


                                    MP3EncV3.0


              Next  Generation  High-End  MPEG  Layer-3  Encoding

                Fraunhofer  Institute  for  Integrated  Circuits

                          http://www.iis.fhg.de/audio/



                                    25th  March  1998


Contents
1    For  the  impatient                                                                                     5
     1.1     Introduction   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     5
     1.2     Some examples     .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     6
     1.3     Command line switch reference .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     7


2    MP3Enc  Features                                                                                        8
     2.1     Basics      .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     8
             2.1.1       Samplerate  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     8
             2.1.2       Bitrate  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     8
             2.1.3       Stereo mode    .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     8
             2.1.4       Encoding speed   .  .   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .     9
             2.1.5       Input file specification  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   10
             2.1.6       Output file specification .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   11
     2.2     Advanced features  .  .  .  .  .  .   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   11
             2.2.1       Overriding default settings     .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   11
             2.2.2       Tids & bits   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   12


3    Troubleshooting                                                                                         14
     3.1     Is it really a bug?    .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   14
     3.2     Reporting the bug  .   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   14
     3.3     Sample bug report  .   .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .  .   15


                                                            1


License   agreement


USE OF THE SOFTWARE IS SUBJECT TO THE SOFTWARE LICENSE
TERMS SET FORTH BELOW. USING THE SOFTWARE INDICATES YOUR
ACCEPTANCE OF THESE TERMS. IF YOU DO NOT ACCEPT THESE
TERMS, YOU MUST RETURN OR DELETE THE SOFTWARE IMMEDIATELY.
This software is distributed as demoware.  You are entitled to use this software
MPEG Layer-3 codec for 30 days for evaluation purposes.  If you want to continue
to use this software codec after the evaluation period, or if you want to use this
software commercially, you are required to buy the software.  In the sense of these
license terms, USE shall mean running one of the programs and/or make use of
bitstreams generated by this software.  You may give copies of the demo version of
this software to other people as long as no file is changed and no file is omitted.
You may not sell, rent or lease the software to others.
If you have bought this product, you are entitled to use this product for your own
use.  You may not sell, rent or lease the software to others without written
permission of Fraunhofer Institute for Integrated Circuits (IIS). You may use only
one copy of the software at one time.  You may not use this software on a network
or on more than one computer at the same time without a licence for concurrent
use.
You must not give away your personal registration code.  Doing so will result in an
infringement of copyright.  Fraunhofer IIS and/or OPTICOM retain the right of
claims for compensation in respect of damage which occured by your giving away
of the registration code.  This claim shall also extent to all costs which Fraunhofer
IIS and/or OPTICOM incur in defending itself.
The license and right for the use of this Software does not include the right to
create, generate, encode or otherwise modify data or bit streams:


     o   to be used, sold, published, distributed, disposed of or otherwise marketed
         via pre-recorded media, such as but not limited to CD-ROM, magnetic
         tapes, memory cards and the like;


     o   to be used, sold, reproduced, published, distributed, disposed of or
         otherwise marketed via any kind of network, if a user will have to pay a
         monetary or equivalent compensation for the access, copying etc.  of such
         data or bit stream;


     o   for the purpose of broadcast and/or radio and/or multicast service
         transmission such as but not limited to    Internet Radio    and the like.

                                                            2


WARRANTY AND DISCLAIMER


There are no warranties associated with this software.  While we believe that our
software is reasonably bug free and well behaved, we are in no way responsible if
our software does not work the way you would expect it to work.  No matter if it
locks up your computer, garbles your floppy disks or does any other harmful
things to your computer -- it is entirely your problem.  Fraunhofer IIS and/or
OPTICOM are not liable for any infringements or damages of third parties' rights
in consequence of your use of this product.  Fraunhofer IIS and/or OPTICOM are
in no event liable for, respectively do not warrant the trustworthiness, quality,
industrial exploitability, serviceability of this product for the supposed purpose or
any other purposes.  All orders are subject to the general terms and conditions
"Allgemeine Verkaufs- und Lieferbedingungen" of OPTICOM.

