Hi Redbutton, (i can't directly reply to your message due to youtube weird changes...) But working on 88.2 khz will give you more 'samples' to work with and the anti-aliasing filter is way beyond the audible range. I only apply dithering once in the very last stage during export / downsampling of the master. Hope that helps.
Your videos are phenomenal. Thank you! I've been learning about sample rates and bit-depths in my electronics class at college and I'm a hobbyist guitarist that likes to record. Watching these videos really puts what I learn at college into a real world understanding for that helps me to learn relevant content for both fun and for my grades... Thank you taking the time to explain everything so well with little nonsense and with examples! Nice!
I have been recording my D&D games, and had no clue about any of the options my audio recording software was tossing in my face. This video explained a good bit of it and made it simple to understand! Thank you very much!
What an amazing video! One thing that he did not mention at the end was that just like in videography, if you edit in 10-Bit video and in the end downgrade to 8-Bit (because that's for example all the video on RUclips), your video quality will still be better in the end than if you had edited in 8-Bit. Even if you down sample a higher sample rate to 44.1 khz at the end, you will end up with a better 44.1 khz track than if you edited in 44.1 khz from the start.
Thank you! Just started dabbling in audio engineering after many years of being staunchly opposed to learning it (I.e. wanting to be purely a musician) and this has been a wonderful learning vid for me.
The question is, if we can only perceive up to a 20k hz frequency (generally speaking) what point is there in a higher sample rate that 44k? Is it even perceptible to our biology?
Especially when the vast majority of people aren't even capable of hearing those higher frequencies. I suppose the extra headroom of 24 bit is nice for recording. However, for the average consumer, 16 bit/44.1 khz would surely suffice.
I LOVE YOUR ViDEOS!!!! You are making my life so much easier. The school makes everything hard and complicated. You show in one video what they don't say within months in one coherent lecture.
This is the best video i’ve found explaining these! Awesome work. I record analog synths and personally hear a big enough difference that i release both cd quality and 24bit 96khz. Most videos just dismiss 96khz all together but it’s dumb to dismiss it when there are uses for it for a lot of people. The important thing is to know benefits of both so everyone can decide what they need. Hating on 96khz is unnecessary :)
That sounds like it could have multiple causes, but it seems you can hear yourself properly in the headphones, which I assume your mic-signal is being fed from your interface? If that's the case it should be coming in your interface correctly. It could perhaps be a buffersize setting from your DAW / Interface?
You should be able to find the samplerate settings used by reason under > Settings >> Audio. (To make full use of this, you should also be trying to use samples with higher sample-rates as well.)
Hey, I know you hear that a lot, but your videos are insanely good! Im so grateful for all the hard work and passion you put into this project. I really hope this pays of for you soon. I will definitely purchase your book when it comes out to support you. I LOVE YOU Greetings from Germany :)
I have been watching a lot of your videos recently and they have really helped further my understanding of sound engineering and producing. Keep up the amazing work dude!
Hi, I personally would than also prefer to mix at higher sample-rates, since you have that 'sample-data' available. If you're recording at 88.2 or 96KHz I would also mix and master at these sample rates and only convert the final master down to 44.1.
a 24 bit system does not have any more resolution the a 16 bit system, just a greater signal to noise ratio. A 10k tone will sound exactly the same at 16 or 24 bits. The only difference will be a slightly higher noise floor which you will not hear at a S/N of 96db.
And isn't the sample rate not a matter of quality, but one of the highest possible audio frequency that can be recorded? I thought the way it works is: bit depth = steps from loudest to quietest (not how loud it can go) sample rate = the highest frequency that can be recorded (divided by 2) (not quality of frequencies below the highest) I thought the only quality issue is whether or not the system makes mistakes. If everything is done right in a system that samples 48,000 times a second, then everything below 24,000Hz will be recorded perfectly. That's the theory anyway. Recording at 49,000Hz won't make 10kHz sound better, it will just allow you to capture 24.5kHz. Making mistakes will hurt all of the frequencies, not just the ones close to the maximum. I think mathematicians are failing the audio community on this. They need to explain it better so average people can understand.
Great video on explaining bit-depth and sample rates. I am new to audio production so this information is not what I am used to. The Math part got me hyped, though.
