MP3 Advanced window contains 4 tabs, which are:
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 (slider in the middle) is recommended for lower bitrates (<160 kbps), high quality (moving the slider in the direction of "better") for bitrates >160 kbps.
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.
Phone, SW, AM, FM, Voice, Radio, Tape, HiFi, CD, Studio, R3Mix.
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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.
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.
At Expert tab you can find the opportunity to set the following options:
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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