record individual tracks as loud as possible with the maximum possible dynamic range. For example reverbs and delays could benefit from being recorded hotter ITB if we want to preserve couple of extra bits in them (correct me if I am wrong). Thank you!
Thank you for this video! I actually just watched the 1st video before this one and replied to someone’s CLA comment and I mentioned aliasing Haha! I too have been a victim of the aliasing myth going around the internet. It had me very interested for about a week and I read tons and tons of info about it.... and it got really frustrating. So many different opinions. Lol Then I read an article about the producer for the Weeknd track “The Hills” and how he used the decapitator plug in all over that mix. It has over a billion streams! No one seemed to care about aliasing with that track haha. So I’m thinking no one really cares at the end of the day. The only thing I can think of is that maybe the soul can hear aliasing lol. I have no clue! 😭 The song is really what matters most and that it’s captured in a way that sounds “good”. That aliasing shit really had me messed up for a minute! So grateful for your video and your channel! You got a no bullshit kinda vibe! No more fuckarounditis for me! 😆
About the upsampling within plugins. On most of them I don't even notice if it's on or off tbh. The only exception for me so far is JST Clip. It's like a million times better with upsampling turned on and I immediately hear when I forgot to turn it on.
Just a guess, but I think the spike at 22kHz in your analyzer comes from ringing-artefacts due to a very steep low pass filter applied to the signal (a.k.a. Gibbs phenomenon).
Thanks David. Good job. One comment though (or maybe I misunderstand you), you keep saying that extra sample resolution "adds zeros" to upsampled data. That isn't actually what happens. If zeros were inserted, for example when going form 44.1 to 88.2, then the data would drop to zero between every two sample values along a waveform, which would end up being noise. Depending on the specific conversion process, it will probably interpolate the sample values to fall in place at the new rate. Of course, if you are saying that it 'adds zero information', then I misunderstood, as that is correct. :)
Great Channel! Watch a lot of your content. that said I can't help but wonder with the analog setup you have in the background, that MOTU only has eight in and outs, just the analog gear in your Wes Audio Super Warrior alone takes up all your I/O. Are you using the AVB somewhere else? because it doesn't make sense that you would limit yourself to 8 I/O from the MOTU in a professional setup where time is money. Are you Working mostly ITB? I understand you are setting up a new studio but your background rack always resembles that, other then your really old videos when you had the SSL and patch bays.
Hi David, I know this is off topic, but I was hoping you could possibly do a comparison in a future video. I'm interested in buying an analog tape machine, but I also love the sound of those Neve 542 tape emulators. I'm really interested to see if the Neve's really have that tape sound or if they are best used for saturation and not replacement of tape. Do you possibly have access to a tape machine to do a comparison between the two?
David, can you clarify how to manage "oversampling recapture", for example, an analog 2Bus chain when returnig back in the daw? I mean: Mix in the box, i/o Plugin to go out in my analog chain and than back again in the daw, bounce? Any suggestion for the right way to do it? Thank you!
if I mix something in a very high sample rate to avoid aliasing, then I render those tracks out at that high sample rate, and then bring those high sample red, renders back into a lower sample rate session to convert to a lower sample rate, in an effort to avoid aliasing from plug-ins, does the act of downsampling a mix in and of itself, create aliasing?
No but you're WAY overthinking it, let alone this would be an absolute nightmare workflow, what do you do this for every single processing for every single track? And most important it's useless, use oversampling on the plugins that do have it, and that's it. Not all plugins need oversampling because not all plugins cause aliasing, not all aliasing can be solved with oversampling, not all aliasing is even audible or has significant impact on a mix. Downsampling does not cause aliasing anyway
David, can you clarify one thing please? I do use analogue outboard gear for both mixing and mastering. You say that it`s a good idea to recapture the signal after the gear at the highest possible sample rate. Let`s say I have all tracks at 44.1k and use analogue gear. Wouldn't it make better sense to upsample those tracks to 48 before running analogue gear so they are already 48k? Thanks a lot.
