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 <msmith@xiph.org>

   Manpage Author:
          Stan Seibert <indigo@aztec.asu.edu>


   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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