If you are mixing down a loop for a video game say a video game crowd cheer, what bit rate should you use? The game I want to change the crowd for is ridiculously low the original is 88kps the lowest I can get my export is 2116kbs and it won't play in game. The size of the original is 78.8 kb the size of the one I want to export is 1.21 mb is this why it won't play? It says it can not be played on the mci of the system, it has to be a wave file and does play on windows media player but not in the game, how can I fix this? It must be a wave file, how did they get theirs to be so small?
Great video.. 1 question I have.... Can I record my midi tracks and vocal tracks in 44.1? Then after change to 48 to mix and Export in 48? I'm very curious about this...What do you think?? Peace Outttttt Great TIPS From your boy Poorhouzzzzzzzz
Great video. I record audio for my video at 48/24bit WAV and was looking how the audio track could be used for Podcasts. I think from what you said in your video I should be good to use this file format for Podcasting as well. I would love to hear your opinion.
wooow this short video answer all my question in my head, thanks so much for sharing this knowledge! :D One question, i recorded vocal in studio that have a 48.000 sample rate and 24 bit depth, and i work in my own laptop in 44,1 sample rate and 24 bit, the vocal must be convert to 44,1 right? Is there a bad thing or disadvantage when converting the vocal track from 48.000 to 44,1 sample rate? I'm confuse about this :|
Great tutorials! But what's up with the website? There seems to not be much there. Nothing like the comments in the videos about what you can find there. i.e., I see no references to articles and such. Where are they hiding? Or did the site get hijacked, is it dead, or.....?
U know ur content is out standing and only u have that on whole youtube... man... ONLy Some promotion needed... Try to promote ur channel.. and of possible give a tour n live setup info. thanks ..
Hi, congrats for your videos, they are great and very explicative… today's recording standard is 192 kHz, but some companies are now creating converter systems of 384 kHz like antelope audio and Digital Audio Denmark… how much time do you give for 384khz to become the standard of musical production, what do you think about this? thanks
Would it be correct to say that when describing a wave curve in an XY graph, the sample rate is the amount of steps/second on the X axis and the bit depth is simply the amount of increments on the Y axis?
I need Tech/CD Burning help please. My issue is, someone convinced me to use a high bit rate when exporting my WAVs and I ended up exporting them @ 32 bit float at 480 HZ, using Audacity. But when I confirmed that the average CD player does NOT support 32 bits or 24 bits, I decided to export them @ the STANDARD 16 Bits at 441 Hz. And my CDs are still doing this random skipping crap during playback. WTF can I do to fix this!!!? I just want to kick back and listen to my new album without cringing at all of the random skipping! YES, I already tried burning at slower speeds(as low as 1X) and I've used iTunes, Windows Media Player and CD Burner XP. Are they still skipping because the 16 bit @ 441 Hz have traces of the 32 bit @ 480 Hz in the audio file? Please help. I'm exhausted trying to figure this crap out, its been over a week. The MP3s do NOT skip during playback on any device. And neither do the WAVs, WMAs nor do the CDs skip during playback on the laptop. Are the CDs I used just shit or is it something else? Because I feel the Universe is totally against me right now and I want to grab God by the throat and tell it to fuck off and leave me be lol! I have NEVER had this issue before. . . (BTW I recorded the whole album on a 4 track recorder and only used Audacity to raise the volume on the songs before exporting them)
Please ask, sir, this is related to the Type C to Jack Audio Converter Adapter which already has a DAC, there are several options on the marketplace, some are 24bit/96 khz, then some are 32 bit/384khz, which one is the best at capturing small sounds but sounds clear which one bro? this is for the case of playing PUBG games to be more sensitive to listening to distant enemy steps.
Hey, Wick, what do you think of recording at higher sample rates (e.g. 96khz) and then downsampling to 44.1khz for the mix, with good SRCs (like Izotope RX, r8brain Pro, etc)?
Difficult question, since that really depends on your system and resources.. If your computer can easily handle recording and mixing at 96 khz, I would go for that! (Which is a lot more to handle processing-wise)
Absolutely, I would definitely love to mix at 96khz; but with the sessions I get, there are many tracks, so my system gets bogged down. If I made it a ritual to bounce the tracks often, this this could work, but I do some complex sidechaining of dynamic EQs on tracks, FX channels, and busses, so the bouncing is not always an option. I would theorizing that taking the highest quality picture I can get (rec @ 96) and then downsampling with good SRC for the mix could be better than rec @ 44.1.
