For testing, I want to convert an MP3 and WAV file I have to Opus, what are the steps to doing this?
ffmpeg -i infile.ext <options> outfile.opus
The audio converter shipped with the opus-tools can convert audio in raw, wave or AIFF format. The minimal syntax uses default settings:
opusenc input.wav output.opus
We may want to add a better bitrate as the default 96 kbps with the option
--bitrate N.nnn (for all options consult the manpage for opusenc).
To convert mp3 "on the fly". i.e. without creating a temporary file we can pipe the output from avconv to opusenc like this:
avconv -i input.mp3 -f wav - | opusenc --bitrate 256 - output.opus
Ubuntu 14.04 and Debian 8 ship with version 9 of
libav-tools in their repositories, and it has built-in support for Opus through the package
Example 1: Reencode an audio file as opus
With version 9 of
libopus0 installed you can simply, for example, do:
avconv -i file.mp3 -map 0:a -codec:a opus -b:a 100k -vbr on file.opus
What the options do
-i file.mp3sets the input file.
-map 0:awill select all audio streams (
a) from the input file
0. Read more about
-codec:a opusselects the opus encoder for the audio streams (
a). Read more about
-b:a 100ksets the audio's bitrate to 100 kilobit/s. