Bandwidth Place 2

With MP3 if you try to convert a 320 Kbps MP3 to a 128 Kbps MP3 you will loose even more quality because you are taking a lossy codec and converting it to a lossy codec, and even if you convert an MP3 to a lossless codec there's no point because you've already lost information from the track.

So if you want to convert an MP3 you should burn it to a CD then rip it.

So that's why I want to make my files lossless, then I can make whatever I want from them and it will sound good because I would be using all the information the original has.

palmboy05 wrote:why dont u ask more of those audio guys?

Antix wrote: So if you want to convert an MP3 you should burn it to a CD then rip it.

Q12321 wrote: No. You will actually lose more quality to be honest, because you are taking a compressed file, trying to uncompress it to WAV (which is close to uncompressed), then compressing it again. That's why "when you do a direct conversion from MP3 to say WAV you some how lose more in the conversion."

2005 wrote:no you cannot, if you rip it to 200 kbps then thats the highest quality you can get from that source, you cant rip peices off a piece of paper and throw em away and then try to put em back.

"As we see, by simple analog-digital conversion of analog sound material with high sample rate and quantization resolution, it is possible to store audio material in computer memory without almost any loss of quality. Then a question is raised: what for is this great amount of various audio compression technologies (like MP3 and other)?

There are various reasons for that. As a matter of fact, the wish to keep original quality of audio materials after its conversion from analog into digital form, encounters definite difficulties. According to Nikewist (Kotelnikov) theorem, sample rate sets upper bound of frequencies in the signal, namely, maximum frequency of spectral components of digital signal is equal to half of its sampling rate. Bluntly speaking, to obtain full spectral image of original analog signal in frequency range 0 - 22050 Hz (the maximal range of perceptible frequencies for human) it is necessary to choose sample rate to be no less than 44.1 kHz. It means that the wish to keep original quality of audio material obliges us to choose high parameters values of analog-digital conversion. However, the higher the values are the greater amount of memory is needed to store digital data. For example, standard audio CD (650 Mb) stores audio data in format PCM 44.1 kHz / 16 bits / stereo. Such parameters correspond to two-channel record with 65536 (216) quantization levels of amplitude, which values are taken 44100 times per second. Making very simple calculation, we ascertain that standard audio CD contains about one hour of music. Basically, it is not such a big value, taking into account that medium audio collection may be thousands hours long. It is necessary to notice here, that standard file type for storing digital audio today is .WAV file. It is a universal container, which allows storing digital audio with different sampling rates and quantization resolutions.

So, as we see, to have an opportunity to store great amounts of audio at high quality it is necessary to resort to various tricks which help in storing of audio data occupying less memory. The tricks we're talking about are compression (coding) methods which gain in data size at the expense of some loosing of original sound quality. About these tricks we are going to talk now.

There are two prevalent ways of coding of audio data (except for simple storage in pure digital form "as is" which was shown above).

Footnote:
* Data coding - representation of the data in a certain system of coding symbols and their structures. Enciphering, and also compression of the data are special cases of coding.
* Under data stream we mean contents of a file; data downloaded from the Internet or any other consecutive digital data.

1. Lossless data compression is a way of audio data encoding which allows total-lot data restoration from compressed form back to the original stream. Such way of data compression is used when it is necessary to store data without any loss of quality. For example, working with audio in sound studio, after data was recorded, it needs to be stored in sound archive for the following processing or publishing. Today's lossless data compression methods (like Monkey's Audio, Flac, WavPack, TTA, OptimFrog, etc) allow reducing data size by 20-50% along with providing hundred-per-cent restoration of the original data from compressed stream. Such coders are some kind of data archivers (as, for example, ZIP, RAR and others), but intended specially for compression of audio data.

Footnote:
* Coder is a program (or hardware device) implementing certain algorithm of data coding (for example, ZIP archiver or MP3 encoder). It transforms initial data stream in its source format into certain encoded (compressed) form (format).
* Decoder is a program (or hardware device) which implements decoding of the encoded data.
* Codec (COder/DECoder) is a program / software driver / hardware device intended for data coding / decoding.

Lossless compression is ideal in respect to source data safety, but it is unable to provide with high compression.