This information may be subject to change.
All  brand  and  product  names  are  trademarks  and/or  registered  trademarks
of  their  respective  owners.  All  rights  reserved.
                                                            3


Why and how to buy this software



Why  should  you  buy  this  software?


     o   Your license will be valid for all V3.xx versions of the encoder


     o   As a registered user, you can get free support by email, mail, fax or phone
         (see page 14)


     o   You support the development of better versions of  MP3Enc and the
         development of even better compression algorithms.


     o   No more 30-second limit!


     o   The unlimited MPEG Layer-3 Decoder l3dec V2.74 is included.


How  to  buy  this  software


Please see the OPTICOM Website, http://www.opticom.de/ for information on
prices and ordering.  You can also get information by sending a fax requesting
prices to +49 (0) 9131 / 691-325 or by sending email to sales@opticom.de. If all
else fails, OPTICOM can be reached by snail-mail at:


         OPTICOM
         Am Weichselgarten 7
         D-91058 Erlangen
         Germany


Please do not direct any questions about pricing and ordering information at
Fraunhofer IIS or to the technical support facilities.
                                                            4

Chapter   1


For   the   impatient


If you are new to audio compression, you should read section 1.1 for an
introduction about audio compression and MPEG Layer-3.
If, however, you want to jump right into the business of sound compression, then
Section 1.2 will show you some prefabricated command lines that will give you
compressed audio streams right away.
If you are an expert in audio coding already, the command line switch reference
page (see page 7) might come in handy.

1.1        Introduction


There is a lot of confusion surrounding the terms audio compression, audio
encoding, and audio decoding.  This section will give you an overview what audio
coding (another one of these terms...)  is all about.



The  purpose  of  audio  compression


Up to the advent of audio compression, high-quality digital audio data took a lot
of hard disk space to store.  Let us go through a short example.
You want to, say, sample your favorite 1-minute song and store it on your
harddisk.  Because you want CD quality, you sample at 44.1 kHz, stereo, with 16
bits per sample.
44100 Hz means that you have 44100 values per second coming in from your
sound card (or input file).  Multiply that by two because you have two channels.
Multiply by another factor of two because you have two bytes per value (that's
what 16 bit means).  The song will take up


                 samples                             bytes            s
         44 100 ____________ . 2 channels      . 2 ___________. 60 ______ = about 10 MByte
                       s                             sample          min


of   storage space on your harddisk.
If you wanted to download that over the internet, given an average 28.8 modem,
it would take you (at least)


                                      bits            bits         s
              10 000 000 bytes    . 8 _______=(28 800 ______. 60 ______) = about 45 minutes
                                      byte              s        min


                                                            5

CHAPTER 1.   FOR THE IMPATIENT                                                              6



                             Just  to  download  one  minute  of  music!

Digital audio coding, which - in this context - is synonymously called digital audio
compression as well, is the art of minimizing storage space (or channel bandwidth)
requirements for audio data.  Modern perceptual audio coding techniques (like
MPEG Layer-3) exploit the properties of the human ear (the perception of sound)
to achieve a size reduction by a factor of 12 with little or no perceptible loss of
quality.
Therefore, such schemes are the key technology for high quality low bit-rate
applications, like soundtracks for CD-ROM games, solid-state sound memories,
Internet audio, digital audio broadcasting systems, and the like.



The  two  parts  of  audio  compression


Audio compression really consists of two parts.  The first part, called encoding,
transforms the digital audio data that resides, say, in a WAVE file, into a highly
compressed form called bitstream.  To play the bitstream on your soundcard, you
need the second part, called decoding.  Decoding takes the bitstream and
re-expands it to a WAVE file.
The program that effects the first part is called an audio encoder.  MP3Enc is
such an encoder; there are others, see http://www.fhg.iis.de/audio/.
The program that does the second part is called an audio decoder.  One
well-known MPEG Layer-3 decoder is WinPlay3, another l3dec.  Both can be
found on http://www.fhg.iis.de/audio/.



Compression  ratios,  bitrate  and  quality


It has not been explicitly mentioned up to now:  What you end up with after
encoding and decoding is not the same sound file anymore:  All superflous
information has been squeezed out, so to say.  It is not the same file, but it will
sound  the same - more or less, depending on how much compression had been
performed on it.
Generally speaking, the lower the compression ratio achieved, the better the
sound quality will be in the end - and vice versa.  Table 1.1 gives you an overview
about quality achievable.
Because compression ratio is a somewhat unwieldy measure, experts use the term
bitrate  when speaking of the strength of compression.  Bitrate denotes the
average number of bits that one second of audio data will take up in your

compressed bitstream.  Usually the units used will be kbps, which is   kbits/s, or
1000 bits/s.   To calculate the number of bytes per second of audio data, simply
divide the number of bits per second by eight.

