Provided by: opus-tools_0.2-1build1_amd64 
NAME
opusenc - encode audio into the Opus format
SYNOPSIS
opusenc [ -h ] [ -V ] [ --help-picture ] [ --quiet ] [ --bitrate kbit/s ] [ --vbr ] [
--cvbr ] [ --hard-cbr ] [ --music ] [ --speech ] [ --comp complexity ] [ --framesize 2.5,
5, 10, 20, 40, 60 ] [ --expect-loss pct ] [ --downmix-mono ] [ --downmix-stereo ] [ --no-
phase-inv ] [ --max-delay ms ] [ --title 'track title' ] [ --artist author ] [ --album
'album title' ] [ --tracknumber 'track number' ] [ --genre genre ] [ --date YYYY-MM-DD ] [
--comment tag=value ] [ --picture filename|specification ] [ --padding n ] [ --discard-
comments ] [ --discard-pictures ] [ --raw ] [ --raw-bits bits/sample ] [ --raw-rate Hz ] [
--raw-chan N ] [ --raw-endianness flag ] [ --ignorelength ] [ --serial serial number ] [
--save-range file ] [ --set-ctl-int ctl=value ] input.wav output.opus
DESCRIPTION
opusenc reads audio data in Wave, AIFF, FLAC, Ogg/FLAC, or raw PCM format and encodes it
into an Ogg Opus stream. If the input file is "-" audio data is read from stdin.
Likewise, if the output file is "-" the Ogg Opus stream is written to stdout.
Unless quieted opusenc displays fancy statistics about the encoding progress.
OPTIONS
General options
-h, --help
Show command help
-V, --version
Show the version number
--help-picture
Show help on attaching album art
--quiet
Enable quiet mode. No messages are displayed.
Encoding options
--bitrate N.nnn
Set target bitrate in kbit/s (6-256 per channel)
In VBR mode this specifies the average rate for a large and diverse collection of
audio. In CVBR and Hard-CBR mode it specifies the specific output bitrate.
The default for input with a sample rate of 44.1 kHz or higher is 64 kbit/s per
mono stream and 96 kbit/s per coupled pair.
--vbr Use variable bitrate encoding (default)
In VBR mode the bitrate may go up and down freely depending on the content to
achieve more consistent quality.
--cvbr Use constrained variable bitrate encoding.
Outputs to a specific bitrate. This mode is analogous to CBR in AAC/MP3 encoders
and managed mode in Vorbis coders. This delivers less consistent quality than VBR
mode but consistent bitrate.
--hard-cbr
Use hard constant bitrate encoding.
With hard-cbr every frame will be exactly the same size, similar to how speech
codecs work. This delivers lower overall quality but is useful where bitrate
changes might leak data in encrypted channels or on synchronous transports.
--music
Override automatic detection and tune low bitrate encoding for music. By default,
music is detected automatically and the classification may vary over time.
Tuning impacts lower bitrates that involve tradeoffs between speech clarity and
musical accuracy, and has no impact at bitrates typically used for high quality
music encoding.
--speech
Override automatic detection and tune low bitrate encoding for speech. By default,
speech is detected automatically and the classification may vary over time.
Tuning impacts lower bitrates that involve tradeoffs between speech clarity and
musical accuracy, and has no impact at bitrates typically used for high quality
music encoding.
--comp N
Set encoding computational complexity (0-10, default: 10). Zero gives the fastest
encodes but lower quality, while 10 gives the highest quality but slower encoding.
--framesize N
Set maximum frame size in milliseconds (2.5, 5, 10, 20, 40, 60, default: 20)
Smaller framesizes achieve lower latency but less quality at a given bitrate.
Sizes greater than 20ms are only interesting at fairly low bitrates.
--expect-loss N
Set expected packet loss in percent (default: 0)
--downmix-mono
Downmix to mono
--downmix-stereo
Downmix to stereo (if >2 channels input)
--no-phase-inv
Disable use of phase inversion for intensity stereo. This trades some stereo
quality for a higher quality mono downmix, and is useful when encoding stereo audio
that is likely to be downmixed to mono after decoding.
--max-delay N
Set maximum container delay in milliseconds (0-1000, default: 1000)
Metadata options
--title title
Set the track title comment field to title
--artist artist
Set the artist comment field to artist. This may be used multiple times to list
contributing artists individually. Note that some playback software does not
display multiple artists gracefully.
