Digital Audio 101 - Bit Depth, Sampling Rate, Interpolation - PB14

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This video covers the very basics of Digital Audio.
In the 2nd part I cover some harder principles, and going into more depth, you can watch Part 2 here: • Digital Audio 102 - PC...
I have again been experimenting with a new teaching style to keep the video engaging, so please do let me know what you think!
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Опубликовано:

22 июл 2024

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Комментарии : 64

@smaklilu90 4 года назад

No flashy animation, No crappy analogy, a tutorial that explains the point concisely and clearly!

@dhwanim175 2 месяца назад

so useful and concise- best explantation i have seen!

@vil9386 9 месяцев назад

Wow, what I thought would take me hours of reading/watching to understand these concepts got me under 5min. I wanted to have a quick grasp of these concepts for a voice recognition software that I'm working on, and this one helped me wonderfully. Thank you Veranova.

@AllysonLeah 9 лет назад

Good lord. This finally makes sense. Please keep teaching! You are so, so gifted. Kind gratitude for your work.

@marcusmgi8210 11 лет назад

I just pulled-up my DAW and zoomed in on a recording of myself jamming this morning...... and got goosebumps. Thanks for the valuable info, cleared up a lot of questions Ive always wanted to ask.

@andreymilyayev2783 8 лет назад

The best explanation I have seen. Vibewise too.

@MikeNorvak 11 лет назад

Along with the great and clear explanation, your video editing skills are as good as the content it self.

@uqmessias 5 лет назад

This is the best tutorial I could find for PCM. Thanks for that \o/

@ztrewqqwertz8997 6 лет назад

Wow, really good explanation, you just blowing the Information into my brain. Thank you.

@ApacienciaNECESSARIAparaOgoogl 11 лет назад

one of the best tut around utube....few views for such quality! thx for sharing it!

@officialmrgeekygamer 9 лет назад

Perfect explanation, super helpful

@30secondstocrash 3 года назад

Nice! Very well explained!

@kintubeats 3 года назад

Just watched this video for the 10nth time and i finally get it now 😆thanks!!!

@whatiknowis 3 месяца назад

Great vid!!

@JayPeek 9 лет назад

this is literally the Big Bang to understanding music creation. this simple quick tut. Synthesis theory has 5 wave forms, 3 elements (pitch volume, and timbre), and 7 components or ways in which to manipulate sound waves using the elements and wave forms respectively. (See NY school of synthesis 1985 vid by D. Friedman). This simple tut however can show you how to either be a synthesizer maker or build an original sound of your own _or_ how ugly duckling samples are cleaned up (using interpolation) and then remixed into a modern music swan (a recontextualization). What you are really watching here is how to turn the key to the door to your music career.

@MeiklesAndDimes 5 лет назад

Thanks! Very clear and concise!

@johnz3r 12 лет назад

Great video, learned a lot

@VasaMusic438 10 лет назад

Very Good and Clear, Thanks so much !!

@Layarion 7 лет назад

i'm trying to figure out the importance of "audio bitrate" on Open Broadcaster Software. gives the options of like 64,96,192,and 320

@EugenioBissacott 11 лет назад

Great ! good lesson!!

@DigitalBuoy 12 лет назад

well illustrated

@Alianger 4 года назад

Can you post an example of an interpolated/non-interpolated soundwave

@ScottishAtheist 12 лет назад

Fantastic!

@cyrilstephen5646 4 года назад

So I'm assuming sample rate os associated more with the representation of frequencies and bit depth is associated with the dynamic range? Please tell me if I'm wrong.

@abhinavitsme 11 лет назад

what is the best technique for distorted music interpolation without distorting the music further..linear interpolation like AR model tends to screw up the music

@johneymute 11 лет назад

Is liner interpolation common in any cd player?

@JayBenOh 12 лет назад

Ahh, so that's interpolation! There are several options in the interpolation-tab if i want to render something in FL-studio. Which one does actually make sense? (Or will this be in the next video? :-P)

@reynaldowify 6 лет назад

Hi, have a simple question, than nobody answers correctly to me CD have a bitdepht of 16, ok, and i always hear that 16 bit x 6 dB, offers a amplitude of 96 dB. I can understand that 3 dB in signal, means 6 dB in power Howevwer, and first, a wave must go forwards and backwards, so, there must be a bit for sign, leaving 15 bits for amplitude, since the amplitude of a wave is the difference between cero level, and the maximum posive, the same between cero an maximum negative, not the difference peak to peak Am i wrong?

@nolfnolfer1490 9 лет назад

what about the -1 side of the wave? you only showed from 0 to 1, whereas I think that it goes from 1 to -1

@JonFromWA 10 лет назад

I'm still not quite understanding Bit Depth. The definition that I gathered from "The Audio Dictionary" by Glenn D. White & Gary J. Louie is "Determines the number of steps and dynamic range of the digital system", in as simple definition as I could muster. Would you agree with this statement?

@Yousifalzoubi 4 года назад

What if the audio is muted for few seconds? will I have positive or negative binary? or I will get zeroes?

@djgmbh377 2 года назад

So analog isn't needed anymore or it only needed in some fields

@p1k1m4ru 12 лет назад

thanks for knowladge

@codycasteel25 9 лет назад

thank you

@Rollmops94 6 лет назад

I wonder one thing: If I would be very bored and I´d draw 50.000 of these dots. Could I in theory get one second of a Michael Jackson sounding song out of it that Michael Jackson actually never sang? I hope you understand me. Could I draw music?

