Ubuntu Manpage: oggenc - encode audio into the Ogg Vorbis format

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).