_____Bitrate_______Bandwidth___________Quality_comparable_to_or_better_than______________________

      8 kBps            2.5 kHz         POTS (telephone sound)
    16 kBps             4.5 kHz         shortwave radio
    32 kBps             7.5 kHz         AM radio
    64 kBps              11 kHz         FM radio
  128 kBps               15 kHz         CD


                                Table 1.1:  Bitrate versus sound quality


1.2        Some  examples


     o   Encode a WAVE-file myfile.wav to a bitrate of 128000 bits/s, writing to a
         plain bitstream myfile.mp3

         mp3enc  -br  128000  -if  myfile.wav  -of  myfile.mp3


     o   Encode a plain PCM file (2-channel, 44.1 kHz) to a plain 56 kBit/s Layer-3
         stream, using the encoder as filter

         readFromSoundCard  |  mp3enc  -sti  -sto  -iff  "nc=2  sr=44100
         bps=16"  -br  56000  -qual  3  |  streamToWeb

1.3        Command  line  switch  reference


__switch_________parameter_____________________________________________________see_section________

  -br            bitrate                                                        2.1.2
  -if            input file name
  -of            output file name
  -iff           input file format
  -l3wav         write Microsoft RIFF/WAVE layer-3 file                         2.1.6
  -sti           take input from pipe (stdin)
  -sto           write output into a pipe (stdout)
  -qual          quality                                                        2.2.2
  -esr           effective sampling rate                                        2.2.1
  -crc           CRC checksum                                                   2.2.2
  -dm            downmix stereo file to mono                                    2.2.2
  -v             be verbose
  -no-is         do not use intensity stereo                                    2.1.3


Chapter   2


MP3Enc   Features



2.1        Basics


2.1.1        Samplerate


Sample rate is the rate at which the samples are read from your sound card when
you sample.  Sample rate is directly linked to audio bandwidth achievable:  A
sound file with a sample rate of 8 kHz does not contain frequencies beyond
4 kHz.  This means that you should always use the highest sample rate that your
sound card supports when you sample a signal.
The encoder changes the sample rate of your audio data to match it to the audio
quality of the bitstream produced by the encoder.  This process is called
downsampling.



2.1.2        Bitrate


The main parameter controlling the sound quality is the bitrate  that the encoder
runs at.  In a nutshell, the higher the bitrate, the better the quality.
The bitrate of the encoder is linked to the samplerate that the encoded file will
have.  Usually, the encoder will choose a samplerate that is suited best for
encoding at that bitrate.  You can override this samplerate using the -esr switch
(see section 2.2.1).
The bitrate of the bitstream output is selected via the -br switch.  The bitrate is
specified in bits/second.  The bitrate is the total bitrate for all encoded channels,
i.e. if you select -br  112000         and encode in stereo, both channels will be stuffed
into one bitstream of 112000 bits/second.
The encoder supports bitrates of 8, 16, 18, 20, 24, 32, 40, 48, 56, 64, 96, 112,
128, 160, 192 and 256 kBit/s.  While all of these can be used with mono signals,
stereo works from 20 kBit/s on upwards.



2.1.3        Stereo  mode


If encoding stereo, the bitrate of the encoder is linked to a stereo mode.  MPEG
Layer-3 knows four modes for stereo encoding.


                                                            8

CHAPTER 2.   MP3ENC FEATURES                                                                9
                                   ________Bitrates_____________stereo_mode___________

                                         8000-    18000         mono only
                                       18000-     96000         MS/IS stereo
                                       96000-     192000        MS stereo
                                     192000-      256000        stereo


                                    Table 2.1:  different stereo modes



dual  channel         (also known as dual mono) In this mode, the encoder treats the
         two input channels as separate entities, assuming there is no similarity
         between the channels.  This would be appropriate if you e.g. have a bilingual
         signal where one channel contains a german speaker and one contains an
         english speaker.


stereo     In this mode, like in dual channel above, the encoder makes no use of
         potentially existing correlations between the two input channels.  It can,
         however, negotiate the bit demand between both channel, i.e. give one
         channel more bits if the other contains silence.


