RE: compressing mp3 files
- From: marcinont <marcinont@xxxxxxxxxxxxxxxxxxxxxxxxx>
- Date: Wed, 21 Dec 2005 10:41:16 -0800
What?
"Bhavesh" wrote:
> MPEG standards in general and MP3 in particular
>
> MPEG-1 Layer 3 (known as "MP3") is most widespread and popular today. It has
> won its popularity quite deservedly - it is the first widespread lossy-codec
> which reached such a high data compression factor, together with very good
> sounding quality. A little bit of history. MPEG is an abbreviation of "Moving
> Pictures Coding Experts Group". MPEG has been started at January, 1988. Since
> the first assembly in May, 1988, the group began to grow, and has grown up to
> unusual dense experts collectively. Usually, in MPEG assembly about 350
> experts participate, from more than 200 companies. The largest part of
> participants are the experts occupied in various scientific and academic
> establishments. Today MPEG group has developed the following standards and
> algorithms:
>
> MPEG-1 (November 1992) - the standard of coding, storage and decoding of
> moving pictures and audio data;
> MPEG-2 (November 1994) - the standard of data coding for digital TV;
> MPEG-4 - the standard for multimedia applications;
> MPEG-7 - universal standard for multimedia, intended for processing,
> filtration and management of multimedia data.
> Let us consider the set of standards MPEG-1. This set, according to ISO
> standards (International Standards Organization), includes three algorithms
> of different levels of complexity: Layer 1, Layer 2 and Layer 3. Our well
> known friend MP3 in exact designation is "MPEG-1 Layer 3". The general
> structure of encoding process is identical in all Layers. At the same time,
> in spite of similarity of the Layers in the general approach to encoding, the
> Layers differ on target use and internal mechanisms. By the way, this fact
> determines the degree of similarity of the algorithms which have "grown" from
> MPEG-1 (such as, Ogg Vorbis and MusePack). Each Layer has its own format of
> data stream and decoding algorithm. MPEG-1 algorithms are mainly based on
> known properties of perception of sound signals by a hearing aid of human (we
> have mentioned above about these techniques).
>
> Briefly about encoding algorithm used in MPEG-1. At the beginning of
> encoding, the source audio stream with the help of filters is divided on
> bandwidth. The continuation of the encoding process depends on used Layer.
>
> In the case of Layer 3 (MP3) the signal in each obtained bandwidth is
> decomposed on frequency components by applying MDCT (Modified Discrete Cosine
> Transform - a special case of Fourier Transform) that gives a set of
> coefficients. Further processing is focused on simplification of the signal
> in order to perform re-quantization of its spectral coefficients. Obtained
> spectrum is cleared (by filtering) of obviously inaudible components -
> low-frequency noise and high imperceptible spectrum components. At the next
> stage, considerably more complex psycho acoustic analysis is applied (as was
> described earlier) on the audible part of spectrum. After all these
> manipulations, the source signal is deprived of more than half of its
> information. In completion of all, compression of obtained stream by the
> simplified analogue of Huffman algorithm is performed (this is lossless
> compression method), that allows to reduce noticeably the stream size.
>
> In the case of Layer 2 the simplification process is quite similar. The
> difference consists in the object of re-quantization: re-quantization is
> performed on amplitude signal in each sub-band and not on the spectrum
> coefficients (some non-MP3 lossy encoders are based on the same technique).
>
> Complete set MPEG-1 is intended for coding signals with sample rates of 32,
> 44.1 and 48 kHz. Three MPEG-1 Layers that were mentioned above have
> distinctions in encoding mechanisms and, thus, they provide different
> compression factors and sounding quality of resulting streams. Layer 1 allows
> keeping signals in format 44.1 KHz / 16 bits without significant losses of
> quality at bitrate of 384 Kbps that gives 4 times profit of data size. Layer
> 2 provides, subjectively, the same quality at 192 - 224 Kbps, when Layer III
> (MP3) gives the same results at 128-160 Kbps. It is impossible to speak about
> advantages and disadvantages of one Layer compared to another, because each
> Layer is developed to achieve its own aim. For example, the advantage of
> Layer 3 actually consists in allowing of data compression 8-12 times
> (depending on bitrate) without significant losses of original sound quality.