2. There is also another way of audio compression - compression with loss of quality (so called, lossy coding). The aim of such coding: in any way to achieve sounding similarity of compressed data with original audio material, along with maximal profit in size. This aim is reached today by using various algorithms which "simplify" an original audio signal, throwing out from it "unnecessary" almost inaudible (or indiscernible by a human ear) details. After lossy coding, decoded signal sounds similar to the original material, but actually ceases to be identical to it. There are several lossy codecs available. The most known are: MPEG-1 Layer 3 (it is the correct official name of the well known "MP3"), MPEG-2/4 AAC (MPEG-2/4 Advanced Audio Coding), Ogg Vorbis ("OGG" in abbreviated form), Windows Media Audio (WMA), MusePaсk (MPC) and others. The benefit of using such audio compression methods is quite obvious: compression factor provided by such coders is on average within the limits of 7-14 (times) and these results reached at almost undistinguished losses of sounding quality. Practically it means that one regular audio CD track copied to hard-disk as a .WAV-file is about 35-55 Mb long (PCM 44.1 kHz / 16 bit / stereo), but being compressed into MPEG-1 Layer 3 (MP3) it becomes 3-7 Mb long, and all this with more than satisfying sound quality.

As we have said, compression of data in lossy-coders is accomplished by simplification of audio data. The main idea of almost all lossy encoders is based on using of so-called psycho acoustic algorithms of sound analysis (psycho acoustic model), which is used to "simplify" audio data during compression. Mechanism of coder, which is based on simplification of signal spectrum (there are also coders, which are based on other techniques) works approximately as follows. The encoder analyses the signal during compression, determining frequency spectrum areas with inaudible to a human ear nuances and moving them off. These nuances are masked or poorly heard frequencies, short-term inaudible bursts, low-level noises and so on. Such processing simplifies the form of an original sound wave, making it "smoother". Compression factor of the original signal depends on a degree of its "simplification"; good compression is achieved by "aggressive simplification" when the coder considers as insignificant a great many nuances. I.e. the stronger simplification, the better compression is achieved. Too aggressive compression, naturally, results in strong degradation of quality, because many of audible details could be considered as insignificant and maybe deleted by the coder. Distinctive feature of all modern lossy-coders is the opportunity of fine adjustment of coding parameters. This feature, combined with competent approach and correct understanding of compression methods, allows achieving high data compression factors at completely imperceptible original quality loss.

Now let's talk a bit deeper about the method of signal simplification (once again, on the example of encoder which is based on spectrum simplification techniques). The mechanism of audio signal simplification can be explained as follows. Initial audio data is divided on blocks of the certain length. After division, each block is processed separately. During coding, each block is decomposed into frequency spectrum. As we have said, the less nuances the signal has (i.e., the more "simple" the signal is, or in other words, the less frequency components its spectrum contains), the more effective compression can be reached. There are some possibilities to simplify the signal. For example, it is possible to filter (to exclude from the spectrum) all frequency components above some boundary. This will null the signal in high bandwidth, improving compression, but this also will have bad influence on sounding quality. However, main way of simplification is applying of psycho acoustic model: coder analyses the signal considering which spectrum components are unnecessary or inaudible, and excludes them from the spectrum.

Footnote:
* Bitrate - amount of bits used for storing of one second of audio. For a standard .WAV-file in format PCM 44.1 KHz / 16 bit / stereo the bitrate is: 44100 (amplitude values per second) * 16 (bits per one amplitude value) * 2 (channels) = 1411200 bits per second = about 1378 Kbps (Kilobit per second).

* Usually user can specify preferred bitrate (as good as other parameters) or bitrates range before compression. The lower bitrate is, the fewer bits the coder may allocate for storage of one second of audio data and, thus, the deeper simplification the signal passes during compression (that accordingly influences quality of sounding). The most prevalent average bitrate value for MP3's downloaded from the Internet changes in the range of 128 - 192 Kbps.

It is necessary to notice especially, that using of psycho acoustics as sound simplification method during compression leads to impossibility of getting exactly the same signal as source signal on decoding. This is because of irreversible changes of original data during compression. That is why such compression method is called "lossy coding". So, every time when you use lossy-coders you should bear in mind this aspect. For instance, if you compress music for your audio collection, you should not limit the bitrate too much, because in doing so you might mar the quality of resulting audio stream. On the other hand, with competent approach, as a compression result you may get very good compression factor (as is the purpose of coding) plus high sounding quality."