music Sample and Bit Rate vs File Size of Soundtrack Music

[TheMusicBox]

Hi everyone

I just have a question about the preferred sound quality for soundtrack music.

I haven't really paid that much attention to this before, but I notice that I can produce files with a bit rate of anything between 80 and 320 kbps and sample rate of either 44.1 or 48 kHz.

As far as I understand it, the quality gets better the higher these numbers go (allegedly - I'm not convinced I can hear a difference to be honest!). However, I also realise that the file size dramatically increases the larger these numbers get too, which may or may not prove inconvenient at a later date.

Any suggestions as to what are the preferred settings or ideal scenario (sample rate and bit rate vs file size)?

I know I could probably read pages and pages on this subject online somewhere (and likely understand little of it!), but what is this forum if not too ask each other such questions?

Is there a sound guy/music supervisor or something out there with a preference?

Cheers

 

[AcousticAl]

... but I notice that I can produce files with a bit rate of anything between 80 and 320 kbps and sample rate of either 44.1 or 48 kHz.

Those are MP3/AAC numbers, meaning compressed formats that lose data. If you’re providing music for TV/film, you need to be providing uncompressed WAV or BWAV files. 16-bit/44.1kHz is CD-quality audio, but 24-bit/48kHz is much better. Audio for video traditionally runs at a native 16-bit/48kHz, but 24-bit is becoming more common.

On a project-by-project basis, though, you need to provide whatever format the client (or the sound designer) asks for.

As for what the numbers mean: yes, higher numbers mean better quality.

Bit rate means how many bits (or how many 1s and 0s) are captured per sample. The kHz rate means how many samples are captured per second. 16-bit means each word is 16 characters. 24-bit means 24 1s and 0s per word. 44.1kHz means there are 44,100 words (samples) captured per second. 48kHz means 48,000 words (samples) captured per second.

This is digital audio. Where analog is a constant and unbroken stream of sound as an electrical signal, in digital it’s all bits and samples per second. Those 1s and 0s are used to recreate the sound on playback. The more 1s and 0s you have, the better the fidelity of sound reproduction.

 

[TheMusicBox]

16-bit/44.1kHz is CD-quality audio, but 24-bit/48kHz is much better. Audio for video traditionally runs at a native 16-bit/48kHz, but 24-bit is becoming more common.

Thank you so much. With the 24-bit/48kHz settings on a WAV, I now notice that my 3 minute cue now takes up 51.6 MB!! Is that horrendous? Or I guess file size is a small price to pay for good quality.

I also seem to remember that Apple Macs get a bit stroppy with WAV files. Is there something I need to do to make sure this isn't a problem?

 

[AcousticAl]

Thank you so much. With the 24-bit/48kHz settings on a WAV, I now notice that my 3 minute cue now takes up 51.6 MB!! Is that horrendous? Or I guess file size is a small price to pay for good quality.

That’s actually not a whole lot of data these days. It may be too large to email, but that’s why we have things like DropBox.

I also seem to remember that Apple Macs get a bit stroppy with WAV files. Is there something I need to do to make sure this isn't a problem?

I have no idea what you’re talking about. I’ve been using Mac-based ProTools and FCP for a long, long time and I’ve never had issues with WAV files. All my ProTools sessions run in WAV format. Now, 20 years ago, a lot of us were converting all our audio to SDII files in ProTools, but it’s been a long time since that was a thing.

 

[TheMusicBox]

That’s actually not a whole lot of data these days. It may be too large to email, but that’s why we have things like DropBox.

I have no idea what you’re talking about. I’ve been using Mac-based ProTools and FCP for a long, long time and I’ve never had issues with WAV files. All my ProTools sessions run in WAV format. Now, 20 years ago, a lot of us were converting all our audio to SDII files in ProTools, but it’s been a long time since that was a thing.

I think it's been a while since I had a problem anyway. Thank you for all your reassurances! Much appreciated.

 

[Alcove Audio]

Good post, Al.

I've been running 24/48 .wav files for a very long time now. I even do all of my music sessions at these rates on the off chance they will be used for visual purposes. Yes, they are a bit larger, but absolutely minuscule compared to video files.

 

[BazTheHat]

One thing to remember it's that MP3 will often cut some of the very low frequency content to help make the file size smaller. Can't remember where the cut off point is, but if you've got some low thumping sub then mp3 will probably lose it.

In general, 128kbps is fairly difficult to tell apart from 320kbps on a consumer system, but the bigger and more impressive systems the music gets played on the more you're likely to hear artefacts, especially high and quiet ones as they're areas that get squashed to get that file size down.

In general, a 1 min stereo wav file at 16bit/44.1 kHz should take up about 10mb of space. Double that at 88.2khz. Mp3s are designed to be easier to download over the net and so are much smaller but quality gets reduced.

One last thing - Macs used to prefer AIFF files and not wavs, and vice versa on PC. These days no one cares, you can use either.

Hope that makes sense!

 

[TheMusicBox]

One thing to remember it's that MP3 will often cut some of the very low frequency content to help make the file size smaller.

Ah, I see. I thought some of my big baddie themes weren't coming through properly!

One last thing - Macs used to prefer AIFF files and not wavs, and vice versa on PC. These days no one cares, you can use either.

I knew there was something! Thanks for all this everyone.