MS  stereo        In this mode, the encoder will make use of a correlation between both
         channels.  The signal will be matrixed into a sum (   mid   ) and difference
         (   side   ) signal.  For quasi-mono signals, this will give a significant gain in
         encoding quality.

         This mode does not destroy phase information like IS stereo (see below)
         and thus can be used to encode DOLBY  ProLogic(tm)   surround signals.


MS/IS  stereo           In this mode, high-frequency parts of the signal will be
         downmixed to mono and transmitted with a direction information (which is
         basically a pan).  This mode (called    intensity stereo    will loose phase
         information and should not be used for high-quality encoding.


Table 2.1 gives you an overview which mode will be used for which bitrate.



2.1.4        Encoding  speed


Several factors influence the speed of the encoder.  They include:


     o   Number of channels in the output signal.  If your output signal has only one
         channel, the encoder will run at twice the speed compared to stereo
         encoding.


     o   Output sample rate.  If the encoder produces a file at 22.050 kHz (that is, a
         file that contains 22050 samples per second), it runs at twice the speed
         compared to one that produces twice the number of samples per second
         (i.e. produces a 44.1 kHz output).


     o   Mismatch between input and output sample rate.  If your input and output
         sample rates differ, the encoder will have to run a resampling filter and thus
         will be slower.  (Integer ratios between input and output sample rate
         perform slightly better than non-integer ratios, though).

     o   Time-domain bandlimiting.  The encoder needs to band-limit the signal to
         compress it.  By default, the encoder will use a high-quality time domain
         filter to do this band-limiting.  You can tell it to use a faster filter, possibly
         sacrificing some quality (see 2.2.2).


     o   Full huffman search and careful iteration.  You can tell the encoder to try
         hard to do the best encoding possible, at the expense of a factor of up to
         three in running time (see 2.2.2).


Version V3.0 of the encoder reaches realtime speed on a Pentium 166 when
encoding at 64 kBit/s, 22,050 kHz, stereo.  On a SUN Sparc Ultra-1 (143 MHz)
the performance is similar.



2.1.5        Input  file  specification


The encoder can read AIFF, AIFF-C, WAV/RIFF and raw PCM data files.  While
the first three only work from a file, plain PCM data can be fed into the encoder
via a pipe.  This is useful for live encoding (also known as streaming ).



Input  from  file:  filename


-if  filename           will tell the encoder the filename it reads it input from.  If the file
         is a RIFF/WAVE file or an AIFF/AIFC file, the encoder will automatically
         adapt to the sound file format.  For other formats or plain PCM data, see
         below.



Piping  data  into  the  encoder


-sti     tells the encoder to get its input from stdin rather than from a file.  This
         only works when the input is plain pcm data (see below).



plain  PCM  data  input


If the encoder gets its input as plain pcm data (or if it does not recognize the
sound format by itself), you need to tell it all about the structure of the PCM
stream, i.e.  the number of bits per sample, the number of channel and the
samplerate.


-iff  fileformat             This is a string containing name=value pairs, separated by
         blanks.  Table 2.2 gives a reference which names and values are possible
         here.

         For stereo files, the encoder assumes that the PCM data is interleaved and
         that the sample for the right channel follows that for the left channel.


As an example, -iff  "nc=2  sr=44100  bps=16" would be used to read a
44.1 kHz stereo file with 16 bits per sample while -iff  "nc=1  sr=8000  bps=8"
would tell the encoder that the data is mono, sampled at 8 kHz with 8 bits per
sample.
Remember that this feature is only needed for input from files other than
RIFF/WAV, AIFF and AIFC.

__Name__________________|_____Value(s)______________Explanation_________________________________________________________
  sr                    |  any                    The rate the PCM signal is sampled at [Hz]
  nc                    |  1, 2                   The number of channels in the signal
  bps                   |  8, 16, 24, 32          The number of bits per sample
  little-endian         |                         The signal is little-endian (Intel format)
  big-endian            |                         The signal is big-endian (Motorola format)


                               Table 2.2:  input file format specification



2.1.6        Output  file  specification


On output, the encoder can be instructed to write a plain Layer-3 bitstream or a
wave file containing the Layer-3 stream.  These wave files can be played by the
media control on a machine running under Microsoft Windows that has the
Layer-3 ACM codec installed (you can get one by installing Microsoft Netshow,
http://www.microsoft.com/netshow/).
If the output is a plain Layer-3 stream, it can be piped into other applications.
This is useful for live streaming.


