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       oggenc - encode audio into the Ogg Vorbis format


       oggenc  [  -hrQ ] [ -B raw input sample size ] [ -C raw input number of
       channels ] [ -R raw input samplerate ] [ -b  nominal  bitrate  ]  [  -m
       minimum  bitrate  ]  [ -M maximum bitrate ] [ -q quality ] [ --resample
       frequency ] [ --downmix ] [ -s serial ] [ -o output_file ] [ -n pattern
       ]  [  -c  extra_comment  ] [ -a artist ] [ -t title ] [ -l album ] [ -G
       genre ] [ -L lyrics file ] [ -Y language-string ] input_files ...


       oggenc reads audio data in either raw, Wave, or AIFF format and encodes
       it  into  an  Ogg  Vorbis stream.  oggenc may also read audio data from
       FLAC and Ogg FLAC files depending upon compile-time  options.   If  the
       input  file  "-"  is  specified,  audio data is read from stdin and the
       Vorbis stream is written to stdout unless the  -o  option  is  used  to
       redirect  the  output.  By default, disk files are output to Ogg Vorbis
       files of the same name, with the extension changed to ".ogg" or ".oga".
       This  naming convention can be overridden by the -o option (in the case
       of one file) or the -n option (in the case of several files).  Finally,
       if  none  of these are available, the output filename will be the input
       filename with the extension (that part after the  final  dot)  replaced
       with ogg, so file.wav will become file.ogg.
       Optionally, lyrics may be embedded in the Ogg file, if Kate support was
       compiled in.
       Note that some old players mail fail to play streams with more  than  a
       single Vorbis stream (the so called "Vorbis I" simple profile).


       -h, --help
              Show command help.

       -V, --version
              Show the version number.

       -r, --raw
              Assume input data is raw little-endian audio data with no header
              information. If other options are  not  specified,  defaults  to
              44.1kHz  stereo 16 bit. See next three options for how to change

       -B n, --raw-bits=n
              Sets raw mode input sample size in bits. Default is 16.

       -C n, --raw-chan=n
              Sets raw mode input number of channels. Default is 2.

       -R n, --raw-rate=n
              Sets raw mode input samplerate. Default is 44100.

       --raw-endianness n
              Sets raw mode endianness to big endian (1) or little endian (0).
              Default is little endian.

              Informs  oggenc  that the Vorbis Comments are already encoded as
              UTF-8.  Useful in situations where the shell is using some other

       -k, --skeleton
              Add  a  Skeleton  bitstream.   Important  if  the  output Ogg is
              intended to carry multiplexed or chained streams.   Output  file
              uses .oga as file extension.

              Support for Wave files over 4 GB and stdin data streams.

       -Q, --quiet
              Quiet mode.  No messages are displayed.

       -b n, --bitrate=n
              Sets  target bitrate to n (in kb/s). The encoder will attempt to
              encode at approximately this bitrate. By default, this remains a
              VBR  encoding.  See  the  --managed  option  to  force a managed
              bitrate encoding at the selected bitrate.

       -m n, --min-bitrate=n
              Sets minimum bitrate to n (in kb/s). Enables bitrate  management
              mode (see --managed).

       -M n, --max-bitrate=n
              Sets  maximum bitrate to n (in kb/s). Enables bitrate management
              mode (see --managed).

              Set bitrate management mode.  This  turns  off  the  normal  VBR
              encoding,  but  allows  hard  or  soft bitrate constraints to be
              enforced by the encoder. This mode is much slower, and may  also
              be  lower quality. It is primarily useful for creating files for

       -q n, --quality=n
              Sets encoding quality to n, between -1 (very low) and  10  (very
              high).  This  is  the  default mode of operation, with a default
              quality level of 3. Fractional quality levels such  as  2.5  are
              permitted.  Using  this  option  allows the encoder to select an
              appropriate bitrate based on your desired quality level.

       --resample n
              Resample input to the given sample rate (in Hz) before encoding.
              Primarily useful for downsampling for lower-bitrate encoding.

              Downmix  input  from stereo to mono (has no effect on non-stereo
              streams). Useful for lower-bitrate encoding.

       --advanced-encode-option optionname=value
              Sets an advanced option. See the Advanced  Options  section  for

       -s, --serial
              Forces  a  specific  serial number in the output stream. This is
              primarily useful for testing.

              Prevents comments in FLAC and Ogg FLAC files from  being  copied
              to the output Ogg Vorbis file.

