I was assigned to watch this for my college audio-editing class, and I found this quite an informative video! I have learned several new terms today: sample rate, bit depth, and bit rate, as well as some other terms. Thanks for the video! I'll subscribe!
Yes exactly, wav is only a container and its "codec" (if you can call it that) is usually PCM. The wav container can hold almost any other codec actually but most devices/ software won't know what to do with it 😂
Interpolation happens any time the software has to make an educated guess about what is happening between the samples. This might happen if you are stretching or re-pitching audio, or just resampling a 44.1k clip into a 48k project.
@@DavidMacDonald thanks, but it’s hard to find which sinc formula is better to make that “guess” with and i’ve seen some linear interpolations that add samples to the file.
@@elijahjflowers interpolation always adds samples to the file. That’s its job. It isn’t ever going to be perfect and different algorithms will give different results in different circumstances. You just have to experiment.
Correct! Bitrate describes the amount of data used to store each second of audio. I suppose you could identify a data per second of uncompressed audio but it wouldn’t really mean anything comparable to bitrate in a lossy-compressed file.
@@WAVSAudioStudio all digital audio has bit depth. Anything that is an MP3 started life as uncompressed audio with a bit depth. That got encoded/compressed to MP3 and lost some data, but to play it, it gets decoded again and turned back into a table of amplitude values, which are stored with some bit depth. I would wager the overwhelming number of MP3s are compressed 16-bit audio, though many might have been 24-bit audio during the creation process.
@@DavidMacDonald Well said! But I could see from DAW that are bouncing 32 bit float MP3 files. I have only acces to set bitrate and sample rate but not bitdepth options. Whether we can convert the 32 bit float audio to 16 bit and so? Is it possible to do!! 💯
Sadly more bits does not capture more sound detail. More bits affects the noise floor. The more bits, the less noise or the noise is lower. Pictures and sound are two completely different things. Sound is recorded by math equations. Simply the maths always add up to one solution. The sound is captured perfectly but the quantising (bit depth) adds errors which adds noise. The noise on a 16 bit recording is too low to hear unless you boost it to deafening levels.
One thing critics always seem to miss is that quantization noise is _added_ to the signal. The original signal is still there in all its detail, no matter what bit depth you use. That's because the harmonic content depends on the frequency response, which depends on sampling rate, not bit depth!
I was assigned to watch this for my college audio-editing class, and I found this quite an informative video! I have learned several new terms today: sample rate, bit depth, and bit rate, as well as some other terms.
Thanks for the video! I'll subscribe!
Thanks so much for the comment! Would you mind sharing what college? I'm just curious!
Hands down the best simple explanation of this topic on youtube.
The best explanation on youtube. Thanks Bro
Yay College,... to be honest I learned a lot from this video thank you.
This deserves a lot more views.
Thanks a ton for explain things to me. Really appreciate it!
Dude… thanks for the info. 🖤
very clear and concise thank you for the wisdom
great video, thank you so much!
I would add that not all WAV files are uncompressed, although most of them are. The exceptions are WAV files using ADPCM compression.
@nicksterj Yes, although I had to look that one up. 😊 I have only worked with LPCM and ADPCM in real products.
Yes exactly, wav is only a container and its "codec" (if you can call it that) is usually PCM.
The wav container can hold almost any other codec actually but most devices/ software won't know what to do with it 😂
Which bitrate is the best for 1080p video in handbrake
thanks, do you have any tips for understanding audio interpolation?
Interpolation happens any time the software has to make an educated guess about what is happening between the samples. This might happen if you are stretching or re-pitching audio, or just resampling a 44.1k clip into a 48k project.
@@DavidMacDonald thanks, but it’s hard to find which sinc formula is better to make that “guess” with and i’ve seen some linear interpolations that add samples to the file.
@@elijahjflowers interpolation always adds samples to the file. That’s its job. It isn’t ever going to be perfect and different algorithms will give different results in different circumstances. You just have to experiment.
Thanks.
Is bitrate only applicable for lossy compressed formats like MP3 and aac
Correct! Bitrate describes the amount of data used to store each second of audio. I suppose you could identify a data per second of uncompressed audio but it wouldn’t really mean anything comparable to bitrate in a lossy-compressed file.
@@DavidMacDonald So wav formats don't follow bitrate? And also MP3 formats don't have bit depth?
@@WAVSAudioStudio all digital audio has bit depth. Anything that is an MP3 started life as uncompressed audio with a bit depth. That got encoded/compressed to MP3 and lost some data, but to play it, it gets decoded again and turned back into a table of amplitude values, which are stored with some bit depth. I would wager the overwhelming number of MP3s are compressed 16-bit audio, though many might have been 24-bit audio during the creation process.
@@DavidMacDonald Well said! But I could see from DAW that are bouncing 32 bit float MP3 files. I have only acces to set bitrate and sample rate but not bitdepth options. Whether we can convert the 32 bit float audio to 16 bit and so? Is it possible to do!! 💯
thank you
You're welcome
it will be better with a black background - it too bright
Sadly more bits does not capture more sound detail. More bits affects the noise floor. The more bits, the less noise or the noise is lower. Pictures and sound are two completely different things. Sound is recorded by math equations. Simply the maths always add up to one solution. The sound is captured perfectly but the quantising (bit depth) adds errors which adds noise. The noise on a 16 bit recording is too low to hear unless you boost it to deafening levels.
One thing critics always seem to miss is that quantization noise is _added_ to the signal. The original signal is still there in all its detail, no matter what bit depth you use. That's because the harmonic content depends on the frequency response, which depends on sampling rate, not bit depth!