There's no point in sampling higher than twice the highest audible frequency, other than making the process of preventing noise from folding back. Any noise at a frequency above half the sample rate will "fold back" and appear as a signal below the half sampling rate. That's where the analog low pass filter comes in. But it can add uneven group delay, so by oversampling, you can place the cutoff frequency higher. The phase distortion is much less in the 20 to 20Khz range this way. You then decimate and get rid of the extra samples. Once it's digitized you don't need to drag around more than 44Khz' worth of bits. So I'm not sold on the value of working with 96Khz files.
Modern interfaces do exactly this: Oversampling, than outputting at the sample rate you actually dial. But people will never learn, as we can see here. Doing the same like having 24 MPixel behind a phone camera lens.
mdesm2005 I record analog synths every day and at 44.1 it is missing the whole analog synth sound. I’ve paid 12 000$ worth of analog gear and if i only use 44.1 they will just sound like digital vst sounds. So i mix both 96 lossless and 44.1 cd. It’s stupid to dismiss higher khz rates because there are uses for all rates that are available.
Yes, by oversampling ( digitally ) we can eliminate the images, more or less, but there is an issue here. The reconstruction filter ( that does the over sampling) uses sharp fir-filters and the process is very efficient, but there is one caveat; the filter can only reconstruct static signals that doesn't change their amplitude or frequency. As soon as the wave changes, the filter will be momentarily "confused" whether it shall regard the samples as phase information or amplitude information. In the world of music, the waves changes all the time and the filter won't reproduce the totality of the music correctly. Attacks and decays will be somewhat smeared out in the time domain. Recent research has shown that our hearing is very sensitive to the phase relationship between various transients. Using a higher sampling rate makes this effect much lesser. And we can use less sharp filters that doesn't affect the signal as much as now. The capacity of internet will grow and in the future, there is no need for sticking to 44.1khz. 192khz and lossless compression will do. Traditionalists may shake their heads, but I hope there will be a consensus around PCM and higher samlping rates. Bit depth is not that important. Formats like SACD is a dead end since it relies on sigma delta conversion, which limits the resolution at higher frequencies. This is an averaging conversion and can't respond to hf nuances in a proper way. But this is a limitation of all sigma delta devices. But by using PCM we won't paint our self into any sigma-delta corner.
If I were to line in an mp3 player (Amazon Echo dot) into my PC Line IN. Should I use a 48KHz to play in my Line out? It gave me two options CD(44khz) or DVD(48Khz) format. And should I worry about my PC breaking if I mess with these settings? thanks for your video :)
For electronic music, is it possible to make the song (with both a mix of audio files synths used) at 44,100khz and then when it's time to render and master the song increase the sample rate to 96,000khz? Or will this mess with the track?
If i've got two sounds and 48k but diferent bit rate, will one of them end longer than the other? Or it is only de frame rate the one who make that happen. Thanks!
Please can someone explain my query : If we are recording and digitizing a song being played which has alot of bass and treble together for example and of course the amplitudes of the different frequencies are different. If we take a sample at any one point in time, we are getting a binary number that represents an amplitude. Which amplitude is this if there are various amplitudes for the different frequency components at any one point in time? Isn't sampling done 1 dimensional? How are we able to get all of the information about the different frequencies and levels then?
From what I understand you aren't actually digitizing the "audio fidelity" of an audio sound wave. What you're digitizing is the the electromagnetic signals used by a microphone or loudspeaker/headphone. The microphone, or analog recording device, converts sound into an electrical signal. It's the electrical signal that is being digitized (sampled). I think it's the characteristics of those electrical signals (line-level electrical signals) that are encoded. Not the actual audio frequency ranges, but the signal that creates the frequency outputs, if you know what I mean. Myself, I'm trying to get it clear in my head how a digital word size, whether 8 bit, 16 bit or 24 bit, etc, is encoded (digital line coding) in Pulse Code Modulation for example. There are different ways of encoding data into a carrier wave and I'm not sure what standard is used by PCM.