Plug-ins do operate better and sound better at 96kHz. I mix in 24-bit 44.1kHz, so I decided to test this theory (using hardware at 96kHz is much better than 44.1kHz.). Yes, 96kHz was significantly better, even after importing both files into a blank 24-bit 44.1kHz. They were both processed using an external hardware reverb unit, in bypass mode - NO plug-in processing. Eventhough 96k sounds better to me, I don't have the CPU to use plug-ins in those types of sessions, so I'll cop out to using 32-bit 44.1k. I assume that I'll hear a significant difference! But anyways, Peace and happy mixing!
@@mixbustv -- Thank you for your reply. I did test 32-bit vs 24-bit @ 44.1kHz (through hardware) and they did not null. I wouldn't say that it made NO DIFFERENCE at all. There's clearly something audibly better, at those settings. I also decided to try 192kHz, just out of interest and reverbs were dramatically more detailed and in the background. The mix sounded more finished. I used to mix at 48kHz for a long time, but I'm unsure whether going back is necessary, at the moment. Anyways, great video, great discussion! Peace.
Hi David, a question I always had about oversampling: is it necessary to use the internal oversampling of each plugin separately? Or for example, just using oversampling in the Fabfilter L-2 placed at the end of a master is enough to stop the aliasing that the converter will cause? this doubt really drives me crazy haha
No is not necessary, and in the particular case of brickwall limiters, oversampling can change the sound and it doesn't necessarily sound better. For distortion/saturation plugins, usually oversampling makes a big difference, for the better.
So, at 9:14 you talk about going out of your converter and back in, and how that is detrimental to audio. I was taught this, for sure. But I watched this video, with Eric Valentine who I admire a lot, and well, he runs the signal out once, then 4 times, then 20 times, and you can't hear a difference. I know that logically, audio should degrade, I'm, curious what you think. I'll test it all myself anyhow.
@@mixbustv yeah, for sure, I understand all of your points. And I know the video is to clear up stuff that a lot of people are confused about. Not the case here. But the interesting thing is, if you watch Eric's video on "in the box vs out of the box" he does it as a test, not running through any gear. And even after 20 successive passes, the degradation is negligible. Check his video if you can.
Hey bro ,,if you run spidif out from a converter to a converter to a better converter , does the better converter and make it better..at the same sample rate..
White Sea Studios (mostly known as the Snake oil guy) is a RUclipsr who does plugin reviews. He's usually pretty negative on analog emulations and I used to really dislike his video's but the more I watched it, the more I was agreeing with him. (I still don't agree about his views on the interfaces and stuff.) then he made a video on alliassing. And I think he did very well. He explained well and even proved what he was saying. He later came back about it on his 192khz video. And 10 years ago I was 12. I got into audio engineering when I was 19 so I wasn' t there during the discussion. And right now I am agreeing with him. Especially since I've recently played around with analog gear and heard that it sounds better than most emulations. Even money is starting to become less of an issue when picking between the two since there is a website called Mix:analog that allows you to use analog gear for set amount of time while being very affordable. Yes there are still limitations but their team is also trying to solve those as well. But I am very open to be convinced otherwise. Especially since I like to be informed about these kinda of things.
Lol this is a "cleanup" video. He basically made a shallow video to begin with and we took it to another level in the comments.. He doesnt realize the we changed the subject.. He thinks we're saying he's "WRONG" .. Its because the original video was shallow. ALLLLL this info being covered right now in this video, should've been in the original. And i noticed he skipped getting into aliasing.
10:40 Hi David, do you think that when tracking a project that will be mixed with outboard gear would be better to record at 96 khz or higher right from the start? Thanks!
