Freesound Forums

Hi, guys.

I'm an amateur and hobbyist hip hop producer. My sound card is capable of recording and playing 24-bit 96 kHz wave files. When I work with this quality, the sound of my loops and tracks is just great. The problem is when I play the track on a computer with just an ordinary sound card, the sound is muddy, flat and weak. I understand that today's ordinary sound cards are able to play only 16-bit 44 kHz, so the solution would be to convert my tracks into this quality. But I'm not able to convert without losing too much characteristics (sound quality). I mean I'm pretty sure that if I played the track on my 24-bit card and recorded it with a microphone to a 16-bit file, the recording would sound more or less the same as the 24-bit original. But digital conversion just does not seem to work like this and the 16-bit version is always weak and sounds very different.

I believe the conversion must be doable somehow, because if you consider old skool vinyl analogue samples converted into 16-bit 44 kHz PCM waves sounding just great, there just must be a way.

There is free audio editing software called Audacity... http://audacity.sourceforge.net/
The 1.3 version of Audacity includes a "Resample" feature which would convert a 96KHz track to 44KHz in seconds.
(it also permits reducing the bit depth to 16).

You will lose the very high sound frequencies, above 22KHz, by converting to the 44KHz sample rate,
however many people would not be able to hear sounds above 20KHz anyway ... http://en.wikipedia.org/wiki/Hearing_range

Boosting the high frequency end of the 44KHz version may help matters (Audacity has a graphic equaliser to do this) but a 44KHz sample rate cannot reproduce frequencies above 22KHz, (which is the Nyquist frequency ... http://en.wikipedia.org/wiki/Nyquist_frequency)

BTW the degradation in sound quality on the "ordinary computer" could be because of its poorer quality speakers, rather than its lower sampling rate.

Thanks for the answer.

Man, I've already tried all the basic stuff. And resampling is surely not the way. I don't understand how exactly digital sound works, but resampling a 24-bit sample to a 16-bit one results in a TERRIBLE loss of quality (including clear panning, warmth and energy), not just something most people don't even notice. After the resampling, there is no point in doing anything to the sample to revive it somehow, because it is just a different sound.

If someone works with 24-bit samples and then wants to put his work on a CD, the sound would be totally different. So there must be a way to convert it so that there is not so much loss of quality.

Try it. Resample this (http://www.freesound.org/samplesViewSingle.php?id=29774) 24-bit one-shot drum sample to 16-bit and you'll hear the difference.

Hi Zajo,

I converted the mentioned sample and only heard a slight dynamic change which is not really a problem. No phase problems, no tone change and especially no stereo change.
Maybe its a good idea to upload your 24bit track somewhere (or freesound) so that we could try and turn it into a 16bit file.

I did it here using Acoustica 4.1 as well as Samplitude 10 Professional. All sounded very good.

The slight loss in dynamic range cannot be a problem. Maybe it is a completely different thing with your final track as you don't really have that much more dynamic range due to heavy compression. That might lead to these kind of artifacts.

Best,
Laribum

I just love it

Zajo

Resample this (http://www.freesound.org/samplesViewSingle.php?id=29774) 24-bit one-shot drum sample to 16-bit and you'll hear the difference.

I can only suggest that you may be unintentionally doing something which is altering sound quaility,
e.g. saving the mp3 version at a low bit rate, (e.g 32Kbps instead of 320Kbps ).

Well, laribum and Timbre, assuming your sound cards support 24-bit 96 kHz and that your resampling causes only slight quality loss, there must be a problem with my sound card. It's Audigy 2.

Here is a short part of a song in WMA, 24-bit, 96 kHz. Please, try to convert it to 16-bit, 44 kHz and send it back to me. If there is only slight quality loss, then it's obvious that I have a hardware problem. Thank you, people.

Zajo,

wma? what use is using 24/96 when you're using a compressed audio file format that most probably cuts off anything below 20khz!!
you should post the wave file (or flac file) to try on.
or are you using some kind of lossless compression wma format?

in general ALL compression formats (mp3, ogg, wma, m4p, ...) all drop the high-frequency content in order to compress more.

( and by compress I mean mp3-size-compression, not audio-type-compressor-compression )

- Bram

Man, I'm not sure, but WMA is the only format I can use that supports 24/96. That one WMA you have is 256 kbps, 24 bit, 96 kHz, stereo file. My major problem is that I don't have internet connection at home anymore so I must go back and forth between home and work. But it does not matter in this case. Save that WMA data as wave. Or I can. It just does not matter.

I used WMA, because neither MP3 nor any other compressed format I can use support more than 16/44.

Okay, to make it plain and simple, here is a wave format sound file, 24/96. Try to resample that and send it back to me and I will listen to it on my home computer and hear if it's okay.

Confusingly there are .wma formats which use lossless compression and other .wma formats which use lossy compression.

http://en.wikipedia.org/wiki/Windows_Media_Audio

http://en.wikipedia.org/wiki/Lossy_data_compression

http://en.wikipedia.org/wiki/Lossless_data_compression

If you accidentally used the lossy type of wma file it could noticeably compromise the sound quality.

Timbre, I used WMA only to make the file small so I could easily take it from home to work. And no, it does not do anything to the quality of the original wave, as when I play it on my home computer, it sounds the same as the original.