If I want to sync an audio file recorded in 192kHz with a video recorded at 25fps, ignoring the audio track on the movie, do I need to configure something on the camera ?
thanks for the lesson! I have a question. So I plug in my Lexicon Alpha with Shure58 to my laptop. Everything is fine, I can hear myself perfectly well in the headphones (without any noise). But when I record something in Cubase, there is a very annoying constant noise (like squeak). The noise is still there even if the mic is off( when i push Monitor button). Does it somehow connected to the topic of this lesson?
I am a newbie knob at all this. I bought a Focusrite Saffire 6 usb interface to plug my guitar into the laptop and take advantage of what there is as far as amp simulators go. I downloaded the drivers for the saffire 6 and Windows 10 is ok with it. However when I go to set up the software ..Amplitude 4..or the Bias Amp and when I attempt to change the settings from "Directsound" or "Windows Audio" to ASIO the software crashes. Sometimes I get an error saying that windows is unable to set the sample rate another program is using it (even though everything is closed out) or I get an error saying that the input and output devices don't share a common bitrate and that I should fix it in settings --which does no good? any help would be appreciated ..
Here's some trivia. WHY was the original digital audio/CD sampling rate set to 44.1khz? Its because the human ear can only hear frequencies between about 20hz at the lowest, to frequencies of 20khz at the highest. Some are more sensitive, or not, than others. So, the 44.1khz made the digital sampling rate TWICE what the high end of that range is PLUS some overhead, so that it was still twice as fast as what even the most sensitive human ears can hear, PLUS a little bit more to be sure.
im confused about bit rate being sample rate x bit depth. so cd got 1411kbps, sample rate of 44,1kHz and 16bits, but 44,1 x 16 only equals half of the supposed bit rate, should it be sample rate x bit depth x 2? Also I need an explanation on why is it times 2
I apologize for not appreciating the theory of clocks and samples rates. Just tell me what to do? show me where these adjustments are made foe protools 10.3 and macBook pro
In the last 10 years or so converters became so similar analogue stages tend to dictate their sound. Some manufacturers even use discrete analogue stages, some add transformers for "more sounds". New generation IC's like from THAT became as good as discrete options, often even better...
This is probably the best tutorial I've ever seen on RUclips. Keep it up!
Hi Redbutton, (i can't directly reply to your message due to youtube weird changes...) But working on 88.2 khz will give you more 'samples' to work with and the anti-aliasing filter is way beyond the audible range. I only apply dithering once in the very last stage during export / downsampling of the master. Hope that helps.
Your videos are phenomenal. Thank you! I've been learning about sample rates and bit-depths in my electronics class at college and I'm a hobbyist guitarist that likes to record. Watching these videos really puts what I learn at college into a real world understanding for that helps me to learn relevant content for both fun and for my grades... Thank you taking the time to explain everything so well with little nonsense and with examples! Nice!
Ok
I have been recording my D&D games, and had no clue about any of the options my audio recording software was tossing in my face. This video explained a good bit of it and made it simple to understand! Thank you very much!
What an amazing video! One thing that he did not mention at the end was that just like in videography, if you edit in 10-Bit video and in the end downgrade to 8-Bit (because that's for example all the video on RUclips), your video quality will still be better in the end than if you had edited in 8-Bit.
Even if you down sample a higher sample rate to 44.1 khz at the end, you will end up with a better 44.1 khz track than if you edited in 44.1 khz from the start.
Dracule Mihawk thanks a lot. It makes it easier to understand all those X Y KHz
Seeing this video on 2021 , very helpful! A pity your website isnt available any more. Thank you!
Thank you! Just started dabbling in audio engineering after many years of being staunchly opposed to learning it (I.e. wanting to be purely a musician) and this has been a wonderful learning vid for me.
This is great stuff, discovered your channel during breakfast today, bought both books before lunch :-)
The question is, if we can only perceive up to a 20k hz frequency (generally speaking) what point is there in a higher sample rate that 44k? Is it even perceptible to our biology?
Especially when the vast majority of people aren't even capable of hearing those higher frequencies. I suppose the extra headroom of 24 bit is nice for recording. However, for the average consumer, 16 bit/44.1 khz would surely suffice.
