Provided by: vorbis-tools_1.4.0-11_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
              extension  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)