       -o output_file, --output=output_file
              Write  the  Ogg  Vorbis  stream  to output_file (only valid if a
              single input file is specified).

       -n pattern, --names=pattern
              Produce filenames as this string, with %g, %a, %l,  %n,  %t,  %d
              replaced by genre, artist, album, track number, title, and date,
              respectively (see below for specifying these). Also, %% gives  a
              literal %.

       -X, --name-remove=s
              Remove the specified characters from parameters to the -n format
              string. This is useful to ensure legal filenames are generated.

       -P, --name-replace=s
              Replace characters removed by --name-remove with the  characters
              specified.  If  this  string  is  shorter than the --name-remove
              list, or  is  not  specified,  the  extra  characters  are  just
              removed. The default settings for this option, and the -X option
              above,  are  platform  specific  (and  chosen  to  ensure  legal
              filenames are generated for each platform).

       -c comment, --comment comment
              Add  the  string  comment as an extra comment.  This may be used
              multiple times, and all instances will be added to each  of  the
              input  files  specified.  The  argument  should  be  in the form

       -a artist, --artist artist
              Set the artist comment field in the comments to artist.

       -G genre, --genre genre
              Set the genre comment field in the comments to genre.

       -d date, --date date
              Sets the date comment field to the given value. This  should  be
              the date of recording.

       -N n, --tracknum n
              Sets the track number comment field to the given value.

       -t title, --title title
              Set the track title comment field to title.

       -l album, --album album
              Set the album comment field to album.

       -L filename, --lyrics filename
              Loads  lyrics  from filename and encodes them into a Kate stream
              multiplexed with the Vorbis stream.  Lyrics may be in LRC or SRT
              format,  and  should  be  encoded in UTF-8 or plain ASCII. Other
              encodings may be converted using tools such as iconv or  recode.
              Alternatively,  the same system as for comments will be used for
              conversion between encodings.  So called  "enhanced  LRC"  files
              are  supported,  and a simple karaoke style change will be saved
              with the lyrics. For more  complex  karaoke  setups,  kateenc(1)
              should  be  used  instead.   When  embedding lyrics, the default
              output file extention is ".oga".  Note that adding lyrics  to  a
              stream will automatically enable Skeleton (see the -k option for
              more information about Skeleton).

       -Y language-string, --lyrics-language language-string
              Sets the language for the corresponding lyrics file to language-
              string.   This  should be an ISO 639-1 language code (eg, "en"),
              or a RFC 3066 language  tag  (eg,  "en_US"),  not  a  free  form
              language  name.  Players  will typically recognize this standard
              tag and display the language name in your  own  language.   Note
              that the maximum length of this tag is 15 characters.

       Note  that  the  -a,  -t, -l, -L, and -Y  options can be given multiple
       times.  They will be applied, one to each file, in the order given.  If
       there  are  fewer album, title, or artist comments given than there are
       input files, oggenc will reuse the final one for the  remaining  files,
       and issue a warning in the case of repeated titles.


       Oggenc allows you to set a number of advanced encoder options using the
       --advanced-encode-option option. These are intended for  very  advanced
       users   only,   and   should  be  approached  with  caution.  They  may
       significantly degrade audio quality if misused. Not all  these  options
       are currently documented.

              Set the lowpass frequency to N kHz.

              Set  a  noise  floor  bias N (range from -15. to 0.) for impulse
              blocks.  A negative bias instructs the encoder  to  pay  special
              attention  to  the crispness of transients in the encoded audio.
              The tradeoff for better transient response is a higher bitrate.

              Set the allowed bitrate  maximum  for  the  encoded  file  to  N
              kilobits  per  second.   This  bitrate may be exceeded only when
              there is spare bits in the bit reservoir; if the  bit  reservoir
              is  exhausted,  frames  will  be  held  under  this value.  This
              setting must be used with --managed to have any effect.

              Set the allowed bitrate  minimum  for  the  encoded  file  to  N
              kilobits per second.  This bitrate may be underrun only when the
              bit reservoir is not full; if the bit reservoir is full,  frames
              will  be  held  over  this  value;  if it impossible to add bits
              constructively, the frame will  be  padded  with  zeroes.   This
              setting must be used with --managed to have any effect.

