DAC Bit rate Format Important?

[GiantMidgt]

So recently i Have a Nuforce DAC Icon 2, Running on windows 8 computer. I was going through the nuforce setup forums and it told me to go into
Playbacks Devices ->
 
And then Properties -> Advanced -> Select 24bit 96khz. 

So i went to do exactly that, But i do not have that option available. So i went to check the DAC Specifications and it clearly advertises it supports those Sample Rate and Bit Depths. 
The questions i have are:
1. Is there a difference between 16bit 48000 hz, and 24bit 960000 hz. if so how will i go around to fixing this problem?
2. If i have downloaded music at 320kbps will this throttle the quality of the files? or are these speaker settings only for windows sounds. 
3. What is the correlation between 16bit, 48000 hz / 24bit 96000 hz and 320kbps music?
 

 

[PleasantSounds]

Your 320kbps mp3s are at best equivalent to 16 bit 44100 Hz. 
Not sure why 24/96 option is not showing, but don't stress about it too much.

 

[]eep]

1. Is there a difference between 16bit 48000 hz, and 24bit 960000 hz. if so how will i go around to fixing this problem?
Yes there is. But if the improvement in bitrate results in improved sound depends on the quality of your other equipment and on the quality of your source material. With MP3's (or any LQ lossy format) the question is moot. And you need good speakers/HP to hear the difference. Any mediocre equipment puts you under a glass ceiling. It takes a revealing system to reveal the difference. The µ-dac is not really in this category. So...

2. If i have downloaded music at 320kbps will this throttle the quality of the files? or are these speaker settings only for windows sounds.
Yes. Use >24/96 flac (or cmprb) to try. 24 or 16 bit doesn't really matter but there are hardly any 16/96 files. 24/88 (from SACD) is fine too. If you were to use Foobar you can use the foobar settings to bypass windows and send directly to the Nuforce (at > 48kHz).

3. What is the correlation between 16bit, 48000 hz / 24bit 96000 hz and 320kbps music?
MP3 is castrated anyway. When you grow up (soundwise that is!) you will not be satisfied with less than - lossless- cd quality. 24/96 is better and makes life easier for your dac (to put it very simple). But the difference is and will be subtle at best (absolutely speaking). Relatively speaking it will be a big difference, like when upgrading from a $5000 system to a $50.000 system. But that is on the slippery logarithmic scale of 'high-end' diminishing returns. And the µ-dac is not.

I have a suspicion that your windows does not recognize the inputchip on the Nuforce. This will limit your options to the windows standard 16/48. On the the 'playback' panel it should identify as 'Nuforce DAC Icon 2'.

 

[GiantMidgt]

Hi there, thank you for the very informative reply. It has cleared up many lose ends with the whole situation. I have a feeling the DAC i have may be faulty. Iv tried it on 2 other computers and the max settings i can do is 16 /48khz, it is only recognized as USB Audio DAC aswell. 

Do you know of any ways to make windows recognize the inputchip? or is this a fault with my DAC its self. 

 

[markbrauer]

Warning - I often seem to know just enough to be dangerous...
 
It sounds like a driver problem to me. NuForce says the DAC works with Windows 8 standard ASIO4ALL drivers. 
https://www.nuforce.com/index.php?option=com_k2&view=item&id=59:wheniconnectthenuforceicontomycomputerigetanerrormessagestatingthatthenecessarydriverisnotfounddoyousupplyadriverwithyourproductsandifsohowcanigetone&Itemid=1107
As JEEP says "I have a suspicion that your windows does not recognize the inputchip on the Nuforce. This will limit your options to the windows standard 16/48. On the the 'playback' panel it should identify as 'Nuforce DAC Icon 2'." It is the driver that does the recognizing. Yours may have been changed/deleted or somesuch. Possibly installing another device changed it. I don't know Windows 8 but there should be a settings panel somewhere that identifies the sound driver and gives the opportunity to check that it's current.
 