Thanks Dude.. You made it simple..
Thanks so much man, you make the best tutorials, the best thing about them is they are categorized and Simplified, keep up, you're awesome 😁👌
5:16 I think it's 96000Hz instead of 92000Hz
and 192000Hz instead of 196000Hz.
I LOVE YOUR ViDEOS!!!! You are making my life so much easier. The school makes everything hard and complicated. You show in one video what they don't say within months in one coherent lecture.
You covered a fairly heady subject and made it understandable.....in 8 minutes. Subscribed. Fantastic work, brother.
Wow man i watch alot of different stuff on youtube in various genres and i have to say This is a great and informative video man! good job!
This is the best video i’ve found explaining these! Awesome work. I record analog synths and personally hear a big enough difference that i release both cd quality and 24bit 96khz. Most videos just dismiss 96khz all together but it’s dumb to dismiss it when there are uses for it for a lot of people. The important thing is to know benefits of both so everyone can decide what they need. Hating on 96khz is unnecessary :)
amazing tutorial, ur the best, happy to find your channel, keep it up
That sounds like it could have multiple causes, but it seems you can hear yourself properly in the headphones, which I assume your mic-signal is being fed from your interface? If that's the case it should be coming in your interface correctly. It could perhaps be a buffersize setting from your DAW / Interface?
Best tutorial on bit depth and sampling out there!
You should be able to find the samplerate settings used by reason under > Settings >> Audio.
(To make full use of this, you should also be trying to use samples with higher sample-rates as well.)
Hey, I know you hear that a lot, but your videos are insanely good! Im so grateful for all the hard work and passion you put into this project. I really hope this pays of for you soon. I will definitely purchase your book when it comes out to support you.
I LOVE YOU
Greetings from Germany :)
I have been watching a lot of your videos recently and they have really helped further my understanding of sound engineering and producing. Keep up the amazing work dude!
Hi, I personally would than also prefer to mix at higher sample-rates, since you have that 'sample-data' available. If you're recording at 88.2 or 96KHz I would also mix and master at these sample rates and only convert the final master down to 44.1.
Been binging on your tutorials. Subscribed already. Really easy method of advancing my knowledge. #BigUpsMan
Thank you so much!!! Best explanation of this stuff that I have come across in 15 years of recording!!!
a 24 bit system does not have any more resolution the a 16 bit system, just a greater signal to noise ratio. A 10k tone will sound exactly the same at 16 or 24 bits. The only difference will be a slightly higher noise floor which you will not hear at a S/N of 96db.
And isn't the sample rate not a matter of quality, but one of the highest possible audio frequency that can be recorded?
I thought the way it works is:
bit depth = steps from loudest to quietest (not how loud it can go)
sample rate = the highest frequency that can be recorded (divided by 2) (not quality of frequencies below the highest)
I thought the only quality issue is whether or not the system makes mistakes. If everything is done right in a system that samples 48,000 times a second, then everything below 24,000Hz will be recorded perfectly. That's the theory anyway. Recording at 49,000Hz won't make 10kHz sound better, it will just allow you to capture 24.5kHz. Making mistakes will hurt all of the frequencies, not just the ones close to the maximum.
I think mathematicians are failing the audio community on this. They need to explain it better so average people can understand.
True, if we want to reduce noise floor record at nearly 0db at 32 bits and reduce gain
Great video on explaining bit-depth and sample rates. I am new to audio production so this information is not what I am used to. The Math part got me hyped, though.
Insanely informative, well done.
Crystal clear concepts. Thanks a ton!
Most of this I already assumed for the most part, but this straightened out a few things I was iffy about. For that, I thank you!
Very well done explanation. Nice job!
If you are mixing down a loop for a video game say a video game crowd cheer, what bit rate should you use? The game I want to change the crowd for is ridiculously low the original is 88kps the lowest I can get my export is 2116kbs and it won't play in game. The size of the original is 78.8 kb the size of the one I want to export is 1.21 mb is this why it won't play? It says it can not be played on the mci of the system, it has to be a wave file and does play on windows media player but not in the game, how can I fix this? It must be a wave file, how did they get theirs to be so small?
PERFECT EXPLANATION!!