              Set  the  total size of the bit reservoir to N bits; the default
              size of the reservoir is equal to the  nominal  number  of  bits
              coded  in one second (eg, a nominal 128kbps file will have a bit
              reservoir of 128000 bits by default).  This option must be  used
              with  --managed  to have any effect and affects only minimum and
              maximum bitrate management.  Average bitrate  encoding  with  no
              hard bitrate boundaries does not use a bit reservoir.

              Set  the  behavior  bias of the bit reservoir (range: 0. to 1.).
              When set closer to 0, the bitrate manager attempts to hoard bits
              for  future  use  in  sudden  bitrate  increases (biasing toward
              better transient reproduction).   When  set  closer  to  1,  the
              bitrate  manager  neglects  transients  in  favor using bits for
              homogenous passages.  In the middle, the manager uses a balanced
              approach.   The  default  setting  is  .2, thus biasing slightly
              toward transient reproduction.

              Set the average bitrate for the file to N kilobits  per  second.
              When  used  without  hard minimum or maximum limits, this option
              selects reservoirless  Average  Bit  Rate  encoding,  where  the
              encoder  attempts  to  perfectly  track  a  desired bitrate, but
              imposes no strict momentary fluctuation limits.  When used along
              with  a minimum or maximum limit, the average bitrate still sets
              the average overall bitrate of the file, but  will  work  within
              the  bounds  set  by  the  bit reservoir.  When the min, max and
              average bitrates are identical,  oggenc  produces  Constant  Bit
              Rate Vorbis data.

              Set  the  reaction  time  for  the  average bitrate tracker to N
              seconds.   This  number  represents  the  fastest  reaction  the
              bitrate  tracker  is  allowed to make to hold the bitrate to the
              selected average.   The  faster  the  reaction  time,  the  less
              momentary  fluctuation  in the bitrate but (generally) the lower
              quality the audio output.  The slower  the  reaction  time,  the
              larger  the  ABR  fluctuations,  but  (generally) the better the
              audio.  When used along with min or  max  bitrate  limits,  this
              option  directly  affects  how  deep and how quickly the encoder
              will dip into its bit reservoir; the higher the number, the more
              demand on the bit reservoir.

              The  setting  must  be greater than zero and the useful range is
              approximately .05 to 10.  The default is .75 seconds.

              Disable use of channel coupling for multichannel  encoding.   At
              present,  the  encoder  will  normally  use  channel coupling to
              further increase compression with stereo and  5.1  inputs.  This
              option   forces   the  encoder  to  encode  each  channel  fully
              independently using neither lossy nor lossless coupling.


       Simplest version. Produces output as somefile.ogg:
              oggenc somefile.wav

       Specifying an output filename:
              oggenc somefile.wav -o out.ogg

       Specifying a high-quality encoding averaging 256 kbps (but still VBR):
              oggenc infile.wav -b 256 -o out.ogg

       Specifying a maximum and average bitrate, and enforcing these:
              oggenc infile.wav --managed -b 128 -M 160 -o out.ogg

       Specifying quality rather than bitrate (to a very high quality mode):
              oggenc infile.wav -q 6 -o out.ogg

       Downsampling and downmixing to 11 kHz mono before encoding:
              oggenc --resample 11025 --downmix infile.wav -q 1 -o out.ogg

       Adding some info about the track:
              oggenc  somefile.wav  -t  "The  track  title"  -a  "artist   who
              performed  this"  -l  "name of album" -c "OTHERFIELD=contents of
              some other field not explicitly supported"

       Adding embedded lyrics:
              oggenc somefile.wav --lyrics lyrics.lrc --lyrics-language en  -o

       This  encodes the three files, each with the same artist/album tag, but
       with different title tags on each one. The string given as an  argument
       to  -n  is  used  to generate filenames, as shown in the section above.
       This example gives filenames like "The Tea Party - Touch.ogg":
              oggenc -b 192 -a  "The  Tea  Party"  -l  "Triptych"  -t  "Touch"
              track01.wav  -t  "Underground"  track02.wav  -t  "Great Big Lie"
              track03.wav -n "%a - %t.ogg"

       Encoding from stdin, to stdout (you can also use  the  various  tagging
       options, like -t, -a, -l, etc.):
              oggenc -


       Program Author:
              Michael Smith <>

       Manpage Author:
              Stan Seibert <>


       Reading  type  3  Wave  files (floating point samples) probably doesn't
       work other than on Intel (or other 32 bit, little endian machines).


       vorbiscomment(1),   ogg123(1),   oggdec(1),    flac(1),    speexenc(1),
       ffmpeg2theora(1), kateenc(1)

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