The big problem with the wrong driver, besides not supporting higher resolutions, is that it probably doesn't support "bit perfect" streaming. That means everything goes through Windows' sound mixer before being sent to the DAC. This means all sounds the computer makes - beeps, little tunes, warnings - get sent along with your music. Annoying. But worse, to mix all these sounds Windows re-digitizes everything - even your music - much to the detriment of audio quality. To get the most from your DAC, even at lower resolutions, you MUST have a driver that supports bit-perfect streaming. Based on the panel images you posted, I don't think that's the case on your PC.
 
Also, if the computer has been "upgraded" to Windows 8, sometimes older hardware doesn't support everything. I have a Windows 7 machine that started life as Windows XP. The SD card reader has not worked since the upgrade. Can't find a compatible driver.  
 
Good luck.

 

[Music Alchemist]

There is no audible difference between the Red Book standard (16-bit / 44.1 kHz, aka "CD quality") and "HD" audio with higher resolution.
 
The only difference you would ever hear is if you are listening to two different masters of a recording or if your equipment has trouble playing certain formats.
 
Read this article for a technical explanation.
 
Many people cannot hear a difference between lossless and 256 kbps AAC. If you have plenty of hard drive space, just use lossless if possible.

 

[RonaldDumsfeld]

PleasantSounds & Music Alchemist offer good advice.
 
I'm well confused about that link posted from the uForce website.
 

 
All NuForce products designed to avoid the need for any type of additional driver installation. Our products operate using the industry standard ASIO4ALL USB audio driver supplied with virtually all modern operating systems (including Windows, Mac & Linux).

 
Is that true? Anyone? My understanding has always been that ASIO4ALL is a personal project from Michael Tippach. He wanted an ASIO driver for himself and figured he might as well give it away to other people as a download as an act of goodwill. 
 
This is the website
 
http://www.asio4all.com/ 
 
So if I'm getting this right (I hope not) uForce is selling a DAC that needs someone else's free software to work properly. Shurely shomething wrong here Offisher?

 

[Roseval]

Microsoft, Apple and all Linux distro's standard with ASO4All?
Looks like total nonsense to me.
Maybe they mean UAC1?

 

[dprimary]

   
So i went to do exactly that, But i do not have that option available. So i went to check the DAC Specifications and it clearly advertises it supports those Sample Rate and Bit Depths. 
The questions i have are:
1. Is there a difference between 16bit 48000 hz, and 24bit 960000 hz. if so how will i go around to fixing this problem?
2. If i have downloaded music at 320kbps will this throttle the quality of the files? or are these speaker settings only for windows sounds. 
3. What is the correlation between 16bit, 48000 hz / 24bit 96000 hz and 320kbps music?
 

MP3 can only go to 48KHz,  AAC can go to 96k,  your bit rate is not really related to the sample rates and bit depth. If I took a 22k 12 bit file and encoded at 320 bit rate, the amount of data compression required would be minimal. Most music is released at 44.1 sample rate, for most people if you look at their music library the bulk of it will be at the 44.1 sample rate.
 
Changing the sample rate from what it was recorded at can only degrade it. These days sample rate conversation losses are very minimal and often inaudible compared to 20 years ago when you avoided it at all costs.
 
This is where 96K could hurt you if you can configure the encoder to preserve the 96K bandwidth, would would actually be reducing your available data rate for your audible range, decreasing the sound quality. Many encoders would low pass before encoding to prevent that. Encoders used in mastering might be flexible enough to really screw things up in the wrong hands.

 

[Music Alchemist]

  MP3 can only go to 48KHz,  AAC can go to 96k,  your bit rate is not really related to the sample rates and bit depth. If I took a 22k 12 bit file and encoded at 320 bit rate, the amount of data compression required would be minimal. Most music is released at 44.1 sample rate, for most people if you look at their music library the bulk of it will be at the 44.1 sample rate.
 