BEST VIDEO ON THIS TOPIC
Great video.. 1 question I have.... Can I record my midi tracks and vocal tracks in 44.1? Then after change to 48 to mix and Export in 48? I'm very curious about this...What do you think?? Peace Outttttt Great TIPS From your boy Poorhouzzzzzzzz
Very good informative video ! Thanks
Great video. I record audio for my video at 48/24bit WAV and was looking how the audio track could be used for Podcasts. I think from what you said in your video I should be good to use this file format for Podcasting as well. I would love to hear your opinion.
exactly what we learned at signal processing in CS =)
you too great .. explanation is very effective and simple easy to grab your knowlege
Don't really care about being first but I'd really like for you to know that I GREATLY appreciate these lessons.. So... much.
well done... the best explanation I have seen/heard/read
This is a terrific explanation.
wooow this short video answer all my question in my head, thanks so much for sharing this knowledge! :D
One question, i recorded vocal in studio that have a 48.000 sample rate and 24 bit depth, and i work in my own laptop in 44,1 sample rate and 24 bit, the vocal must be convert to 44,1 right? Is there a bad thing or disadvantage when converting the vocal track from 48.000 to 44,1 sample rate? I'm confuse about this :|
awesome video / demo...clarified a muddled mind such as mine MANY THANX !!
this is an EXTREMELY well made video. Thank you!
Excellent! Answered my questions. Thanks.
Amazing video, really helped me out
Very clearly explicated; thank you~
Great tutorials!
But what's up with the website? There seems to not be much there. Nothing like the comments in the videos about what you can find there. i.e., I see no references to articles and such. Where are they hiding? Or did the site get hijacked, is it dead, or.....?
You're good teacher sir !
U know ur content is out standing and only u have that on whole youtube... man... ONLy Some promotion needed...
Try to promote ur channel.. and of possible give a tour n live setup info. thanks ..
Hi, congrats for your videos, they are great and very explicative… today's recording standard is 192 kHz, but some companies are now creating converter systems of 384 kHz like antelope audio and Digital Audio Denmark… how much time do you give for 384khz to become the standard of musical production, what do you think about this?
thanks
Would it be correct to say that when describing a wave curve in an XY graph, the sample rate is the amount of steps/second on the X axis and the bit depth is simply the amount of increments on the Y axis?
hey man, how do you do this video???
This is very educational. Thank you
5:04 that's actually a real task in our school computer class graduation exam. To calculate how much disk space will it take.
Awesome very informative .Thanks a Million
Very well explained!!!
Love your tutorials. Thanks.
Amazing video, thank you!
Thank you so much for the clear explanation!
I need Tech/CD Burning help please. My issue is, someone convinced me to use a high bit rate when exporting my WAVs and I ended up exporting them @ 32 bit float at 480 HZ, using Audacity. But when I confirmed that the average CD player does NOT support 32 bits or 24 bits, I decided to export them @ the STANDARD 16 Bits at 441 Hz. And my CDs are still doing this random skipping crap during playback. WTF can I do to fix this!!!? I just want to kick back and listen to my new album without cringing at all of the random skipping! YES, I already tried burning at slower speeds(as low as 1X) and I've used iTunes, Windows Media Player and CD Burner XP. Are they still skipping because the 16 bit @ 441 Hz have traces of the 32 bit @ 480 Hz in the audio file? Please help. I'm exhausted trying to figure this crap out, its been over a week. The MP3s do NOT skip during playback on any device. And neither do the WAVs, WMAs nor do the CDs skip during playback on the laptop. Are the CDs I used just shit or is it something else? Because I feel the Universe is totally against me right now and I want to grab God by the throat and tell it to fuck off and leave me be lol! I have NEVER had this issue before. . . (BTW I recorded the whole album on a 4 track recorder and only used Audacity to raise the volume on the songs before exporting them)
A great help Wik!
Please ask, sir, this is related to the Type C to Jack Audio Converter Adapter which already has a DAC, there are several options on the marketplace, some are 24bit/96 khz, then some are 32 bit/384khz, which one is the best at capturing small sounds but sounds clear which one bro? this is for the case of playing PUBG games to be more sensitive to listening to distant enemy steps.
very complete and informative video. cheers :-)
Awesome info thank you!
