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Guest
Does compressing mp3 filies affect the sound quaility?How many compressed
filies will fit on a regular CD?
filies will fit on a regular CD?
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Guest
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.
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.
G
Guest
What?
Bhavesh said: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.
G
Guest
Bhavesh said: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.
G
Galley
What you have said is actually redundant, since MP3 files are already
compressed. WAV files are uncompressed and require 10MB per minute of audio.
128Kbps MP3 files only require 1MB. So that would give your roughly 700 minutes
on a 700MB CD-R, which equates to 175 4-minute songs.
compressed. WAV files are uncompressed and require 10MB per minute of audio.
128Kbps MP3 files only require 1MB. So that would give your roughly 700 minutes
on a 700MB CD-R, which equates to 175 4-minute songs.