--album album
Set the album or collection title field to album
--tracknumber N
Set the track number comment field to N
--date YYYY-MM-DD
Set the date comment field to YYYY-MM-DD. This may be shortened to YYYY-MM or
YYYY.
--genre genre
Set the genre comment field to genre. This option may be specified multiple times
to tag a track with multiple overlapping genres.
--comment tag=value
Add an extra comment. This may be used multiple times. The argument should be in
the form "tag=value". See the vorbis-comment specification for well known tag
names: https://www.xiph.org/vorbis/doc/v-comment.html
--picture filename|specification
Attach album art for the track.
Either a filename for the artwork or a more complete specification form can be
used. The picture is added to a METADATA_BLOCK_PICTURE comment field similar to
what is used in FLAC. The specification is a string whose parts are separated by |
(pipe) characters. Some parts may be left empty to invoke default values. Passing
a plain filename is just shorthand for the "||||filename" specification.
The format of specification is [type]|[media-
type]|[description]|[widthxheightxdepth[/colors]]|filename
type is an optional number describing the nature of the picture. Defined values
are from one of:
0: Other
1: 32x32 pixel 'file icon' (PNG only)
2: Other file icon
3: Cover (front)
4: Cover (back)
5: Leaflet page
6: Media (e.g., label side of a CD)
7: Lead artist/lead performer/soloist
8: Artist/performer
9: Conductor
10: Band/Orchestra
11: Composer
12: Lyricist/text writer
13: Recording location
14: During recording
15: During performance
16: Movie/video screen capture
17: A bright colored fish
18: Illustration
19: Band/artist logotype
20: Publisher/studio logotype
The default is 3 (front cover). More than one --picture option can be specified to
attach multiple pictures. There may only be one picture each of type 1 and 2 in a
file.
media-type is optional and is now ignored.
description is optional. The default is an empty string.
The next part specifies the resolution and color information, but is now ignored.
filename is the path to the picture file to be imported.
--padding n
Reserve n extra bytes for metadata tags. This can make later tag editing more
efficient. Defaults to 512.
--discard-comments
Don't propagate metadata tags from the input file.
--discard-pictures
Don't propagate pictures or art from the input file.
Input options
--raw Interpret input as raw PCM data without headers
--raw-bits N
Set bits/sample for raw input (default: 16)
--raw-rate N
Set sampling rate for raw input (default: 48000)
--raw-chan N
Set number of channels for raw input (default: 2)
--raw-endianness [0/1]
Set the endianness for raw input: 1 for big endian, 0 for little (default: 0)
--ignorelength
Ignore the data length in Wave headers. Opusenc automatically ignores the length
when its implausible (very small or very large) but some STDIN usage may still need
this option to avoid truncation.
Diagnostic options
--serial n
Force use of a specific stream serial number, rather than one that is randomly
generated. This is used to make the encoder deterministic for testing and is not
generally recommended.
--save-range file
Save check values for every frame to a file
--set-ctl-int x=y
Pass the encoder control x with value y (advanced). Preface with s: to direct the
ctl to multistream s
This may be used multiple times
EXAMPLES
Simplest usage. Take input as input.wav and produce output as output.opus:
opusenc input.wav output.opus
Produce a very high quality encode with a target rate of 160 kbit/s:
opusenc --bitrate 160 input.wav output.opus
Record and send a live stream to an Icecast HTTP streaming server using oggfwd:
arecord -c 2 -r 48000 -twav - | opusenc --bitrate 96 - - | oggfwd
icecast.somewhere.org 8000 password /stream.opus
NOTES
While it is possible to use opusenc for low latency streaming (e.g. with --max-delay set
to 0 and netcat instead of Icecast) it's not really designed for this, and the Ogg
container and TCP transport aren't the best tools for that application. Shell pipelines
themselves will often have high buffering. The ability to set framesizes as low as 2.5 ms
in opusenc mostly exists to try out the quality of the format with low latency settings,
but not really for actual low latency usage.
Interactive usage should use UDP/RTP directly.
AUTHORS
Gregory Maxwell <greg@xiph.org>
SEE ALSO
opusdec(1), opusinfo(1), oggfwd(1)