sunnyimran :
So I got the idea after owning a entry level 5.1 PC audio , that 90% of items to play specially from internet are not multi channel.
so for to time to come , we need some good players / hardware configurations to at least enjoy plain stereo on multi channels although not true multi channel surround, but still better than plain stereo.
I searched several of my favorite audio tracks, little old, could not find them in DTS or other surround.
This all has grown my interest in digital audio. please specify some good links to get more insight of digital audio.
if you want the quick rundown, higher sample rates and higher bit depths will only help improve the sound quality results.
but
if you are wanting more information than that, you would probably want to dive into the realm of sound processing effects.
i have went through the task of learning foobar2000 and why it is better to use kernel streaming or asio for output, and it boils down to a more direct approach to passing the data along to the soundcard.
the sound processors work pretty much like this,
they review the audio data and see the low/medium/high sounds.. then they make the low sounds lower and the high sounds higher, to create a more dynamic amplitude of sound.
they can also look at the soundwaves and see some of the gaps provided by the low sample rate, then try to fill in those gaps with information using simple algebra.
the same can be said for the bit depth, but this is more advanced since there is a lot more guessing work.
i dont know if audio uses more than one type of programming language.
but if it does exist..
then a sound processor would take the junk programming language and translate it to a higher quality one.
for example.. if your digital to analog chip reads 1's and 0's .. maybe it sounds better if you send *'s and .'s
different symbols might change the voltage across the chip, and if you are saving or feeding more voltage.. it might produce a better audible result.
since there is a compiler that acts like a dictionary for all of the audio data, some symbols and characters would have a higher noise floor to represent a recording of higher signal to noise ratio.
this means you could record with a higher noise floor, and then translate the data to lower the noise floor and fill in the gap with artificial digital silence.
it is like you are inside your house and you can hear your PC fan.. and if you wanted to raise the amplitude of the noise from the fan, you could do that by changing the characters and symbols used to represent the soundwave from the fan.
your noise floor is going to be what captures the sounds of vehicles driving by outside of the house.
if you manually adjust the soundwave of the fan, you can make the fan louder and the driving vehicles stay where they are.
this would seem like the fan is closer to the microphone, but you cant fool the time domain.. so the result would be artificial.
but here is why they are supposed to use it.
if you turn off the PC and your room has an ambient noise of like 50dB .. then when you turn the PC back on and record the fan noise, you could seperate the fan from the 50dB of room noise ... or you could remove the room noise altogether.
kinda like recording yourself talk and removing the noise from the PC fan in the background.
people will hear you without the fan noise.. and that means the listener can focus on your voice and nothing else.
this is all easier said than done.. because it takes training and team work with the software programming to accomplish something.
or
buy a piece of software already created that does it.
have you ever come across a 'remove hiss' filter?
sometimes the hiss noise is viewed and the frequency is captured, then it applies a filter for that frequency to remove the hiss.
when it is done, you can hear the rest of the audio has changed a little bit from the removal of the frequency IF the frequency was in the same area of whatever you recorded.
another way to do it is to physically find all of the data that represents the hiss, and the hiss only.
then simply delete it or fill in the character's and symbols with digital silence.
it is like working with metal.
if there is rust somewhere, you can grind the rust off, or you can simply make a straight cut with a saw and build a new piece with a flat edge on the top.. then weld the new piece on and sand down the welds to make it all flat on the front side.
if the new piece is the perfect shape, you wont know the piece has been replaced when it is all primed and painted.
there are probably hundreds of audio effects 'plugins'
i dont think i have seen 100 plugins yet, because i havent browsed 'em all.
but
there are dozens and dozens of them.
some plugins cost like $100 for one.
and some plugins come with those studio mastering programs that allow you to manipulate a recording.
these programs can cost thousands of dollars !!