@@mixbustv I wasn't being ironic, although I kinda wish I was. It doesn't even make any sense thinking about it. Samplerate is the clock speed, not part of processing, let alone causing rounding errors. The worst part is I should've known better as a plugin designer hobbyist, as I somehow picked up this old myth only just recently. I was searching why my current project sounds much cleaner at 48kHz. I now suspect it must be the internal filters that move up, or the resonance bump cramps less early and sounds therefore cleaner. I suspect that has a lot to do with this precieved clarify at 48kHz. Like you predicted it's probably just due to the higher nyguist.
Do you perhaps have any videos for re-amping? I'm not sure how to play the audio through the guitar amp from the sound card without it creating an extra noise. A looping pedal would work but not able to get my looping software to work. Let me know
I suggest getting a re-amp specific box(radial makes good ones). Signal flow will be Soundcard>reamp>amp>preamp/DI>soundcard(or just a Hi-Z input on your soundcard if it has one). It’s possible some extra noise will be added due to the unbalanced cables to and from the amp, but that can be taken care of with transformers from preamp, or an external box if it’s really an issue.
Depending on your needs you could just buy an IK Multimedia Axe I/O interface... It has a built in 1/4" output specifically for re-amping. It's the easiest way to re-amp that I know of.
I find this talk about the sampling rates being a potential threat for less experienced mixers to become less focused on their skill of mixing itself. It is much better to mix at 44.1 paying more attention to your balancing, eq/compression/depth creation /panning/ fx skills than caring about the sample rates. Focus on your mixing skills and mix at the sample rates you got the session in. If you are the producer starting recording the song from scratch then use 96khz if your computer is powerful enough and don't think about it anymore. Just my 2 cents
Mixing at 44.1 even if you're give files at that sample rates makes little sense, *especially* if you're using outboard. 48 has a big impact and improvement over 44.1 with negligible CPU hit. Is this simple concept a threat? Really? If that threatens a brain then mixing is definitely not for them. Recording at 96? For what? 48 is more than enough, 88 if you want to really go overboard, 96 is silly.
@@mixbustv This would be true if we lived in a perfect audio world. Unfortunately, we do not. Not all converters are created equal, especially if someone is using the cheap ones. Almost most of them sound much better @96k over 48k. I personally use both Apogee Symphony 16 for OTB mixing and after summing with my Dangerous 2 Bus, I record it back through my Universal Audio 2192. I tried recording at every sample rate and at least on my converters and to my ears nothing beats 96k. When it comes to upconverting the session from 44.1 to 48 for mixing as you suggest, unless you use high-quality conversion software, you will destroy the quality of audio for sure. Not too many people own this kind of software because it's expensive. I'm not sure about you but I am also a producer and I'm using a bunch of software synths, which to my ear sound radically better at 96k. That's why my advice is to always stick to 96k (or possibly 88k if you hear no difference) throughout the whole process. As a summary let me paste the official opinion of mixers like Bob Clearmountain, Joe Chiccarell, George Massenburg. Even though they speak about surround sound mixing here I do not see how surround sound audio would differ sample rate-wise from stereo audio. They state: "The mixing engineer may assume three things. First, the highest practical rate should be chosen for original recording and mastering. Second, all known resampling (sample-rate conversion) techniques are to some degree audible, although recent work has shown that with great care and provided with considerable DSP resources, one may resample, both up and down, very effectively. And third, an integer ratio conversion (i.e., 96kHz to 48kHz, or 88.2kHz to 44.1kHz) is more likely to have fewer resampling artifacts. It must always be understood that when upsampling audio (converting to a higher sample rate, i.e., from 48kHz to 96kHz, or 44.1kHz to 96kHz), one will never restore the upper octave filtered out at the original A/D conversion (i.e., when one converts at 44.1kHz, there will be little signal above 20kHz from then on). However, there are possible benefits to be gained from upsampling when playback occurs. In any event, sample rate is not the sole, nor even the most significant factor in perceived audio quality; the design and quality of the A/D and D/A converters makes the most important sonic difference. A state-of-the-art audiophile 44.1kHz / 16-bit converter can easily sound better than a 192kHz / 24-bit converter built using an inferior chipset. "
@@chrislysiak9561 Sorry but most of your information is pretty wrong, I don't care who you quote. I don't feel like going into this here, I'll just leave it at this: upsampling is NOT POSSIBLE. Period. A file recorded at 44.1 will always be at 44.1 you just add empty numbers (bigger file) if you "upsample" so no matter what software you use, is NOT audible because is doesn't do ANYTHING but making the file bigger. We work with lower SR file at a higher SR project because plugins work better at 48, not converters, converters have the same chip, the same specs the same everything at any sample rate, what changes is the quality of the *recorded* file. As for how much difference and to what point, again, there is plenty of literature showing drawbacks at 192 which would be such minimal difference from the one happening between the bare minimum, 44.1 and 48. With that said, whatever rocks your boat, just don't post incorrect info here. Upsampling is NOT possible.
