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oggenc(1)				   Vorbis Tools 				oggenc(1)

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 ] [ --scale ] [ -s serial ] [ -o out-
       put_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 multi-
	      plexed 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  approxi-
	      mately  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  --man-
	      aged).

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

       --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.

       --scale
	      Input scaling factor (helps with clipping inputs).

       --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  specify-
	      ing 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 char-
	      acters 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  record-
	      ing.

       -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  set-
	      ups,  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 remain-
       ing 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 reser-
	      voir 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 pas-
	      sages.  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 bet-
	      ter 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 reser-
	      voir; 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 indepen-
	      dently 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  file-
       names,  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  "Under-
	      ground" 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)

Xiph.Org Foundation			 2008 October 05				oggenc(1)
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