Thoughts on dithering.....

[Oren]

...Yes, when I cannot see any difference at all in the wave form in Audition, it makes perfect sense to not be dithering around with it.   ...

 

As for the noise shaping, it has 11 different patterns, and it suits me fine to leave dithering out of the equation, since I haven't a clue, (yet), as to what they are all about.

Eleven - Woh!!!

So as of now, I am turning off dithering everywhere, and leaving it that way unless and until the time comes when I might have a reason to use it.

Seems reasonable...

As far as sampling rate goes, is there an advantage to using 24/48  over  24/44.1, since my destination is 16/44.1 ?

There is a tremendous amount of disagreement on this point, even among the experts. Personally, I can't hear the difference between the three. Theoretically, there is some advantage to using a higher bit-rate and sampling rate when processing digital audio, but once again... I can't hear any difference in the results.
For a while I was using 24bit/48kilohertz for all my audio files, but could not realistically perceive any improvement in sound quality. Consequently, I've reverted to using 16/44.1, unless someone gives me an audio file that's mixed down to one of the higher rates.

How much of this trend to push audio processing to the limits of a modern computer's capability actually results in real-world audio improvement?

[Wyatt]

There is a tremendous amount of disagreement on this point, even among the experts. Personally, I can't hear the difference between the three. Theoretically, there is some advantage to using a higher bit-rate and sampling rate when processing digital audio, but once again... I can't hear any difference in the results.
For a while I was using 24bit/48kilohertz for all my audio files, but could not realistically perceive any improvement in sound quality. Consequently, I've reverted to using 16/44.1, unless someone gives me an audio file that's mixed down to one of the higher rates.

How much of this trend to push audio processing to the limits of a modern computer's capability actually results in real-world audio improvement?

Interesting..thanks for your advice.

Now maybe I can let this question rest for another year or two!   

[Fred S]

I use 24 bit when I'm just playing around, especially with sampled pianos, where I do "think" I hear a difference. But for recording and distribution, I just use 16/44.1. What has always bothered me about higher quality is that you eventually end up dithering it down most of the time anyway to CD quality. Then whatever sonic quality improvement achieved is lost. Further, as you are developing your project, you are theoretically listening to something different than the eventual result. So, although you want the best rendering possible, what's the point...really. If you work in 16/44.1, your hearing exactly what the end consumer will be listening to. Looking at it in that way, it actually seems like an advantage to work in CD quality.

[rharv]

I can hear the difference between recording the same thing at 16 and 24 bit.  That I have tested.

It still does make sense to work in 24 bit (IMHO) and dither at the very end.

The difference is in the size of each sample (or chunk) getting grabbed digitally.

24 bit grabs larger samples than 16 bit.  These samples occur at a certain speed (44.1kHz for example) which means they are small digital samples of a sound over time.

Now if you could grab a small sample (16bit) with tweezers, imagine the sample you could bet with a big set of tongs!

How these samples are bigger is because they take a larger bite of time.  This adds more depth to the sound.  During processing every effect has better data to work with.. so gives better results.

The resulting 'depth' gets retained during dithering (as much as possible)

There IS a difference, which is why studios use 24 bit versus 16..

The key is to use a good A/D/A to begin with and a good dithering program at the end, so the depth is retained as much as possible.
I used to really be confused about it all, and then spent some time reading, listening, testing ..
still not an authority but I can hear the difference now.

Moon can probably explain about the exponential difference this size sampling makes.. it sounds like an 8 bit difference, but it isn't.. its much greater

This difference in dithering results is why I said earlier that Ozone is the best one I have used to date.

[Fred S]

Makes sense!

[Wyatt]

It still does make sense to work in 24 bit (IMHO) and dither at the very end.

I'm with you on the bit depth part..but I still have a question about the sample rate.

I have stuck to 16 bit up until now, because I couldn't hear the difference, (though admittedly, that isn't setting the bar very high). I knew that in my restoration work, noise reduction works better, (with less aliasing), at higher bit depth, but I have developed a number of strategies that have given me very accurate, clean noise reduction @ 16 bit..so I kept it simple.