> At the same time, speed of a compression provided by this Layer is the lowest
> (it is necessary to note, that on modern CPU's this restriction is not
> appreciable at all). Layer II is potentially capable to provide higher
> quality of coding on account of "easier" internal signal processing during
> transformation. However, Layer II does not allow to reach so high compression
> factors, which may be reached by using Layer III.
>
> Nuances of coding
>
> The technique of audio coding is complex enough and has a set of nuances.
> All of them cannot be explained within the framework of one article; however
> all the most important should be considered, as almost every user meets with
> them when encoding.
>
> Data encoding into MP3 (as well as into WMA and OGG) is performed by blocks:
> the coded file is divided on so-called frames of a certain equal length and
> each frame is encoded separately and is stored in a target stream. Thus, the
> target stream also has frame structure. Each frame can be encoded not on any
> bitrate, but only on one of those included in the standard table for MPEG1
> Layer 3 (Kbps): 32, 40, 48, 56, 64, 80, 96, 112, 128, 160, 192, 224, 256, 320
> (coding on intermediate bitrates is not stipulated by the standard, though it
> is possible). Because each frame is processed individually, it is possible to
> speak about data compression with constant (CBR) and variable (VBR) bitrate.
>
> CBR (Constant Bitrate) is a way of encoding when all frames are encoded on
> identical bitrate. In other words, bitrate of the whole encoded stream
> remains constant all along the stream.
>
> VBR (Variable Bit Rate) is a way of encoding when each separate frame is
> encoded with its own bitrate, calculated by encoder. The choice of bitrate
> for each frame is performed by the encoder according to performed psycho
> acoustic analysis.
>
> There is also one more encoding mode - ABR (Average bitrate). Encoding in
> this mode (it is true, at least, for MP3 coders) is similar to CBR encoding.
> However this encoding is performed on variable bitrate keeping the same
> average. Not going into technical details, we shall note that VBR and ABR
> encoding is much more flexible and, often, more favorable and qualitative,
> rather than in CBR mode.
>
> It is important to note, that ABR, VBR and CBR modes are used also in many
> coders rather than MP3.
>
> We shall consider now existing encoding techniques of stereo data stipulated
> in MPEG-1 Layer 1, 2, 3 standards. These methods, probably, with some
> different interpretations, are valid not only in MPEG, but also in other
> codecs.
>
> Dual Channel. This mode is intended for encoding of audio information in two
> channels as absolutely independent. In other words, encoding of audio occurs
> separately in each channel without tracking dependence of a signal in
> channels. As is implied from the name, this mode is mainly intended for
> coding of data with two parallel independent channels (for example, speech in
> English and German languages), and NOT with two channels carrying stereo
> information of sounding. In general, this mode is not recommended to be used
> for coding of stereo signal.
> Stereo. This mode differs from the Dual Stereo mode in reservoir usage.
> Reservoir - is a mechanism that is responsible for assignment of bits for
> encoded frames in the target stream. During encoding in stereo mode both
> channels are processed using the same reservoir, when in Dual Stereo mode,
> the signal is encoded, using independent reservoir for each channel. There
> are no other differences between the modes.
> Joint Stereo is common definition of the encoding methods of stereo
> information, which are based on the use of its redundancy. There are two
> versions of this method described in MPEG-1.
> MS Stereo. In this mode the encoded signal is re-divided on a middle channel
> (common constituent for both right and left channels) and a side channel
> (differented constituent of the channels) and processed as in Stereo mode,
> using some additional tricks.
> Intensity Stereo. In this mode encoded signal is divided on bandwidths. Then
> only bottom frequency ranges pass the actual encoding. In the top range, the
> encoder only registers average signal power in each bandwidth and actually
> doesn't encode the signal there. Encoding of stereo information in the bottom
> ranges is performed using MS Stereo or Stereo modes.
> It is necessary to note, that usage of MS Stereo mode does not introduce any
> additional errors in the signal. When re-dividing <left> + <right> channels
> on <middle> + <side> channels, nothing occurs, except for harmless and
> completely convertible mathematical calculations. At the same time, this
> simple reception of stereo data encoding allows the coder to accomplish its
> potential more effectively, rather than in mode Stereo.
>
>
> --
> Bhavesh
>
>
> "marcinont" wrote:
>
> > Does compressing mp3 filies affect the sound quaility?How many compressed
> > filies will fit on a regular CD?
> >
.
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