Changing the sample rate from what it was recorded at can only degrade it. These days sample rate conversation losses are very minimal and often inaudible compared to 20 years ago when you avoided it at all costs.
 
This is where 96K could hurt you if you can configure the encoder to preserve the 96K bandwidth, would would actually be reducing your available data rate for your audible range, decreasing the sound quality. Many encoders would low pass before encoding to prevent that. Encoders used in mastering might be flexible enough to really screw things up in the wrong hands.

 
Human hearing has not changed. 16-bit / 44.1 kHz is the absolute highest anyone will ever benefit from during playback. Using different settings (higher or lower) usually won't change the sound at all. For example, if you convert a 16-bit / 44.1 kHz file to 24-bit / 96 kHz (or the reverse, for that matter), it sounds the same. If you configure a DAC or player to do 24-bit, it can usually still play everything else as well, and everything will still sound the same. The only exceptions I can think of offhand are 1) when certain devices, programs, settings, etc. refuse to play the file at all, 2) when a system is unable to properly play various resolutions, producing distortion in the process, and 3) when a DAC upsamples audio in a way that is basically just equalizing it.

 

[dprimary]

   
Human hearing has not changed. 16-bit / 44.1 kHz is the absolute highest anyone will ever benefit from during playback. Using different settings (higher or lower) usually won't change the sound at all. For example, if you convert a 16-bit / 44.1 kHz file to 24-bit / 96 kHz (or the reverse, for that matter), it sounds the same. If you configure a DAC or player to do 24-bit, it can usually still play everything else as well, and everything will still sound the same. The only exceptions I can think of offhand are 1) when certain devices, programs, settings, etc. refuse to play the file at all, 2) when a system is unable to properly play various resolutions, producing distortion in the process, and 3) when a DAC upsamples audio in a way that is basically just equalizing it.

I for the most part agree with what you a saying, which is why I pointed out that you would not want to encode an AAC at 96K since you would be wasting bandwidth of the 320 encoding for things you cannot hear making the audible part of the encoding more like a 160 or 192 bit rate, which is more audible. 
 
From the production prospective is where I disagree, if you take a 44.1/16 file and start mixing it, it could go though thousands to millions of calculations and it does start to degrade which is why when it became feasible  the industry moved to 24 bit. 20bit is just about all the resolution that physics allows, so already you have about 20 db of resolution to use up in rounding before it could start to degrade. It is not something music listeners should ever encounter unless for some reason you want to listen to recordings through music production software and run it through dozens of processing plug-ins.
 
In production enviroments wide bandwidth systems have been around for a long time, hopefully most people in production understand the tradeoffs and pitfalls of having signals you can't hear. Having a high level ultrasonic signal that you you don't know about tends to let the factory installed smoke out of equipment. If it happens in a high end studio or on a multi-million dollar touring system you won't have a job any longer. Modern touring systems run at 96KHz not because anyone believes you can hear the difference you cannot even get 10KHz across a stadium the air absorption of the high frequencies is too great. It is because it reduces the DSP latency. Which brings up how does people think 30KHz travels in the air?  High frequencies drop fast in air.onger. Modern touring systems run at 96KHz not because anyone believes you can hear the difference you cannot even get 10KHz across a stadium the air absorption of the high frequencies is too great. It is because it reduces the DSP latency. Which brings up how does people think 30KHz travels in the air?  High frequencies drop fast in air

 

[Music Alchemist]

  I for the most part agree with what you a saying, which is why I pointed out that you would not want to encode an AAC at 96K since you would be wasting bandwidth of the 320 encoding for things you cannot hear making the audible part of the encoding more like a 160 or 192 bit rate, which is more audible. 
 
From the production prospective is where I disagree, if you take a 44.1/16 file and start mixing it, it could go though thousands to millions of calculations and it does start to degrade which is why when it became feasible  the industry moved to 24 bit. 20bit is just about all the resolution that physics allows, so already you have about 20 db of resolution to use up in rounding before it could start to degrade. It is not something music listeners should ever encounter unless for some reason you want to listen to recordings through music production software and run it through dozens of processing plug-ins.
 