Now that I am using Reason in conjunction with my 'real' instruments, (ooooooohw..did I just say that?), I am ready to hang onto its native 24 bit without dithering until the bitter end, as per your suggestion.  (Thanks for explaining that 32 bit float isn't actually upsampling, btw).

However, I still am not entirely comfortable with the sample rate situation. I mean really, did they have to confuse this issue mathematically by having two standards that are in such an odd relationship to each other as 44.1 and 48?

Reason works at a sample rate of 24/96. My sound card's A/D converter works at 24/192. Now if I were downsampling from 192 or 96 to 48, I would be a lot more comfortable with the situation, because 48 is exactly half of 96, but isn't downsampling to 44.1 going to introduce more problems? Even if I export audio from Reason @ 88.2, (twice 44.1), then it has still gone through that odd numbered conversion in Reason.

So in the end I can't avoid this odd conversion, it has to happen somewhere..I have to choose between downsampling from Reason in the beginning, or waiting till later and doing it in Audition.

So should I follow a similar principle with the sample rate conversion, as with the bit depth?  Specifically, should I export audio from Reason in its native sample rate of 96, and then wait until I am ready to burn to cd, and then convert sample rate and bit depth at the same time?

[rharv]

The important bitrate happens as the audio is converted to digital .  This takes place inside the soundcard A/D/A convertors.  The software may tell the card what bitrate to use, but the actual function happens inside the card.

The software, during recording, is simply saying "OK I'm ready to recieve data at such and such a format" (which is why the software used to record isn't really as important as the soundcard - PCM wav audio data 'is what it is' once inside the computer)

Reason,  Audition, Powertracks, Reaper, whatever, the software is receiving what the soundcard sends for  a format.
After that anything done is mathematical. So if the soundcard sends 16/44 then there is no problem for the software, it just adapts.  If its 24/96 then it knows that and adapts... but the adaptations are not going to change the sound;  So adjusting the recording rate to work with software is defeating the real source- soundcard convertors.
That is where the rubber meets the road and the sound you get to work with is determined.
All the recording softwares actually handle the audio as 32 bit floating point for the math they do.  So saying the 'native rate' for a software makes no sense to me; it handles whatever the soundcard gives it.  Maybe Reason could be an exception because it actually 'generates' audio in some instances and this audio may be at a certain rate, dunno..
  Upconverting just adds zeroes that will be removed later. I can't see that making any improvement

I mentioned 24/48 because during tests people can't tell anything beyond that... that is the best rate-for-filesize because beyond that people just don't hear it.
 
There is a bigger difference between 16/44  and 24/44 than there is between 24/44 and 24/48.

The 16 to 24 bit depth is a big change, where as taking a sample 44,000 times compared to 48,000 times per second isn't that big a difference.  Its the fact that every one of those samples grabbed a larger chunk of sound in 24 bit, and THAT adds up.

[Wyatt]

The important bitrate happens as the audio is converted to digital .  This takes place inside the soundcard A/D/A convertors.  The software may tell the card what bitrate to use, but the actual function happens inside the card.

I have an EMU 1212m..so good convertors..

The software, during recording, is simply saying "OK I'm ready to recieve data at such and such a format" (which is why the software used to record isn't really as important as the soundcard - PCM wav audio data 'is what it is' once inside the computer)

  Maybe Reason could be an exception because it actually 'generates' audio in some instances and this audio may be at a certain rate, dunno..
  Upconverting just adds zeroes that will be removed later. I can't see that making any improvement

Reason doesn't record any audio..just midi. It can export midi tracks as audio, which is when I can select the bit depth and sample rate. The midi data from my keyboard does go through my soundcard. The .rex files I believe are already 24 bit, but I have no idea what the sample rate is on them.

I mentioned 24/48 because during tests people can't tell anything beyond that... that is the best rate-for-filesize because beyond that people just don't hear it.
 
There is a bigger difference between 16/44  and 24/44 than there is between 24/44 and 24/48.

The 16 to 24 bit depth is a big change, where as taking a sample 44,000 times compared to 48,000 times per second isn't that big a difference.  Its the fact that every one of those samples grabbed a larger chunk of sound in 24 bit, and THAT adds up.

So now that technology has outstripped human perception, as Oren also alluded to, there are things that really do matter and things that really don't.

Thank you for helping to sort all that out.