Just ignore the whole WMA stuff, as this topic has certainly nothing to do with that. I thank you for your interest. Please, let's try the wave conversion and we'll see.

http://media.freesound.org/files/sample_44100_16bit.wav

by the way, your wma is actually sampled at 48KHz, not at 96KHz.

- bram

Bram, where do you take that info from? I clearly remember rendering the track to a 96 kHz WMA and my home computer and my work computer both show that the file is at 96.

Thank you for the conversion. I'm going to take the file home and listen to it. I'll let you know then.

Okay, so I have the results.

Astonishingly, my quality sound card does a pretty bad job at playing either 16/44 or 24/96, or just at playing them in the same way. There is almost no audible difference between the original 24/96 and the converted 16/44. The problem with my sound card is that it plays 24/96 MUCH louder than 16/44 by default (which is weird), resulting in a seemingly much warmer and better sound. So, when I play a 24/96, it sounds very rich and warm and striking, but when I play a 16/44, it sounds silent and dull. So, my previous attempts at converting 24 to 16 were always successful, I just could not tell because of the card bug. Simply, the weird behaviour misled me. When I open 16-bit and 24-bit files in turns in my wave editor, the card sometimes go into the 'dull' mode, playing even 24-bit waves dull and silent.

Now, I'm on to find out how to fix the bug. The good thing is that I know that the card is capable of playing rich and warm sound. I just need to find out how.

Thank you all for your time and interest. I wish you love.

Which Sound Card (Manufacturer and Model) are you using ? Sorry If you've mentioned it already because I could not see it.

This is really weird behavior what you explained in last post.

I was very surprised with this discovery myself. The card is of Audigy 2 series, moderate quality, I think (I cannot get exact info since it's in my home computer and I'm at work now). By saying "quality sound card", I meant compared to other typical sound cards you normally get with your PC.

Hi Zajo,

Well the Audigy is far from being a quality soundcard, I'd say sound quality wise its about equal with the very latest chips that come on motherboards, or other similar consumer grade soundcards..

It's a bit confusing as to how you're working exactly. When you make your music in your sequencer, are you actually running at a project sample rate of 96K?. If so then thats fine, but if not, if say for eg you're working at 44K or 48K, then exporting at 96K won't make it sound any better, the project would need to be at 96K all the way through.

Secondly, always save as a .wav or .aiff file, uncompressed. WMA format just isn't a very good idea imho.

So ok lets assume your DAW was set to 96K sample rate and you've finished your production, applied any final polishing fx on the masterbus and exported the file as a 24-bit 96K .wav file.

Firstly you should resample the final track to 44K sample rate, use a good sample rate convertor, someone already suggested the audacity which is actually pretty good!. I personally use Voxengo R8Brain Pro (both high quality and nicely tweakable between linear and minimum phase).

What you want to do then is use a dither plugin to help in the reduction to 16bits, there's debate about whether dither algorithms only help maintain quality on fade ins/outs and very quiet bits, but I actually feel it helps reduce distortion of the waveform even in the loudest parts of the music.

It will never sound 100% as good as 24-bit, but you can get more than close enough to satisfy most people. Certainly with the kind of speakers most people listen to music you won't hear a difference.

Dithering doesn't actually turn the 24-bit file into a 16bit file, only prepares it as a 16-bit stream of data, so you will have to 'Save As' making sure its set to save as 16bit file.

Now you'll be left with (hopefully) a 16-bit 44K master .wav. With this you can then go on to encoding an mp3, wma or whatever other lossy formats you want to share with friends labels etc.

One final and important note. Some of the recent Creative Soundblaster cards have a thing called Crystalliser on it, which when enabled (they claim) magically makes recordings even better like sprinkling fairy dust on it.

For god sake whatever you do, make sure that crap is turned OFF!. It's a total gimmick and won't do your music any favours believe me.

Hope some of that information helped

A professional making sound and music for video games I worked with told me that Audigy was a quality sound card and that it was enough for his needs. I know there are incomparably better cards out there, but this is what he told me and what he used to create sound and music. But I don't want to talk about this, really.

Whenever I use a 24-bit 96 kHz sample in my song, I set the output to that level. If not, I let it at 16/44. But, well, to be honest, I'm not that stupid to make such trivial mistakes. The only problem, as I have written already, was that my sound card has an issue. It plays either very silent or very loud, and I'm not sure why and when, or rather what exactly triggers the change, but it has something to do with sampling rate. I've tried all possible configurations my sound card allows and nothing really helped. So the only problem was that I was mistaken by this weird behaviour and I falsely considered the loudness as warmth and quality boost, while it was really nothing but a volume boost. I started to work with 24/96 only recently, and, perhaps, that's why I did the mistake.

Man, I've been working with sound and music for 11 years so I certainly know how and when to use various audio formats (compressed or uncompressed). Why is WMA as an output format not a good idea? I'm not going to send 70 MB wave files to my friends to listen to my tracks. I'm going to use wave files for mastering or burning a CD, but I'm far from that now. And, WMA is one of formats that can play 24/96. MP3, to my knowledge, cannot, or at least my encoder cannot encode it so.

Crystalliser? Man, don't even think I've ever used any of that crap. I don't even use hardware equalizer or bass boost function. I'm always for the cleanest and clearest and unaffected sound possible.

If I seem offended, pardon me, please. I know that letters cannot express your thoughts that well so from what I've written, I may seem as a beginner to you. Nevertheless, I thank you for your help, Snozzle.