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Best bit depth and sample rate for non bit perfect playback.

post #1 of 3
Thread Starter 

Hi, what do you all think is the best bit depth and sample rate settings for Windows sound properties? I am not really interested in using any bit perfect playback like WASAPI or ASIO, as I am constantly switching what I am listening to i.e. music, movies, video games, web browser.

 

Currently I have it set to 24 bit, 48000Hz.

 

Also, does anyone know how the "supported formats" tab in the SPDIF out tab works? Is having a check mark next to any of the encoded formats or sample rates merely just for the purpose of testing a receiver? Does it do anything when multiple sample rates are selected?

post #2 of 3

Heya,

 

Frankly it doesn't matter what you set anything to. Whatever you play back is going to be played at whatever it was recorded/encoded at. Nothing more. Taking something's bit rate, and sample rate, and raising it up doesn't do anything. It just changes a number in a piece of software. But the sound is literally the same. The whole "upsampling" thing is a huge gimmick and pointless. If your music was recorded at 16 bits and 44.1khz, that's what it is. Taking it up to 48khz doesn't do anything at all. And converting it, even with re-encoding it, does zero. You can't add sound that isn't already there, other than noise. As for bit rate, specifically, that also is very subjective and based on the codec used. Not all codecs are equal. Some can produce the same level of quality of compression, as another, but at different bit rates. So again, this is pretty pointless to worry about. You should instead focus on simply having lossless or near lossless level compression, regardless of what container format you're using (MP3, FLAC, etc).

 

Very best,

post #3 of 3
Quote:
Originally Posted by MalVeauX View Post
 

Heya,

 

Frankly it doesn't matter what you set anything to. Whatever you play back is going to be played at whatever it was recorded/encoded at. Nothing more. Taking something's bit rate, and sample rate, and raising it up doesn't do anything. It just changes a number in a piece of software. But the sound is literally the same. The whole "upsampling" thing is a huge gimmick and pointless. If your music was recorded at 16 bits and 44.1khz, that's what it is. Taking it up to 48khz doesn't do anything at all. And converting it, even with re-encoding it, does zero. You can't add sound that isn't already there, other than noise. As for bit rate, specifically, that also is very subjective and based on the codec used. Not all codecs are equal. Some can produce the same level of quality of compression, as another, but at different bit rates. So again, this is pretty pointless to worry about. You should instead focus on simply having lossless or near lossless level compression, regardless of what container format you're using (MP3, FLAC, etc).

 

Very best,

Yes, upsampling is a gimmick, yet some DACs are optimized for hi-res playback and will measure better that way. At least I remember seeing such specs on some DAC. There shoudln't be a difference but there might still be a difference due to bad implementation.

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