@@mixbustv You're attacking the strawman you created. Both me and the other sound engineers I quoted say pretty much the same thing. Upsampling adds zeros. That's it. It was me that said we should mix at the sample rate you receive the session in, to which you stated that we should never mix at 44.1 but we should do it at 48khz. THIS IS UPSAMPLING. You are advocating upsampling, not me. I only said that if you have a choice to record a project at 48 or 96 I suggest going 96 simply because many converters sound better at that rate, especially the cheap ones. If you don't agree with this, then we have to agree to disagree. When it comes to the sound of the plugins, it all depends on the plugin. Some indeed sound better at 48 while virtual synths almost always sound better at 96k. I yet have to find one that sounds better at 48. Anyway, I do not wish to argue here. I just think that topics like that confuse people more than help them. With this comment, I will finish this topic. Peace!
![Natanael Cano - Amor Eterno [Official Video]](http://i.ytimg.com/vi/EO4prCs6WpM/mqdefault.jpg)
#1 vote for Aliasing-video! :D
Pls! There is a lot of misinformation on the subject
That will be a short video: anyone below a experienced pro level probably shouldn’t worry too much, the biggest problem of your mixes most likely isn’t aliasing. The end ;)
Agreed. Would love to see an aliasing video. Waves plugins for example seem to have a lot of aliasing. Should I care??
Haha! Another great video David! thanks so much for clarifying this. as always there's little gems i didn't know about things every time you explain something. thank you so much for striving to make things as clear as possible and dispelling myths. you're always a great source of accurate information.
Absolutely essential information here. Thanks David!
love the no bullshit approach, strait to the point every time, i learn something new every video thank you
Great video. Can you expand on the plugins that work better at higher sample rates please? Some examples?
I'm glad he's dealing with all this misinformation that's all over the place
Preach brother! All the way around another great video. You always speak from a knowledgable place.
Thanks again for such wonderful info David. I really appreciate all that you do.
Great David! Thanks!
he has great points. however his test at the end did not show him using distortion and other plugins that would cause foldback. in other words all he did was up-sample and dither with no processing. The aliasing only occurs when using low sample rates with a subar plugin not optimized well. nontheless if in 44.1 one should oversample the distortion plugins. if it has no option to oversample, then use a plugin like metaplugin.
Which is not the subject of the video. And oversampling is not a free ride either and there are other ways to avoid aliasing, which is not an issue in 2021
@@mixbustv such as a supersonic low pass? Curious on your “other ways” 🙂
David, one thing should be clarified. Yes, when you're printing tracks with outboard gear, you want highest sample rate possible. But ... To do tha when you're running 44.1 files through the gear you need to capture them into the second DAW (with higher sample rate set). Otherwise your DAW automatically downsample new tracks back into 44.1 to fit the session. So... When you have 44.1k tracks and run 48k session for example (and want to capture your tracks back with 48 into the same DAW), you need to upsample all tracks to 48k b fore even starting the session (even that upsampling doesn't add anything new, just plain zeros). Am I right?
What if I am only ITB and want to record/export/bounce a VST synthesizer with no plugins on the insert?