Hey, Wick, what do you think of recording at higher sample rates (e.g. 96khz) and then downsampling to 44.1khz for the mix, with good SRCs (like Izotope RX, r8brain Pro, etc)?
doesn't working in higher sample rate and then downsample it to 44.1 khz in mixing state alter the pitch?
Ok, so I'm a geek. Can you please explain why 8 and 1024 when dividing the bits and kilobytes?
Difficult question, since that really depends on your system and resources.. If your computer can easily handle recording and mixing at 96 khz, I would go for that! (Which is a lot more to handle processing-wise)
Absolutely, I would definitely love to mix at 96khz; but with the sessions I get, there are many tracks, so my system gets bogged down. If I made it a ritual to bounce the tracks often, this this could work, but I do some complex sidechaining of dynamic EQs on tracks, FX channels, and busses, so the bouncing is not always an option. I would theorizing that taking the highest quality picture I can get (rec @ 96) and then downsampling with good SRC for the mix could be better than rec @ 44.1.
There's no point in sampling higher than twice the highest audible frequency, other than making the process of preventing noise from folding back. Any noise at a frequency above half the sample rate will "fold back" and appear as a signal below the half sampling rate. That's where the analog low pass filter comes in. But it can add uneven group delay, so by oversampling, you can place the cutoff frequency higher. The phase distortion is much less in the 20 to 20Khz range this way. You then decimate and get rid of the extra samples. Once it's digitized you don't need to drag around more than 44Khz' worth of bits. So I'm not sold on the value of working with 96Khz files.
Modern interfaces do exactly this: Oversampling, than outputting at the sample rate you actually dial. But people will never learn, as we can see here. Doing the same like having 24 MPixel behind a phone camera lens.
mdesm2005 I record analog synths every day and at 44.1 it is missing the whole analog synth sound. I’ve paid 12 000$ worth of analog gear and if i only use 44.1 they will just sound like digital vst sounds. So i mix both 96 lossless and 44.1 cd. It’s stupid to dismiss higher khz rates because there are uses for all rates that are available.
Yes, by oversampling ( digitally ) we can eliminate the images, more or less, but there is an issue here. The reconstruction filter ( that does the over sampling) uses sharp fir-filters and the process is very efficient, but there is one caveat; the filter can only reconstruct static signals that doesn't change their amplitude or frequency.
As soon as the wave changes, the filter will be momentarily "confused" whether it shall regard the samples as phase information or amplitude information.
In the world of music, the waves changes all the time and the filter won't reproduce the totality of the music correctly. Attacks and decays will be somewhat smeared out in the time domain. Recent research has shown that our hearing is very sensitive to the phase relationship between various transients.
Using a higher sampling rate makes this effect much lesser. And we can use less sharp filters that doesn't affect the signal as much as now.
The capacity of internet will grow and in the future, there is no need for sticking to 44.1khz.
192khz and lossless compression will do. Traditionalists may shake their heads, but I hope there will be a consensus around PCM and higher samlping rates. Bit depth is not that important.
Formats like SACD is a dead end since it relies on sigma delta conversion, which limits the resolution at higher frequencies. This is an averaging conversion and can't respond to hf nuances in a proper way. But this is a limitation of all sigma delta devices. But by using PCM we won't paint our self into any sigma-delta corner.
Quick question.
What does "float" mean, for example, I'm exporting audio for an album im doing, and the highest bit depth is 32bit (float)
Excelent explanation! Thanks!
first minute: whats the y-axis ? is it metre, pillows or carpets?
If I were to line in an mp3 player (Amazon Echo dot) into my PC Line IN.
Should I use a 48KHz to play in my Line out?
It gave me two options CD(44khz) or DVD(48Khz) format.
And should I worry about my PC breaking if I mess with these settings? thanks for your video :)
For electronic music, is it possible to make the song (with both a mix of audio files synths used) at 44,100khz and then when it's time to render and master the song increase the sample rate to 96,000khz? Or will this mess with the track?
If i've got two sounds and 48k but diferent bit rate, will one of them end longer than the other? Or it is only de frame rate the one who make that happen. Thanks!
