Digital Audio Myths - Listening on a PC

goose wrote: oliverstoned wrote: You can't prove it "scientifically". It's just like proving that your favourite prog band is better than the last Britney spear record. But when you hear a good tube amp versus a transistor, there's no need to explain. There's only music and emotion. You've said before it doesn't matter if people prefer digital to analogue because analogue is better. But all you can prove by listening is that you prefer the sound of analogue, not that it's any more accurate than digital.     Re the "cutting off" of sounds. There are only two losses that occur during processing from analogue to CD-audio (PCM)   Firstly the cutting off of any frequencies above 22.05kHz. This is basically irrelevant from a musical point of view - although these frequencies can affect people, they're inaudible.   Secondly the rounding up or down of every sample to one of 65,536 values. I can't prove that that isn't audible (quite possibly it is, I'll work on test files within the next couple of days) but consider that DVD-A is capable of 16,777,216 values and up to 48kHz. If any other medium sounds particularly different to this, then it must be less accurate. Maybe that sounds nicer to some peoples' tastes, but science really is real.

Now if you look at schematics over the amplitude of transistors when they distort you will see that the amplitude turns alot sharper than the amplitude over the tubes. Just like in digital signals the signals can only be either 1 or 0 there is not a smooth transition between these two so you can say really about the transistors they are either not distorted or distorted it is no smooth transiton between these two

MikeEnRegalia wrote: maidenrulez wrote: MikeEnRegalia wrote: maidenrulez wrote: Wich basically means that if we record speech wich got a bandwith of about 3khz then you would need a samplingfrequency of 6khz Exactly. The reason why modern soundcards (and SACD) use sampling frequencies up to 192khz is that when you process digital sound (add reverb, eq ... anything) the rasterisation used during A/D conversion can cause artefacts ... nasty enharmonic overtones. So you need a large safety margin, although twice the frequency is enough if the signal is just played back. Yup NOMATTER how good your sound cards there is really a huge limitation in using the PCI-bus because it has great limitiations when it comes to sound.  What does the bus have to do with it?

Infact CREATIVE wich is a huge manufacturer of sound card admitted that there where no point in making their sound card's better until they have gotten a better BUS to fit it... Actually they are working for a special bus only for sound-cards that should be optimal

maidenrulez wrote: MikeEnRegalia wrote: maidenrulez wrote: MikeEnRegalia wrote: maidenrulez wrote: Wich basically means that if we record speech wich got a bandwith of about 3khz then you would need a samplingfrequency of 6khz Exactly. The reason why modern soundcards (and SACD) use sampling frequencies up to 192khz is that when you process digital sound (add reverb, eq ... anything) the rasterisation used during A/D conversion can cause artefacts ... nasty enharmonic overtones. So you need a large safety margin, although twice the frequency is enough if the signal is just played back. Yup NOMATTER how good your sound cards there is really a huge limitation in using the PCI-bus because it has great limitiations when it comes to sound.  What does the bus have to do with it? Infact CREATIVE wich is a huge manufacturer of sound card admitted that there where no point in making their sound card's better until they have gotten a better BUS to fit it... Actually they are working for a special bus only for sound-cards that should be optimal

The bus doesn't affect digital playback in any way. Of course if they want to transmit 7.1 signals in 24bit/192khz you'd need PCIe ...

maidenrulez wrote: MikeEnRegalia wrote: maidenrulez wrote: Wich basically means that if we record speech wich got a bandwith of about 3khz then you would need a samplingfrequency of 6khz Exactly. The reason why modern soundcards (and SACD) use sampling frequencies up to 192khz is that when you process digital sound (add reverb, eq ... anything) the rasterisation used during A/D conversion can cause artefacts ... nasty enharmonic overtones. So you need a large safety margin, although twice the frequency is enough if the signal is just played back. Yup NOMATTER how good your sound cards there is really a huge limitation in using the PCI-bus because it has great limitiations when it comes to sound.

 What does the bus have to do with it?

MikeEnRegalia wrote: maidenrulez wrote: Wich basically means that if we record speech wich got a bandwith of about 3khz then you would need a samplingfrequency of 6khz Exactly. The reason why modern soundcards (and SACD) use sampling frequencies up to 192khz is that when you process digital sound (add reverb, eq ... anything) the rasterisation used during A/D conversion can cause artefacts ... nasty enharmonic overtones. So you need a large safety margin, although twice the frequency is enough if the signal is just played back.

Yup NOMATTER how good your sound cards there is really a huge limitation in using the PCI-bus because it has great limitiations when it comes to sound.

i might have been a bit bad explaining it but yes the samplingfrequency should be double the frequency of the original signal as goes for the forumla:

Samplingfrequncy: f_s=1/T_s

Now lets take on the battle between cd's and vinyls

Now a standard cd is sampled in 16bits and 44,1khz normally a human would just hear sound in the spectre of 18hz to 22,05khz wich means that esentially the sampling frequency has been doubled to ensure that the sound signal sampled would be more accurate to the original singal.

Now for the analog singals to be converted to digital signals a number of spaces to make the sample now for 16bits there would be 65535 spaces to place and then there will be lines aligned between them. Now if you tried to made a normal sinus signal this way you would nearly get the round form wich appears on the bottom and peak values of its amplitude. HOWEVER it will never be as good as the original analog signal because it would not be completly round and therebefore sound a bit sharp yet.

say cobb have you ever heard a full blown high end hi-fi system?

it just amazing to listen to classical music with big sympony orchestras on some proper hi-fi systems it is almost as the symphony orchestra is right there in the room