Provided by: vorbis-tools_1.4.0-1ubuntu2_amd64 bug

NAME

       oggenc - encode audio into the Ogg Vorbis format

SYNOPSIS

       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 ...

DESCRIPTION

       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).

OPTIONS

       -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 this.

       -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.

       --utf8
              Informs oggenc that the Vorbis Comments are already encoded as  UTF-8.   Useful  in
              situations where the shell is using some other encoding.

       -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.

       --ignorelength
              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).

       --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 streaming.

       -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
              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 details.

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

       --discard-comments
              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 "tag=value".

       -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.

ADVANCED ENCODER OPTIONS

       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.

       lowpass_frequency=N
              Set the lowpass frequency to N kHz.

       impulse_noisetune=N
              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.

       bitrate_hard_max=N
              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.

       bitrate_hard_min=N
              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.

       bit_reservoir_bits=N
              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.

       bit_reservoir_bias=N
              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.

       bitrate_average=N
              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.

       bitrate_average_damping=N
              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_coupling
              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.

EXAMPLES

       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 out.oga

       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 -

AUTHORS

       Program Author:
              Michael Smith <msmith@xiph.org>

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

BUGS

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

SEE ALSO

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