Information on lossy audiocoding and the MPEGplus-project



Information on lossy audiocompression

Information, software, links about the MPEGplus project

Why apply lossy audiocompression:


How does it work:


Psychoacoustics:


global masking threshold of a 1 kHz sine-wave
 

global masking threshold of 4 sine-waves



Quantization:
 






This is the project name of my audiocoder. The encoder was developed during my study in my spare time. The motivation was the nonsatisfactory about MP3 encoders at the beginning of this project (in 1997-1998). Now the encoder has reached a very high quality that can be compared to the quality of MP3 encoders. MPEGplus utilizes decompositon into small frequency bands and therefore belongs to the group of so called subband coders.
The main advantage of this encoder is the highly tuned psychoacoustics, which allows pure VBR encoding (variable bitrate). MPEGplus aims to reach transparent results with default-parameters on nearly all audio signals one wants to encode.
 

About MPEGplus in general:


Lossless coding:

Psychoacoustic Model:


Quality and performance:



Download:


Sources:


Binaries:


Logos:   logos.zip (ca. 78 KB)
 
 

Programs with MPEGplus support:


Playback:

Winamp via plugin (http://www.winamp.com)

MediaJukebox via plugin (http://www.musicex.com/mediajukebox/)

DeliPlayer via plugin (http://www.deliplayer.com)

XMMS via plugin (under development) (http://sourceforge.net/projects/mpegplus/)
 

Batch-Encoding:
 
Easy CD-DA Extractor via external encoder (http://www.poikosoft.com)
CDex via external encoder (http://www.cdex.n3.net/)
EAC via external encoder (http://www.exactaudiocopy.de)
Audiograbber via external encoder (http://www.audiograbber.com-us.net/)

Monkey's Audio via external encoder (http://www.monkeysaudio.com)

MediaJukebox via external encoder (http://www.musicex.com/mediajukebox/)

WinDAC32 via Script

MP+ Frontend von M.Spueler http://www.mpegplus.de

CD Copy for en-/decoding and burning http://www.cdcopy.sk
 

FAQ:


 
When activating the equalizer mp+/mpc files seem to loose high frequencies in comparison to mp3 files. Why does this happen?
The option "EQ controlled by WinAMP" should be deactivated. WinAMP's build-in EQ will lead to loss of high frequencies (above 16 kHz) and will cause much more CPU-load than the mp+/mpc plugins' "fast EQ".
Even if I encode a file with the option "-bw 22050" (bandwidth set to 22.05 kHz) the mp+ file typically "cuts" off at about 18.5 kHz.
The switch "-bw x" only sets the maximum possible bandwidth. The encoder will only store frequencies that are assumed to be audible. The psychoacoustic model works very carefully in relation to the encoded bandwidth. Even if the encoded bandwidth does not equal the maximum bandwidth there should not be any audible difference.
For storing the full bandwidth one has to set "-VBRmode 2 -minSMR x" (x>0) or simply use "-insane" (this profile utilizes full bandwidth encoding).
What are the minimum and the maximum bitrates supported by mp+/mpc?
The encoder can theoretically use bitrates up to 1.32 MBit/sec. Normally such high bitrate will not occure. The minimum bitrate is used for zero-samples and will reach about 3.4 kbit/s.
Will forthcoming decoders/plugins support the current bitstream versions?
Of course! All forthcoming versions will support the existing bitstreams (SV4 to SV7) as well as the file extension ".mp+" and ".mpc".
Will mp+/mpc reach much higher audio quality with forthcoming versions or is the encoder near the final state in terms of quality?
In terms of quality the encoder has almost reached the final state. The modifications with version-changes are mostly related to debugging of file-i/o and parsing and not to the encoder-kernel.
The current encoder has proven its reliability in terms of quality in several tests by many users and on some hundreds tracks. Only direct A/B-comparison using high-quality headphones and under hard effort of listening may show minor differences to the original with a small number of tracks. None of the encoded files showed heavy or annoying artifacts.
Will the "-standard"-profile do for encoding or should I use "-xtreme" or even "-insane"?
The encoder was tested intensively and optimized in "-standard"-profile, the default setting. In this mode the quality of the encoded tracks reaches - despite the profile's naming - very high level!
The next profile "-xtreme" uses slightly modified parameters to lower the quantization noise further below the masking threshold - it offers even more headroom.
For the "-insane"-profile the parameters are tweaked heavily. Using this mode will store the full bandwidth of the input signal and lead to much higher bitrates than "-standard" or "-xtreme" need. The storage of full bandwidth is not based on psychoacoustic reasons - it was implemented at some users desire.
Summarization: When using "-standard"-profile you will get high quality audio-files. If you want to push it a bit further use "-xtreme". The use of "-insane" is not necessary in general.
If I want to encode the file Track of my favorite band.wav the message "ERROR: File not found!" is displayed.
When encoding tracks with long file-names the filenames must be given with quotation marks:
e.g. mppenc -v "Track of my favorite band.wav"


Will there be a Windows ACM-codec for mp+/mpc?

It is not out of question that there will be an ACM-codec in the future. But at the moment the main work is focussed on debugging and implementation of new features. Designing an ACM-codec is low priority.


Links:


Disclaimer (German):

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Andree.Buschmann@web.de