opusenc(1) opus-tools opusenc(1)

opusenc – encode audio into the Opus format

opusenc [ options ] input_file output.opus

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 statistics about the encoding progress.

-h, --help
Show command help.
-V, --version
Show version information.
--help-picture
Show help on attaching album art.
--quiet
Enable quiet mode. No messages are displayed.

--bitrate N
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 a specific bitrate. This mode is analogous to CBR in AAC and 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 20 ms 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 multichannel speaker configurations to stereo.
--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).

--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.
--genre GENRE
Set the genre comment field to GENRE. This option may be used multiple times to tag a track with multiple overlapping genres.
--date YYYY-MM-DD
Set the date comment field to YYYY-MM-DD. This may be shortened to YYYY-MM or YYYY.
--tracknumber N
Set the track number comment field to N.
--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 <https://www.xiph.org/vorbis/doc/v-comment.html> for well known tag names.
--picture FILENAME | SPECIFICATION
Attach album art for the track. JPEG and PNG image formats are accepted. 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. Except for the filename all parts are optional. A plain FILENAME is equivalent to a ||||FILENAME specification.
The format of SPECIFICATION is: [TYPE]|[MEDIATYPE]|[DESCRIPTION]|[DIMENSIONS]|FILENAME
TYPE is a number denoting the nature of the picture (default 3):
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
There may only be one picture each of type 1 and 2 in a file.
The default DESCRIPTION is an empty string. FILENAME is the path to the picture file to be imported. MEDIATYPE and DIMENSIONS are obtained from the file and any specified values are ignored.
More than one --picture option can be specified to attach multiple pictures.
--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.

--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. The length will always be ignored when it is implausible (very small or very large), but some stdin usage may still need this option to avoid truncation.

--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 FILENAME
Save check values for every frame to a file.
--set-ctl-int [S:]X=Y
Pass the encoder control X with value Y (advanced). Preface with S: to direct the ctl to multistream stream number S. This may be used multiple times.

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

While it is possible to use opusenc for low latency streaming (e.g. with --max-delay 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.

Gregory Maxwell <greg@xiph.org>

opusdec(1), opusinfo(1), oggfwd(1)
2019-09-07 Xiph.Org Foundation