MP3 Advanced window contains 4 tabs, which are:

General Filtering VBR Expert

General

At the General tab you can select the necessary Channels. The suggested choice includes the following variants: Stereo, Joint Stereo, Forced Joint Stereo, Dual Channels and Mono. See the description of each mode in the table below:

In the right part you will find the opportunity to set Encoder Quality. You can specify the output quality; thus you can trade off encoding time against sound quality. The default (normal) is recommended for the lower bitrates (<160 kbps), high quality for bitrates >160 kbps. The voice quality is more or less optimized to generate the best quality for voice

There are some built-in presets you can use. They have for the most part been subject to and tuned via rigorous double blind listening tests to verify and achieve this objective. These are continually updated to coincide with the latest developments that occur and as a result should provide you with nearly the best quality currently possible. You can find them in Preset and Alt-Preset lists.

Presets available:

Phone, SW, AM, FM, Voice, Radio, Tape, HiFi, CD, Studio, R3Mix.

Alt-Presets available:

You can also select Mpeg Tag Version out of ID3 Ver.1 or ID3 Ver.2.

ID3 Ver.1

The audio format MPEG layer I, layer II and layer III (MP3) has no native way of saving information about the contents, except for some simple yes/no parameters like "private", "copyrighted" and "original home" (meaning this is the original file and not a copy). A solution to this problem was introduced with the program "Studio3" by Eric Kemp alias NamkraD in 1996. By adding a small chunk of extra data in the end of the file one could get the MP3 file to carry information about the audio and not just the audio itself.

The placement of the tag, as the data was called, was probably chosen as there were little chance that it should disturb decoders. In order to make it easy to detect a fixed size of 128 bytes was chosen. The tag has the following layout (as hinted by the scheme to the right):

If one sums the size of all these fields we see that 30+30+30+4+30+1 equals 125 bytes and not 128 bytes. The missing three bytes can be found at the very beginning of the tag, before the song title. These three bytes are always "TAG" and is the identification that this is indeed an ID3 tag. The easiest way to find a ID3v1/1.1 tag is to look for the word "TAG" 128 bytes from the end of a file.

As all artists doesn't have a 30 character name it is said that if there is some bytes left after the information is entered in the field, those bytes should be file with the binary value 0. You might also think that you cannot write that much in the genre field, being one byte big, but it is more clever than that. The byte value you enter in the genre field corresponds to a value in a predefined list. The list that Eric Kemp created had 80 entries, ranging from 0 to 79.

ID3 Ver.2

ID3v2 is a new tagging system that lets you put enriching and relevant information about your audio files within them. In more down to earth terms, ID3v2 is a chunk of data prepended to the binary audio data. Each ID3v2 tag holds one or more smaller chunks of information, called frames. These frames can contain any kind of information and data you could think of such as title, album, performer, website, lyrics, equalizer presets, pictures etc. The block scheme to the right is an example of how the layout of a typical ID3v2 tagged audio file may look like.

One of the design goals were that the ID3v2 should be very flexible and expandable. It is very easy to add new functions to the ID3v2 tag, because, just like in HTML, all parsers will ignore any information they don't recognize. Since each frame can be 16MB and the entire tag can be 256MB you'll probably never again be in the same situation as when you tried to write a useful comment in the old ID3 being limited to 30 characters.

Speaking of characters, the ID3v2 supports Unicode so even if you use the Bopomofo character set you'll be able to write in your native language. You can also include in which language you're writing so that one file might contain e.g. the same lyrics but in different languages.

Even though the tag supports a lot of byte consuming capabilities like inline pictures and even the possibility to include any other file, ID3v2 still tries to use the bytes as efficient as possibly. If you convert an ID3v1 tag to an ID3v2 tag it is even likely that the new tag will be smaller. If you convert an ID3v1 tag where all fields are full (that is, all 30 characters are used in every field) to an ID3v2 tag it will be 56 bytes bigger. This is the worst case scenario for ID3v1 to ID3v2 conversion.

Since it's so easy to implement new functionality into ID3v2, one can hope that we'll see a lot of creative uses for ID3v2 in the future. E.g. there is a built-in system for rating the music and counting how often you listen to a file, just to mention some brainstorm results that are included. This feature can be used to build playlists that play your favorite songs more often than others.

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Filtering

Highpass Filter cuts the lowest frequencies and passes the highest. The Lowpass Filter allows only the lower frequencies to be present into the output signal; it will cut the beautiful crystal sound of a violin (frequencies over 10 KHz), but if it could amplify rather than just pass the low frequencies, than it would enhance your favorite disco music with lots of percussions and bass.

Highpass filtering frequency (in kHz): Frequencies below the specified one will be cut off.

Width of Highpass filter (in kHz): The width of the highpass filter. The default is 15% of the highpass frequency.

Lowpass filtering frequency (in kHz): Frequencies above the specified one will be cut off.

Width of Lowpass filter (in kHz): The width of the lowpass filter. The default is 15% of the lowpass frequency.

Regarding to the Nyquist Sampling Theorem the sample rate have to be at least two times higher than the highest frequency of analog audio signal. For example, the human ear can detect sound across the frequency range of 20 Hz to 20 kHz. According to the sampling theorem, one should sample sound signals at least at 40 kHz in order for the reconstructed sound signal to be acceptable to the human ear.

So applying the low-pass analog filter to a file with sample rate of 8kHz you should take into consideration that the frequency parameter of this method have to be at least two times less ( < 4 kHz) than the file's sample rate.

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VBR

At VBR tab you should select Coding Method first. The description of each method can be found in the table below:

Maximum VBR Bitrate: Allows to specify an maximum bitrate when using VBR (Variable Bit Rate), this selecting depends on what base bitrate you have chosen in the main encoder tab. It's recommended to leave this set to 320 kbit/s unless you want low quality VBR files.

VBR Quality: In VBR mode, you are able to specify a quality setting which will affect encoding bitrate allocation. If you use quality 0, the max bitrate will be reached easily, while using quality 9 the bitrate usually will be around the base bitrate. The lower the VBR quality value, the better the audio quality, but also the bigger the output file. Recommended setting for high quality VBR encoding is 1 or 0.

Write VBR Header: This tag is embedded in frame 0 of the MP3 file. It lets VBR aware players correctly seek and calculate playing times of VBR files.

Target bitrate for ABR: The allowed range of the ABR bitrate is 4 - 310 kbit/s, you can use any integer value within that range.

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Expert

At Expert tab you can find the opportunity to set the following options:

• include CRC-Checksum
  When enabled, the encoder will calculate the cyclic redundancy check (CRC) for the MP3 frames, and will add the CRC value to the MP3 stream.
• comply as much as possible to ISO MPEG spec
  With this option, the encoder will enforce the 7680 bit limitation on total frame size. This results in many wasted bits for high bitrate encodings.
• allow block types to differ between channels
  Allows the left and right channels to use different block types. Normally this is not allowed, only because the FhG encoder does not seem to allow it either.
• short blocks
  Encode all frames using short blocks.
• bit reservoir
  Enable bit reservoir.

You can also set ATH Control function, useful for low volume. ATH is used to approximate an equal loudness curve. Select one of the suggested modes:

There are also three De-emphasis options: default, 0/15 microseconds and citt j.17.

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