-of  filename           tells the encoder the filename of the file that the encoder will
         write the bitstream to.  If the file does not exist, it is created; if it does
         exist, it will be overwritten.


-l3wav       tells the encoder to wrap the MPEG Layer-3 file into a Microsoft
         RIFF/WAVE file.



Streaming  data  out  of  the  encoder


-sto     tells the encoder to write its output into stdout rather than in a file.  This
         only works when the output is a raw Layer-3 bitstream (i.e. it does not work
         in conjunction with -l3wav).
2.2        Advanced  features


2.2.1        Overriding  default  settings


Many of the following features override the encoder's idea of best-quality settings.
You should be aware that overriding the encoder default settings is something for
experts.  You might wreck the encoding quality in a number of ways without first
noticing it.  Also, the encoder is not guaranteed to run at all parameter
combinations.  Proceed  at  your  own  risk!


-esr     Output (effective) sample rate.  Usually, the encoder will choose an output
         sample rate from 8, 16, 32, or 48 kHz.  With some soundcards, it is not
         possible to play files with sample rates of 48 kHz, others cannot do 32 kHz.
         With this switch, you can tell the encoder to use another output sample

_________footnotes__________________________________
    1 You can also use this switch to match your output sample rate to an integer fraction of the
      input sample rate to get slightly faster performance


CHAPTER 2.   MP3ENC FEATURES                                                               12
-dual      Use dual channel stereo instead of the default mode (see table 2.1).  At
         bitrates of 128 kBit/s and below, this switch will almost certainly decrease
         the sound quality.


-bw    Tell the encoder to use another bandwidth.  Increasing the bandwidth from
         the default setting will work for some signals, but might produces ringing
         artefacts for others.  Use with care!

         It is not possible to choose bandwidths above half the output sample rate.


-no-is       Tell the encoder not to use intensity stereo (see 2.1.3).  Some special
         signals experience susceptible loss of quality if phase information is
         destroyed; in these cases, you may gain some sound quality using this
         switch.



2.2.2        Tids  &  bits


-crc     For transmission over serial lines with bit errors, parts of the bitstream can
         be protected by calculating a CRC checksum.  If you are just producing for
         harddisk storage, there is no need to set this switch.


-dm    To encode at bitrates ranging from 8 to 18 kBit/s, you need a mono input
         signal.  This switch tells the encoder to downmix a stereo input signal into
         one channel, producing mono output.  The downmix is calculated as the
         sum of the left and right channel, attenuated by 6 dB.


-qual      This switch controls the tradeoff between fast encoder operation and best
         sound quality.  Table 2.3 gives you an overview which features of the
         encoder are switched on/off by the -qual switch.

         In future versions of the encoder, more features might be controlled by this
         switch.  The only facts you should count on:


             o  fastest operation is guaranteed with -qual  0

             o  highest encoding quality is reached with -qual  9 (for some figures on
                encoding speed see section 2.1.4)


CHAPTER 2.   MP3ENC FEATURES                                                               13

__Feature_______________________|___________Explanation_________________________________________________
  Soft time-domain filtering    |           Use  a  high-quality  time  domain  filter  instead  of
                                |           fast MDCT
                                |
  Best match sampling rate      |           Use  the  best  sample  rate  without  regard  to  filter
                                |           running time.  Adapting to this sample rate might
                                |           use CPU-intensive filtering.
                                |
  Full huffman search           |           Find  the  best  huffman  code  book  possible  to  en-
                                |           code  the  spectrum  of  each  frame.  A  few  percent
                                |           bits can be saved in each frame, available for higher
                                |           quality in following frames.
                                |
  Many outer loops              |           Shape the quantization noise very carefully.


                       Table 2.3:  Features controlled by the -qual switch

Chapter   3


Troubleshooting


No software is free of errors.  If you believe you have found an error in the
operation of  MP3Enc, and you have checked the list below, please report the
error to our bugtracking address.

3.1        Is  it  really  a  bug?


Before you report a bug to our engineers, please verify that the bug is really in the
software and not in your configuration.  Table 3.1 helps you track down the bug
yourself and see if it can be fixed.