.. and thanks for your patience, I appreciate it.



Wyatt

[Alienz]

Maybe not quite about dithering but certainly interesting with regard to the 16/24 bit and mp3 quality question, i saw this video some time ago, its a long one though:

http://www.philoctetes.org/Past_Programs/Deep_Listening_Why_Audio_Quality_Matters

[Wyatt]

Maybe not quite about dithering but certainly interesting with regard to the 16/24 bit and mp3 quality question, i saw this video some time ago, its a long one though:

http://www.philoctetes.org/Past_Programs/Deep_Listening_Why_Audio_Quality_Matters

This is really long, but I am finding it very interesting. Kind of amazing to me how much better the first vinyl cut sounds than the CD redbook cut that precedes it.. Especially considering that I'm listening to a recording of this discussion on my computer. Must have been even more noticeable being in the room.

[Oren]

Maybe not quite about dithering but certainly interesting with regard to the 16/24 bit and mp3 quality question, i saw this video some time ago, its a long one though:

Finally somebody is making sense...

Over the last three years, I have looked at the information provided by various audio experts on bit-rate, sampling rate, dithering, digital processing, and compressed digital audio.They have provided a lot of numbers, but not much in the way of clarity.  It is astounding how frequently they contradict each other, and even themselves.

As with the participants in this in this discussion video, I have to conclude that the only real measure of an audio tool or technique is how positively it affects the listening experience.
      _________________________________________________________________________________________________________________

A group of audio experts I have had my eye on for some time (who actually do seem to make sense) recommend setting your DAW at maximum resolution (32bitfloat/96KHz), and outputting files at 24bits/96KHz. Then, for file sharing and general listening, compress the 24/96 file to a 256Kbit VBR file (either Ogg Vorbis or Lame VBR) to produce a small audio file that is superior in audio quality to a 16/44.1 wave file. I have not taken the time to work with any of these suggestions, but it feels like the time is right... This is one of their sites:
http://jthz.com/mp3/#TOP

They referred me to little article on microphone pre-amps.
 A $5.00 solid-state preamp proved to be the listeners' favourite in an internet test:
http://www.audiomasterclass.com/arc.cfm?a=well-could-you-hear-the-difference-could-you
     ________________________________________________________________________________________________________________

It might interest some of you that in all the time I have been mastering digital audio and encoding files to Ogg Vorbis and Lame, no-one has suggested I ask the folks who provided the original files if they had dithered in the process of producing the audio I was about to manipulate. Given our discussion in this thread, it would seem to be quite a critical bit of information...

[kara]

Well I'll be very honest about this....
I'm a musician, not an audio expert or even an audio engineer.
By this definition, I only take care how it sounds in a live environment.
I also do my best to post the best tracks I can here on the forum, but that's it.
The day I'll decide to make a commercial CD, I wont do it myself, I'll go in a studio and let the guys do there job 
And I'll try to play the best I can....

Does this sound naive ?

k

[Wyatt]

Well I'll be very honest about this....
I'm a musician, not an audio expert or even an audio engineer.
By this definition, I only take care how it sounds in a live environment.
I also do my best to post the best tracks I can here on the forum, but that's it.
The day I'll decide to make a commercial CD, I wont do it myself, I'll go in a studio and let the guys do there job 
And I'll try to play the best I can....

Does this sound naive ?

k

Sounds like a smart musician.

[Oren]

Another tidbit: When a 16/44.1 file is introduced into a DAW working with 16 bit internal processing, the workstation file measures the same size as the original wave file.
                       When a 16/44.1 file is introduced into a DAW working with 32 bit floating point processing, the workstation file measures twice the size of the original wave file. This suggests to me the file is fundamentally altered (up-converted?) as a track in your DAW.

[Wyatt]

Another tidbit: When a 16/44.1 file is introduced into a DAW working with 16 bit internal processing, the workstation file measures the same size as the original wave file.
                       When a 16/44.1 file is introduced into a DAW working with 32 bit floating point processing, the workstation file measures twice the size of the original wave file. This suggests to me the file is fundamentally altered (up-converted?) as a track in your DAW.

We may have come full circle, amigo, but it's been a hell of a trip!   

Thank you so much for filling in the blanks.



Wyatt