I believe since no ADC involved in the internal exporting and it is happening at 32 bit floating point; I can and I should record individual tracks as loud as possible with the maximum possible dynamic range.
For example reverbs and delays could benefit from being recorded hotter ITB if we want to preserve couple of extra bits in them (correct me if I am wrong). Thank you!
Thank you for this video! I actually just watched the 1st video before this one and replied to someone’s CLA comment and I mentioned aliasing Haha! I too have been a victim of the aliasing myth going around the internet. It had me very interested for about a week and I read tons and tons of info about it.... and it got really frustrating. So many different opinions. Lol
Then I read an article about the producer for the Weeknd track “The Hills” and how he used the decapitator plug in all over that mix. It has over a billion streams! No one seemed to care about aliasing with that track haha. So I’m thinking no one really cares at the end of the day.
The only thing I can think of is that maybe the soul can hear aliasing lol. I have no clue! 😭 The song is really what matters most and that it’s captured in a way that sounds “good”. That aliasing shit really had me messed up for a minute!
So grateful for your video and your channel! You got a no bullshit kinda vibe! No more fuckarounditis for me! 😆
But you can see the aliasing in plugindoctor. That's why I stopped using SPL Iron because some guys on Gearslutz told me to do so.
About the upsampling within plugins. On most of them I don't even notice if it's on or off tbh. The only exception for me so far is JST Clip. It's like a million times better with upsampling turned on and I immediately hear when I forgot to turn it on.
Just a guess, but I think the spike at 22kHz in your analyzer comes from ringing-artefacts due to a very steep low pass filter applied to the signal (a.k.a. Gibbs phenomenon).
Maybe
Appreciate your good content
Thanks David. Good job.
One comment though (or maybe I misunderstand you), you keep saying that extra sample resolution "adds zeros" to upsampled data. That isn't actually what happens. If zeros were inserted, for example when going form 44.1 to 88.2, then the data would drop to zero between every two sample values along a waveform, which would end up being noise. Depending on the specific conversion process, it will probably interpolate the sample values to fall in place at the new rate.
Of course, if you are saying that it 'adds zero information', then I misunderstood, as that is correct. :)
@@mixbustv Good stuff. :)
This is pretty funny, but the interpolation process is exactly zero stuffing, followed by filtering. So essentially zeros are added.
Great Channel! Watch a lot of your content. that said I can't help but wonder with the analog setup you have in the background, that MOTU only has eight in and outs, just the analog gear in your Wes Audio Super Warrior alone takes up all your I/O. Are you using the AVB somewhere else? because it doesn't make sense that you would limit yourself to 8 I/O from the MOTU in a professional setup where time is money.
Are you Working mostly ITB? I understand you are setting up a new studio but your background rack always resembles that, other then your really old videos when you had the SSL and patch bays.
Thank you David.
Hi David, I know this is off topic, but I was hoping you could possibly do a comparison in a future video. I'm interested in buying an analog tape machine, but I also love the sound of those Neve 542 tape emulators. I'm really interested to see if the Neve's really have that tape sound or if they are best used for saturation and not replacement of tape. Do you possibly have access to a tape machine to do a comparison between the two?
Well said!
14:30 still I am curious what caused that difference. Thanx for the video!
looking great. Not you body i mean your gear behind you. ;-)
what is oversampling? i understand aliasing, i understand you can't upsample, but what is oversampling?
David, can you clarify how to manage "oversampling recapture", for example, an analog 2Bus chain when returnig back in the daw? I mean: Mix in the box, i/o Plugin to go out in my analog chain and than back again in the daw, bounce? Any suggestion for the right way to do it? Thank you!
if I mix something in a very high sample rate to avoid aliasing, then I render those tracks out at that high sample rate, and then bring those high sample red, renders back into a lower sample rate session to convert to a lower sample rate, in an effort to avoid aliasing from plug-ins, does the act of downsampling a mix in and of itself, create aliasing?
