I will try to explain intelligibly why listen to encode CDs to mp3 and listen to the flash playrf is not recommended.
Start with origins. Like audio CDs use digital coding, and can be safely athrabeth(a simple translate to a sound file format). Indeed you can, BUT! When he developed the format of an audio CD, it takes into account only the audio quality and not about any technology optimization(should read reduce) in modern deshovyh sound cards and player thought. Format audio CD predstavlaet a two-channel 16-bit representation of the sound 44100 times per second(calironia). I mean the sound is measured 44100 times per second and the value can be between 0 and 65535, and this for each channel. At this diskreditatsii possible transmission(in pretty bad quality) frequency 22050 Hz. The audio gate is amplitude modulated at a frequency of 19200. To understand what this means, draw on paper a big sinusoid, for example with a height of 10 cells and the distance of the nearest intersection with the zero in 10 cells, too, and draw after it the other Sinuiju, where a distance of zero-crossing in the 1st cell and the height of which is limited to the first sinusoidal. Roughly speaking the second sinusoidal recorded Audio story, but there is not measured in the cells, in the frequency 19200Гц. When the machine converts audiostrobe the signal for points, it works like a radio receiver, the only difference is that the radio receiver demodulator signals are much higher frequency, and supplies the headphones and not on points.
So now cut to the conversion of the track to a file, you might think that it is wrong, and this is a lot of controversy on the forms. But certainly the received WAV file vindus recognize like 1411 kilobits for one second. Where did this number? 16 bits multiplied by 44100 and 2, stereo as-Ni-as, and get 1411200, it seems not true? The double after it is washed down naturally removed. So to argue about? And I bet that is recorded in the file, if the disc is scratched, and a certain part can't be read. Speaking of stereo, originally this word was used only for sound tracks for films and meant not 2 channel as it was deshovoy audio equipment, and for 4, 5 or even more channels, but to save the tapes of these channels become mesatsa in two, and the place is a sly way residents. Now this format is called the standard home theater, and for us that means listening to a disc/file on a computer or expensive audio equipment with dolby icon will need to ensure that the decoding is not applied. Simple audio player and CD drive(if it's just connect the AC to the power supply and plug headphones directly into it) does not occur.
Now go to the mp3 standard. When it was developed, it was considered osoznannaya the ear's sensitivity, I mean the ordinary human ear can't hear above 18-19 kHz(thousands of cycles per second), as well as the fact that after a loud sound ear on opredelit shares of seconds stalls. It's all in the compression cut. However neosoznanna chuvstvitelnosti ear is much higher, ie in trance people can hear much above these limits, and in people with a musical ear, part of this range vyneseno consciousness... but should pay for everything, and for the small size to your favorite songs on mp3 player too. So under compression the frequency of the audio staroba and cropped. It is possible to compress without it, but you did not - I did not ask.
And now the intreresnoe - cost high quality. Sorry for audio CDs. Now the world runs to the high quality of the sound, where the sound atterolevad not 44100 times secure and 2 and three times faster. And it all started with 48000 times... Like it should be on our side, but no! Manufacturers of sound cards(and PlayNow, everything is the same, only the base is cheaper, but more energy more economical. Longer like to listen to plain some batteries have, don't worry... it's at your expense!) began to focus on higher sampling rate, and internal logic is running at a higher frequency(in this logic by the way include the decoder mnogokanalnosti, and even if it doesn't work, it does not mean that the signal it minamal, and resampleimage happens anyway). This means that the audio signal from one diskreditatsii(in this case 44100) becomes higher. Imagine that you sinusoide cut into equal-duration segments, and each piece was replaced by a similar rectangle of height is the fact that we have output citirama, so now we are trying to cut again, more parts, and the part where the junction pryamougolnogo we again averaged - this is the rough algorithm. There are more improved, and each manufacturer are different. Theoretically, everything goes fine, but in practical application there are rounding errors - database not c precision rubber, she has her limits. All this can be ignored for sound, but for audiolobe... as a result of such conversion, the frequency of audiostrobe starts to swim from a few Hz to a dozen, and the signal suffers... It can privesti to the fact that the decoder gate does not catch the signal for a few millisecond, Paulsen catches and loses. Anyway this leads to iskrevlenie waveform from nesnesitelna to quite tangible. But that's not all problems. As this digital technology, the speed of signal processing depends on the delays between logic elements, therefore, trying to arrange them as close as possible that reduces the delay prohozhdenia signal through the wires, and here comes the second problem - the inter-penetration of the signal from different channels. Then the wires begin to work as priming and transmitter in both directions. Theoretically, the lower the frequency the less the effect. But due to minituarization it becomes Aktualno and audiostrobe and sound. Audio stobom clear. But with sound on it faces problems with binarnymi a beat - must be the frequency difference between the left and right ear - now both frequencies in different proprecia located to each ear...
A similar problem occurs in the built-in audio card, only then will penetrate the noises of the CPU, video card, jestokogo drive. This is usually quietly on laptops, powerful computers this is true even for the leaders in the production of motherboards(I have a solid motherboard from ASUS,I'd say a definite leader - this problem is present.)
Well, the opposite, that by the way true for a simple analog technique - sound to the headphone out is pre-amplification, and amplified in different frequency ranges differently. This is called the lowering of the amplitude-frequency characteristics. That is, very simply, the cheaper the outfit the sensitivity of the amplifier nisaka, which leads to mild bass boost and high frequency components of the sound. Which again has mnogostradayuschy audio strobe - he is set at high. This problem can sometimes somehow be solved by twisting the volume on full, and after umanista volume directly on the headphones, but not all headphones have volume control:(
So if hotit to listen to on your computer buy a good expensive sound card. It is advisable to ask a friend programmer to do the research - it is necessary that resampleimage occurred. It is possible immediately to cut off cheap SBLive and Audigy version. And every plashechnye players - there's even a deep research not all will say, check on acceptability of performance a. WAV file. And mp3... if player plays normal WAV files - and you can experiment, but with the current sizes of USB drives - why? In principle, the main problems could have been avoided encoding audiostrobe differently, or even relegate it beyond audio, but what used to be a smart solution, now is the guarantor of monopoly.
PS: possible missing words and left blots - when writing, I have a cat sleeping near the keyboard, often kicking his legs and popada the mouse, then the keys.
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We are not talking about the sound frequency and binaural beats etc. About the method a lot of articles online...ek, only to blurt out 