3.2        Reporting  the  bug


To assist our engineers in the processing of your bug report, we ask you to include
in your mail

     o   The version of the encoder you are using.


     o   Your user name and serial number as reported by the encoder.


     o   The operating system (name and version) you are running the software
         with.  If you are using a sort of UNIX, please cite the output of  uname  -a.
         If you are using Windows, please right-click on the    My Computer    icon
         that usually resides in the top left-hand corner of your screen and report the
         lines following    System    and    Computer   .


     o   The exact command line that you entered before you encountered the error.


     o   The output of the encoder when appending the -v switch to the command
         line.

If you have gathered this information, please fax it to OPTICOM
(fax: +49 (0) 9131 / 691-325) or write an email to l3bugs@iis.fhg.de.
If you have bought this product, you will get an immediate acknowledgement by
email once your bug report has reached us.  You will receive a second email as
soon as the bug report has been processed.  Expect some delay between the first
and second email.



                                                           14

CHAPTER 3.   TROUBLESHOOTING                                                              15
__Symptom_________________________________________|_Check_this:____________________________________________________________
  AL  error  :    AL__detect  :                   | Have you given an input file to the encoder (see
  Unable  to  open  file!                         | section  2.1.5)?  Does  the  input  file  exist?  Is  it
                                                  | readable?
__________________________________________________|________________________________________________________________________
  could  not  open  output                        | Have  you  given  an  output  file  to  the  encoder
  file                                            | (see section 2.1.6)?  Does the output directory
                                                  |
                                                  | exist and is it writeable?  Does a file of the same
                                                  | name exist and is it deleteable?
__________________________________________________|________________________________________________________________________
  No  parameters  for  this                       | Did  you  override  any  of  the  encoders  parame-
  bitrate/samplerate                              | ters  (stereo  mode,  samplerate)?  If  so,  try  an-
                                                  | other samplerate.
__________________________________________________|________________________________________________________________________
  bitrate  too  low/high                          | MPEG   Layer-3   only   allows   bitrates   ranging
                                                  | from 8 kBit/s to 320 kBit/s.
__________________________________________________|________________________________________________________________________
  The Layer-3 file sounds muffled                 | Try  using  a  higher  bitrate.  Try  using  a  higher
                                                  |
                                                  | bandwidth  (see  section  2.2.1).    Try  using  a
                                                  | higher effective sample rate (see section 2.2.1)
__________________________________________________|________________________________________________________________________
  The stereo image is destroyed.                  | Try using the -no-is switch


                          Table 3.1:  Bug symptoms and possible causes



3.3        Sample  bug  report


This is a sample bug report that you may use as a template for your own.


To:  l3bugs@iis.fhg.de
Subject:  Mp3enc  bug


Hello,


I  am  using  mp3enc  Version  V3.0  on  a  PC
(according  to the  System  Properties  dialog,
it  is  running  Microsoft  Windows  NT 4.00.1381;
the  computer  contains  a
x86  Family  5  Model  2  Stepping  12 AT/AT  compatible
and  64,951  KB  RAM).


My  serial  number  and  user  name  (as  reported  by  mp3enc)  are
  Hantan  Blaumilch,  123456.


When  I  run  the  program  as


mp3enc  -br  127957  -if  myfile.wav  -of  foobar.mp3  -v


I  get  the  following  error  message:


***********  MPEG  Layer-3  Encoder  V3.00  (build  Mar    4  1998)  ***************
                                       (C)  1998  by  Fraunhofer  IIS-A


This  program  is  protected  by  copyright  law  and  international  treaties.
Any  reproduction  or  distribution  of  this  program,  or  any  portion
of  it,  may  result  in  severe  civil  and  criminal  penalties,  and  will  be
prosecuted  to  the  maximum  extent  possible  under  law.


in:    44100  Hz,  2  channel(s),  16  bit/sample
out:  44100  Hz,  2  channel(s),  128000  bit/s
MS  Stereo  ON
6144  /    830902  (  1%)
**  mp3enc  error:  Illegal  codebook  encountered.


Regards,
   Hantan


Bibliography


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  [8]  B. Grill, J. Herre, et al.  Improved MPEG-2 audio multi-channel encoding.  In
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