No but you're WAY overthinking it, let alone this would be an absolute nightmare workflow, what do you do this for every single processing for every single track? And most important it's useless, use oversampling on the plugins that do have it, and that's it. Not all plugins need oversampling because not all plugins cause aliasing, not all aliasing can be solved with oversampling, not all aliasing is even audible or has significant impact on a mix. Downsampling does not cause aliasing anyway
wow...so converting an 128 Mp3 in Wav doesn't make it sound better? holy crap! :)
....... so I watched your video and now I feel really really stoned
David, can you clarify one thing please? I do use analogue outboard gear for both mixing and mastering. You say that it`s a good idea to recapture the signal after the gear at the highest possible sample rate. Let`s say I have all tracks at 44.1k and use analogue gear. Wouldn't it make better sense to upsample those tracks to 48 before running analogue gear so they are already 48k? Thanks a lot.
Yes you should mix at 48 at least. But why do you have all tracks at 44 to begin with?
@@mixbustv when I receive tracks from clients they are often 44.1k.
Good info bro
Plug-ins do operate better and sound better at 96kHz. I mix in 24-bit 44.1kHz, so I decided to test this theory (using hardware at 96kHz is much better than 44.1kHz.).
Yes, 96kHz was significantly better, even after importing both files into a blank 24-bit 44.1kHz. They were both processed using an external hardware reverb unit, in bypass mode - NO plug-in processing. Eventhough 96k sounds better to me, I don't have the CPU to use plug-ins in those types of sessions, so I'll cop out to using 32-bit 44.1k. I assume that I'll hear a significant difference! But anyways,
Peace and happy mixing!
@@mixbustv -- Thank you for your reply. I did test 32-bit vs 24-bit @ 44.1kHz (through hardware) and they did not null. I wouldn't say that it made NO DIFFERENCE at all. There's clearly something audibly better, at those settings. I also decided to try 192kHz, just out of interest and reverbs were dramatically more detailed and in the background. The mix sounded more finished. I used to mix at 48kHz for a long time, but I'm unsure whether going back is necessary, at the moment. Anyways, great video, great discussion! Peace.
Hi David, a question I always had about oversampling: is it necessary to use the internal oversampling of each plugin separately? Or for example, just using oversampling in the Fabfilter L-2 placed at the end of a master is enough to stop the aliasing that the converter will cause? this doubt really drives me crazy haha
No is not necessary, and in the particular case of brickwall limiters, oversampling can change the sound and it doesn't necessarily sound better. For distortion/saturation plugins, usually oversampling makes a big difference, for the better.
Good vidéo. So wath are your parameters? 24 /48 or 32 flot/48? thanks.
Can you do a video of a successful “ upsample” -
So, at 9:14 you talk about going out of your converter and back in, and how that is detrimental to audio. I was taught this, for sure. But I watched this video, with Eric Valentine who I admire a lot, and well, he runs the signal out once, then 4 times, then 20 times, and you can't hear a difference. I know that logically, audio should degrade, I'm, curious what you think. I'll test it all myself anyhow.
@@mixbustv yeah, for sure, I understand all of your points. And I know the video is to clear up stuff that a lot of people are confused about. Not the case here. But the interesting thing is, if you watch Eric's video on "in the box vs out of the box" he does it as a test, not running through any gear. And even after 20 successive passes, the degradation is negligible. Check his video if you can.
Exactly - that’s why summing boxes are getting so popular
Hey bro ,,if you run spidif out from a converter to a converter to a better converter , does the better converter and make it better..at the same sample rate..
No. Digital is digital, nothing changes
Have you seen White Sea's video about alliassing?
White Sea Studios (mostly known as the Snake oil guy) is a RUclipsr who does plugin reviews. He's usually pretty negative on analog emulations and I used to really dislike his video's but the more I watched it, the more I was agreeing with him. (I still don't agree about his views on the interfaces and stuff.) then he made a video on alliassing. And I think he did very well. He explained well and even proved what he was saying. He later came back about it on his 192khz video.