@shrodingersman 8 лет назад

thanks for the video, I think it would be interesting if you made a video that had the raw sound without interpolation. Furthermore, what level of resolution would be so high as to not need interpolation

@TheUglyGnome 7 лет назад

"Raw sound" as you called it, doesn't make sense. Samples are (theoretically) infinitely short "blibs", which means they are audio bursts containing all frequencies. Whole sampling theorem only works, if you do the interpolation. And not just any kind of interpolation, but convolution with sinc-function (en.wikipedia.org/wiki/Sinc_function).

@emanuelsaavedra2412 10 месяцев назад

@@TheUglyGnome thank you for clarifying this!

@JayBenOh 12 лет назад

Alright, thanks. :-) Indeed, i'm still on FL-9, mabe i should update . . .

@johnpage4853 9 лет назад

I love your accent.

@SanDoval201970 7 лет назад

ok so if i record at 44.1 kHz , but the mastering house wants the wave file at 96 kHz, so then i render the the proyect recorded at 96khz , so thas this means i still have a good quality file as if was recorded at 96khz? i hop you understand mi question?

@avi1212avi 7 лет назад

Once you have sampled the original sound at 44.1 kHz, all the rest of the data is gone and you have a track sampled at 44.1kHz, if you try to play it back by sampling it at 200 kHz or w/e it won't matter since the source file does not have that additional data anymore. Imagine a graph that stretches from point A to B, I took 10 "samples" from that graph - meaning I put 10 dots on the graph at even distances from each-other, then I deleted the source graph and I only have the dots, I can't sample it again with 20 dots since all the data I currently have is 10 dots and I have to reconstruct the graph with these 10 dots next time I want to use it, Hope I explained it well cuz I just learned it from him just as you :D

@tristanreid9969 7 лет назад

very nice i like sorry for bad english

@resmediamarketing 10 лет назад

I subbed.

@Umziky 11 лет назад

If you harmor you could probably load the sound in there.

@jkman10 7 лет назад

so basically, the higher the frequency and bitrate(or depth?), the higher the resolution of sound and the better the quality?

@evgenetic3479 7 лет назад

theoretically yes, but given human ear limitation, 44/16 is enough to capture all audible to human audio and reproduce it with inaudible distortion (the inherent to 16 bit bit depth quantification error will results in -96 db noise). hi-res (above 44/16) audio is pointless for reproduction purposes.

@carlmansson1950 9 лет назад

Is interpolation done by a DAC? Or is that something different?

@TheUglyGnome 7 лет назад

Yes. Modern DAC first interpolates audio stream into something like 256 times the original sampling frequency using sinc-fuction convolution (en.wikipedia.org/wiki/Sinc_function). Then converts to analog and finally uses analog low-pass filter to get rid of very-high-frequency garbage created by conversion process.

@TheUglyGnome 5 лет назад

@Kelvin Klopper Nope. Dithering is another kind of animal. Dithering is not used in DAC, but in ADC in order to eliminate distortion created by quantizing of samples.

@brendantownsend216 9 лет назад

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?

@jian2069 9 лет назад

I am having the exact same question. And how does something like a low-pass filter or high-pass filter affect the sound wave?

@brendantownsend216 9 лет назад

Josh Kelly I know now.. when sound waves at different frequencies are travelling through the same medium at the same time they get added together producing a single complex wave. Check this out animagraffs.com/loudspeaker/

@jian2069 9 лет назад

Brendan Townsend makes sense thanks for the link man. the internet is fucking awesome.

@user-uv3pb9hd1q 8 лет назад

GJ thnx

@johneymute 11 лет назад

so there is 32bit audio ?why dit i have never heard of? and why do i read that that 32bit does not exist? i readed that pionear even had an 32bit audio system in the 1980's? some roumers say that even 64bit audio is used now by professionals. i don't know,many do say that even sacd don't match vynil audio quality,not strange if you think that there's still quantizing, even noise shaping is added to mask audio distortion,so i am affraid we are technically still far from having perfect audio !

$@fractla3347$

@fractla3347 3 года назад

Anyone else watching in 2020

@thevoid141 7 лет назад

0:34 Lol! :D

@fatimaghassan415 8 лет назад

😭

@brianf.burkejr.1488 9 лет назад

Nice simple explanation of PCM recording, now how about Bit Stream or DSD (Direct Stream Digital) where sound is sampled at a bit depth of 1 and a sample rate of either approx.. 2.6Mhz or 5.2 Mhz. This method gives a more realistic reproduction of the analog audio signal.

@cukedaddy 8 лет назад

+Brian F. Burke Jr. Notice he didn't answer...because it's not more realistic.

@Maybe_Later 7 лет назад

I think i'm falling in love with you.

@jessica-walt 10 лет назад

Your accent is hot.

@nbgydf 11 лет назад

NOT A PROFESSIONAL, ONLY MY UNDERSTANDING, NOT FACT Not necesarily. In a 16 bit recording, you are merely dividing the audio into segments of 2^16; in 24 bit you are dividing into segments of 2^24. you aren't actualy decreasing the volume/sound by going to a lower bit depth, you're just decreasing the definition of the sound.