In production enviroments wide bandwidth systems have been around for a long time, hopefully most people in production understand the tradeoffs and pitfalls of having signals you can't hear. Having a high level ultrasonic signal that you you don't know about tends to let the factory installed smoke out of equipment. If it happens in a high end studio or on a multi-million dollar touring system you won't have a job any longer. Modern touring systems run at 96KHz not because anyone believes you can hear the difference you cannot even get 10KHz across a stadium the air absorption of the high frequencies is too great. It is because it reduces the DSP latency. Which brings up how does people think 30KHz travels in the air?  High frequencies drop fast in air.onger. Modern touring systems run at 96KHz not because anyone believes you can hear the difference you cannot even get 10KHz across a stadium the air absorption of the high frequencies is too great. It is because it reduces the DSP latency. Which brings up how does people think 30KHz travels in the air?  High frequencies drop fast in air

 
But the context of this is audio playback, not production. It makes sense to have higher resolution in the studio due to higher demands of computer processing.
 
How would converting a 320 kbps AAC file to 24-bit / 96 kHz suddenly make it audibly equivalent to a 160 kbps file? (Not that it would be likely to hear the difference between those two bit rates either way.) It should be of the same fidelity as the original 320 kbps file. And as far as I know, you cannot change the sample size (aka bit depth) and sample rate of an AAC file in the first place; you would have to convert to lossless to do that.

 

[dprimary]

   
But the context of this is audio playback, not production. It makes sense to have higher resolution in the studio due to higher demands of computer processing.
 
How would converting a 320 kbps AAC file to 24-bit / 96 kHz suddenly make it audibly equivalent to a 160 kbps file? (Not that it would be likely to hear the difference between those two bit rates either way.) It should be of the same fidelity as the original 320 kbps file. And as far as I know, you cannot change the sample size (aka bit depth) and sample rate of an AAC file in the first place; you would have to convert to lossless to do that.

You are taking a 96K/24bit PCM file and encoding it. You have a set data rate,if you do not low pass filter out the ultrasonics you will be encoding the ultrasonics.  This will reduce the available data rate you can store from the audible range since you can only. Just for example if 20k to 48k ultrasonic audio is encoded lets say it uses 160K of the 320K you only have 160K left to store the the audible 20Hz to 20K audio. Of course it would not really be a 50/50 reduction. Most encoders automatically low pass the audio to reduce the amount of data to be encoded. Encoders at low bit rates low pass pretty far in to the audible range. Most of the encoders seem to be configured to prevent this from happening. It does look you you could force it in command line.

 

[Music Alchemist]

  You are taking a 96K/24bit PCM file and encoding it. You have a set data rate,if you do not low pass filter out the ultrasonics you will be encoding the ultrasonics.  This will reduce the available data rate you can store from the audible range since you can only. Just for example if 20k to 48k ultrasonic audio is encoded lets say it uses 160K of the 320K you only have 160K left to store the the audible 20Hz to 20K audio. Of course it would not really be a 50/50 reduction. Most encoders automatically low pass the audio to reduce the amount of data to be encoded. Encoders at low bit rates low pass pretty far in to the audible range. Most of the encoders seem to be configured to prevent this from happening. It does look you you could force it in command line.

 
I have no idea what you're talking about. If you convert a lossy file to any lossless format, it just fills it with empty space. It does not change the audio at all. If you convert any lossless file to 256 kbps AAC, it's highly unlikely to distinguish between them either.

 

[dprimary]

   
I have no idea what you're talking about. If you convert a lossy file to any lossless format, it just fills it with empty space. It does not change the audio at all. If you convert any lossless file to 256 kbps AAC, it's highly unlikely to distinguish between them either.

other way lossless to lossy