opusenc - encode audio into the Opus format
opusenc [ -h ] [ -V ] [ --bitrate kbit/sec ] [ --vbr ] [ --cvbr ] [ --hard-cbr ] [ --comp complexity ] [ --framesize 2.5, 5, 10, 20, 40, 60 ] [ --expect-loss pct ] [ --downmix-mono ] [ --downmix-stereo ] [ --max-delay ms ] [ --serial serial number ] [ --save-range file ] [ --set-ctl-int ctl=value ] [ --comment tag=value ] [ --artist author ] [ --title 'track title' ] [ --album 'album title' ] [ --date YYYY-MM-DD ] [ --genre genre ] [ --raw ] [ --raw-bits bits/sample ] [ --raw-rate Hz ] [ --raw-chan N ] [ --raw-endianness flag ] [ --ignorelength ] input.wav output.opus
opusenc reads audio data in either raw, Wave, or AIFF format and encodes it into an Opus stream. If the input file is "-" audio data is read from stdin. Likewise, if the output file is "-" opus data is written to stdout.
Unless quieted opusenc displays fancy statistics about the encoding progress.
Show command help
Show the version number
Target bitrate in kbit/sec (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.
Default for >=44.1kHz input is 64kbps per mono stream, 96kbps per coupled pair.
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.
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.
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.
Encoding computational complexity (0-10, default: 10). Zero gives the fastest encodes but lower quality, while 10 gives the highest quality but slower encoding.
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.
Percentage packet loss to expect (default: 0)
Downmix to mono
Downmix to stereo (if >2 channels input)
Maximum container delay in milliseconds (0-1000, default: 1000)
Forces the stream serial number to a specified value (instead of being random). This is used to make the encoder deterministic for testing and is not generally recommended.
Saves check values for every frame to a file
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
Add 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". See the vorbis-comment specification for well known tag names: http://www.xiph.org/vorbis/doc/v-comment.html
Set the artist comment field to artist
Set the track title comment field to title
Set the album or collection title field to album
Set the date comment field to YYYY-MM-DD
Set the genre comment field to genre
Raw (headerless) PCM input
Set bits/sample for raw input (default: 16)
Set sampling rate for raw input (default: 48000)
Set number of channels for raw input (default: 2)
Set the endianness for raw input: 1 for bigendian, 0 for little (defaults to 0)
Always ignore the datalength 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.
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 160kbps:
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 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.
Gregory Maxwell <greg@xiph.org>
opusdec (1) opusinfo (1) oggfwd (1)
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