Thanks! What bit depth and sample rate do you use while mixing and mastering for example a "standard" band song? :)
Please can someone explain my query :
If we are recording and digitizing a song being played which has alot of bass and treble together for example and of course the amplitudes of the different frequencies are different.
If we take a sample at any one point in time, we are getting a binary number that represents an amplitude. Which amplitude is this if there are various amplitudes for the different frequency components at any one point in time?
Isn't sampling done 1 dimensional? How are we able to get all of the information about the different frequencies and levels then?
From what I understand you aren't actually digitizing the "audio fidelity" of an audio sound wave. What you're digitizing is the the electromagnetic signals used by a microphone or loudspeaker/headphone. The microphone, or analog recording device, converts sound into an electrical signal. It's the electrical signal that is being digitized (sampled). I think it's the characteristics of those electrical signals (line-level electrical signals) that are encoded. Not the actual audio frequency ranges, but the signal that creates the frequency outputs, if you know what I mean.
Myself, I'm trying to get it clear in my head how a digital word size, whether 8 bit, 16 bit or 24 bit, etc, is encoded (digital line coding) in Pulse Code Modulation for example. There are different ways of encoding data into a carrier wave and I'm not sure what standard is used by PCM.
Logged in just to Like. Thank you!
Is it okay if I use 44.1kHz/24bit
If I want to sync an audio file recorded in 192kHz with a video recorded at 25fps, ignoring the audio track on the movie, do I need to configure something on the camera ?
That was very well done!
thanks for the lesson! I have a question. So I plug in my Lexicon Alpha with Shure58 to my laptop. Everything is fine, I can hear myself perfectly well in the headphones (without any noise). But when I record something in Cubase, there is a very annoying constant noise (like squeak). The noise is still there even if the mic is off( when i push Monitor button). Does it somehow connected to the topic of this lesson?
Loved it completely
I am a newbie knob at all this. I bought a Focusrite Saffire 6 usb interface to plug my guitar into the laptop and take advantage of what there is as far as amp simulators go. I downloaded the drivers for the saffire 6 and Windows 10 is ok with it. However when I go to set up the software ..Amplitude 4..or the Bias Amp and when I attempt to change the settings from "Directsound" or "Windows Audio" to ASIO the software crashes. Sometimes I get an error saying that windows is unable to set the sample rate another program is using it (even though everything is closed out) or I get an error saying that the input and output devices don't share a common bitrate and that I should fix it in settings --which does no good? any help would be appreciated ..
Here's some trivia. WHY was the original digital audio/CD sampling rate set to 44.1khz?
Its because the human ear can only hear frequencies between about 20hz at the lowest, to frequencies of 20khz at the highest. Some are more sensitive, or not, than others. So, the 44.1khz made the digital sampling rate TWICE what the high end of that range is PLUS some overhead, so that it was still twice as fast as what even the most sensitive human ears can hear, PLUS a little bit more to be sure.
im confused about bit rate being sample rate x bit depth. so cd got 1411kbps, sample rate of 44,1kHz and 16bits, but 44,1 x 16 only equals half of the supposed bit rate, should it be sample rate x bit depth x 2? Also I need an explanation on why is it times 2
You need two channels for stereo.
thank you very much providing dis info...
perfect explanation
I apologize for not appreciating the theory of clocks and samples rates. Just tell me what to do? show me where these adjustments are made foe protools 10.3 and macBook pro
Love it thanks for all the info and knowledge. Thanks alot. Keep these videos coming up please. Peace
Tank you. Very well done
Hi, have one simple question, does;t mean 16 bit, that one is for the sign? And there are only 15 left for amplitude?
Thanks
No 65536 possible values with 0 being -, 32767 being zero and 65536 being +.
In the last 10 years or so converters became so similar analogue stages tend to dictate their sound.
Some manufacturers even use discrete analogue stages, some add transformers for "more sounds". New generation IC's like from THAT became as good as discrete options, often even better...
Interestingly enough, this seems to apply a bit further than just audio analysis like for example Software Defined Radios
wow I am going to have to see this again great video how do I change this on my DAW I use Reason 7
Please I need help, I have a csv file with the UTC time and sampling time. How do I get the video frame rate lined up with the EDA sampling rate?
I appreciate you letting me know that.... :)
Thanks!