And 10 years ago I was 12. I got into audio engineering when I was 19 so I wasn' t there during the discussion. And right now I am agreeing with him. Especially since I've recently played around with analog gear and heard that it sounds better than most emulations.
Even money is starting to become less of an issue when picking between the two since there is a website called Mix:analog that allows you to use analog gear for set amount of time while being very affordable. Yes there are still limitations but their team is also trying to solve those as well.
But I am very open to be convinced otherwise. Especially since I like to be informed about these kinda of things.
Lol this is a "cleanup" video. He basically made a shallow video to begin with and we took it to another level in the comments.. He doesnt realize the we changed the subject.. He thinks we're saying he's "WRONG" .. Its because the original video was shallow. ALLLLL this info being covered right now in this video, should've been in the original. And i noticed he skipped getting into aliasing.
just Record in your Daw in 24b/192kh or if you can 32b/192kh.
Useless at best
10:40 Hi David, do you think that when tracking a project that will be mixed with outboard gear would be better to record at 96 khz or higher right from the start?
Thanks!
@@mixbustv Thanks! Is there a reason why a 3.9 khz "jump" (from 44.1 to 48) makes a bigger difference than a 40.2 "jump" from 48 to 88.2 khz?
@@mixbustv Ok, thanks! Looking forward for new videos
48kHz is cleaner in processing. 1 byte is 8 bit's, 48000 divided by 8 gives nice round numbers, less rounding errors.
@@mixbustv I wasn't being ironic, although I kinda wish I was. It doesn't even make any sense thinking about it. Samplerate is the clock speed, not part of processing, let alone causing rounding errors.
The worst part is I should've known better as a plugin designer hobbyist, as I somehow picked up this old myth only just recently. I was searching why my current project sounds much cleaner at 48kHz. I now suspect it must be the internal filters that move up, or the resonance bump cramps less early and sounds therefore cleaner. I suspect that has a lot to do with this precieved clarify at 48kHz. Like you predicted it's probably just due to the higher nyguist.
Do you perhaps have any videos for re-amping? I'm not sure how to play the audio through the guitar amp from the sound card without it creating an extra noise. A looping pedal would work but not able to get my looping software to work. Let me know
I suggest getting a re-amp specific box(radial makes good ones). Signal flow will be Soundcard>reamp>amp>preamp/DI>soundcard(or just a Hi-Z input on your soundcard if it has one). It’s possible some extra noise will be added due to the unbalanced cables to and from the amp, but that can be taken care of with transformers from preamp, or an external box if it’s really an issue.
Depending on your needs you could just buy an IK Multimedia Axe I/O interface... It has a built in 1/4" output specifically for re-amping. It's the easiest way to re-amp that I know of.
@@mixbustv Shot a lot sire :)
@@MrSNEAKFREAK96 Shot man
I find this talk about the sampling rates being a potential threat for less experienced mixers to become less focused on their skill of mixing itself. It is much better to mix at 44.1 paying more attention to your balancing, eq/compression/depth creation /panning/ fx skills than caring about the sample rates. Focus on your mixing skills and mix at the sample rates you got the session in. If you are the producer starting recording the song from scratch then use 96khz if your computer is powerful enough and don't think about it anymore. Just my 2 cents
Mixing at 44.1 even if you're give files at that sample rates makes little sense, *especially* if you're using outboard. 48 has a big impact and improvement over 44.1 with negligible CPU hit. Is this simple concept a threat? Really? If that threatens a brain then mixing is definitely not for them. Recording at 96? For what? 48 is more than enough, 88 if you want to really go overboard, 96 is silly.
