Sound-Compression

The PCM (pulse-code modulation) established for CD-DA discs is not very compact and doesn't suit for delivering music via the net. That is why developers now are working on a number of complex compression algorithms. All of them differ very much in the sound quality, that is why a user has always to make a choice of an algorithm for its favorite music to be recorded.

• Flac:

  FLAC stands for Free Lossless Audio Codec, an audio format similar to MP3, but lossless, meaning that audio is compressed in FLAC without any loss in quality. This is similar to how Zip works, except with FLAC you will get much better compression because it is designed specifically for audio, and you can play back compressed FLAC files in your favorite player.
  

  If you have a “WAV” or “riff” file you can do lossless compression with this command:
  

  
           flac -7 your.wav         
  
  
  Note that the '-7' flag is high quality compression level. You can read more about FLAC on it (man-page).
  

  Metadata can be added to the FLAC file with something like:

  
           flac -T "TITLE=Freddie the freeloader" -T "ARTIST=Miles Davis" your.wav        
  
  
  
• Ogg Vorbis:

  is a fully Open, non-proprietary, patent-and-royalty-free, general-purpose compressed audio format for high quality (44.1-48.0kHz, 16+ bit, polyphonic) audio and music at fixed and variable bitrates from 16 to 128 kbps/channel. This places Vorbis in the same class as audio representations including MPEG-1 audio layer 3, MPEG-4 audio (AAC and TwinVQ), and PAC.

  Ogg Vorbis provides a high-quality format for you to listen to your music. Its file size is also smaller than MP3 and getting smaller as development continues. Vorbis already enjoys widespread player support and should be compatible with several major hardware players soon. With Vorbis, you can listen to your music with higher quality in less space. Also, using Vorbis means your player and encoder choices aren't bound by licensing terms.

  • oggenc

    (man page ) is part of the “vorbis-tools” package. This program is a complete encoder that creates Ogg Vorbis compressed soundfiles. OggEnc input files must currently be 16 or 8 bit PCM WAV, AIFF, or AIFF/C files. Files may be mono or stereo (or more channels) and sampling rates between 8kHz and 56kHz. Call oggenc with the “-h” flag or see the oggenc man page for more details.

    This will create a “sample.ogg” output soundfile:

    
              oggenc sample.wav        
    
    You can do a format conversion and pipe that to oggenc using unix pipes, for example the following command will convert a soundfile from .snd format to .wav and feed it to the encoder:

    
              sox -t .au sample.snd -t .wav - |        oggenc -o sample.ogg -        
    
    
    Get high-quality encoding averaging 256 kbps (but still VBR):

    
             oggenc infile.wav -b 256 -o out.ogg        
    
    
    VBR: Variable bit rate.
    

    Metadata (title and author can be added to an Ogg file as follows:
    

    
             oggenc audio-cd.wav -q 8 -t "Freddie the freeloader" -a "Miles Davis" -o freddie.ogg       
    
    
    
  • ogg123

    (man page ) is also part of the “vorbis-tools” package. In essence it is simple command line Ogg Vorbis decoder and player. You can play your “ogg files” with ogg123 with:

    
              ogg123 -d oss sample.ogg        
    
  • Play your Ogg files with xmms: The X Multimedia System (xmms) can recognize and play Ogg Vorbis encoded files. Just open xmms as follows:

    
             xmms sample.ogg &       
    
    
    

• mp3:
  does not need an introduction. It is a stereo compression standard that has become widespread in hardware and firmware based players, and is widely used for web distribution of music content. Regretfully even though it is a standard, it is not for free. Patent licenses under the combined patent portfolio of Fraunhofer IIS-A and Thomson multimedia are granted by Thomson multimedia exclusively and cover format encoders and decoders and commercial (revenue-generating) distribution of music. Read all the details in http://www.mp3licensing.com.

  According to the patent owners it is not possible to independently create an mp3 encoder without infringing on the patents. Recently they have started to crack down on open source projects that have created free alternatives to the commercial mp3 encoders. Read more about this in the lame (http://www.sulaco.org/mp3) and bladeenc (http://bladeenc.mp3.no/) web sites

  A couple of widely available encoders are available in PlanetCCRMA, but are only intended to be used as tools for learning about mp3. If you encode your own mp3s you should own a properly licensed encoder.

  • lame

    (man page) means, Lame Ain't an MP3 Encoder and is an educational tool to be used for learning about MP3 encoding. The goal of the LAME project is to use the open source model to improve the psycho acoustics, noise shaping and speed of MP3. Run lame with the “–help” option or see the lame man page for more details.

    Audio files created with Lame can be played back by popular MP3 players such as Xmms mpg123 or madplay.

    For example a command for a fixed bit rate jstereo 128kbs encoding, highest quality :

    
             lame -h sample.wav sample.mp3        
    
    
    A good choice for Music with a broad dynamic range might be :

    
             lame -abr sample.wav sample.mp3        
    
    
    
  • Compressing audio with Lame presets

    The Lame –preset switches are designed to provide the highest possible quality. 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 from LAME.

    To activate these presets:

    1. --preset standard
       

       This preset should generally be transparent to most people on most music and is already quite high in quality.

    2. --preset extreme
       

       If you have extremely good hearing and similar equipment, this preset will generally provide slightly higher quality than the standard mode.

    3. --preset insane
       

       This preset will usually be to much for the common user and most situations, but if you must have the absolute highest quality with no regard to filesize, this is the way to go.

    4. --preset  kbps
       

       Using this preset will usually give you good quality at a speci- fied bitrate. Depending on the bitrate entered, this preset will determine the optimal settings for that particular situa- tion. While this approach works, it is not nearly as flexible as VBR, and usually will not attain the same level of quality as VBR at higher bitrates.

    For example to use the extreme preset you will use this command:

    
              lame –preset extreme infile.wav   outfile.mp3        
    
    
    Add metadata tags to your mp3 file:

    
             lame –preset standard –tt "Mars" –ta "Gustav Holst" cd.wav mars.mp3       
    
    
    
  • Converting MP3s to raw audio to create audio CDs

                mpg123 -s file.mp3 | sox -c2 -s -w -t raw \
                      44100 - -t wav - $>$ newsoundfile.wav
    
    
    mpg123 converts file.mp3 to the .wav file newsoundfile.wav and makes it ready to be part of audio CDs.

  • bladeenc
    is a freeware MP3 encoder. It is based on the same ISO compression routines as mpegEnc, so you can expect roughly the same, or better, quality . The main difference is the appearance and speed. BladeEnc doesn't have a nice, user-friendly interface like mpegEnc, but it is more than three times faster, and it works with several popular front-end graphical user interfaces. Run bladeenc without arguments to get a usage description. Unfortunately BladeEnc is not longer in development but its features appear to be part of the Ogg team now.