@@mixbustv This would be true if we lived in a perfect audio world. Unfortunately, we do not. Not all converters are created equal, especially if someone is using the cheap ones. Almost most of them sound much better @96k over 48k. I personally use both Apogee Symphony 16 for OTB mixing and after summing with my Dangerous 2 Bus, I record it back through my Universal Audio 2192. I tried recording at every sample rate and at least on my converters and to my ears nothing beats 96k. When it comes to upconverting the session from 44.1 to 48 for mixing as you suggest, unless you use high-quality conversion software, you will destroy the quality of audio for sure. Not too many people own this kind of software because it's expensive. I'm not sure about you but I am also a producer and I'm using a bunch of software synths, which to my ear sound radically better at 96k. That's why my advice is to always stick to 96k (or possibly 88k if you hear no difference) throughout the whole process. As a summary let me paste the official opinion of mixers like Bob Clearmountain, Joe Chiccarell, George Massenburg. Even though they speak about surround sound mixing here I do not see how surround sound audio would differ sample rate-wise from stereo audio. They state: "The mixing engineer may assume three things. First, the highest practical rate
should be chosen for original recording and mastering. Second, all known
resampling (sample-rate conversion) techniques are to some degree audible,
although recent work has shown that with great care and provided with
considerable DSP resources, one may resample, both up and down, very
effectively. And third, an integer ratio conversion (i.e., 96kHz to 48kHz, or
88.2kHz to 44.1kHz) is more likely to have fewer resampling artifacts.
It must always be understood that when upsampling audio (converting to a higher
sample rate, i.e., from 48kHz to 96kHz, or 44.1kHz to 96kHz), one will never
restore the upper octave filtered out at the original A/D conversion (i.e., when one
converts at 44.1kHz, there will be little signal above 20kHz from then on).
However, there are possible benefits to be gained from upsampling when
playback occurs. In any event, sample rate is not the sole, nor even the most
significant factor in perceived audio quality; the design and quality of the A/D and
D/A converters makes the most important sonic difference. A state-of-the-art
audiophile 44.1kHz / 16-bit converter can easily sound better than a 192kHz /
24-bit converter built using an inferior chipset. "
@@chrislysiak9561 Sorry but most of your information is pretty wrong, I don't care who you quote. I don't feel like going into this here, I'll just leave it at this: upsampling is NOT POSSIBLE. Period. A file recorded at 44.1 will always be at 44.1 you just add empty numbers (bigger file) if you "upsample" so no matter what software you use, is NOT audible because is doesn't do ANYTHING but making the file bigger.
We work with lower SR file at a higher SR project because plugins work better at 48, not converters, converters have the same chip, the same specs the same everything at any sample rate, what changes is the quality of the *recorded* file. As for how much difference and to what point, again, there is plenty of literature showing drawbacks at 192 which would be such minimal difference from the one happening between the bare minimum, 44.1 and 48.
With that said, whatever rocks your boat, just don't post incorrect info here. Upsampling is NOT possible.
@@mixbustv You're attacking the strawman you created. Both me and the other sound engineers I quoted say pretty much the same thing. Upsampling adds zeros. That's it. It was me that said we should mix at the sample rate you receive the session in, to which you stated that we should never mix at 44.1 but we should do it at 48khz. THIS IS UPSAMPLING. You are advocating upsampling, not me. I only said that if you have a choice to record a project at 48 or 96 I suggest going 96 simply because many converters sound better at that rate, especially the cheap ones. If you don't agree with this, then we have to agree to disagree. When it comes to the sound of the plugins, it all depends on the plugin. Some indeed sound better at 48 while virtual synths almost always sound better at 96k. I yet have to find one that sounds better at 48. Anyway, I do not wish to argue here. I just think that topics like that confuse people more than help them. With this comment, I will finish this topic. Peace!
@@chrislysiak9561 no arguments man, yes mine wanted to be a clarification, as is you can't change the quality of the original file, just that
Instagram demistifying pleeeease!!!!😂👍🏼
Long story short : upsampling dosent benefit the source but benefit the processing. I think. Thought?
Fleur De Lys definitely maybe.
Don't waste your time, just record and work in 44.1khz.
Stop tampering the comments with bulshit statements
if u dont care about music quality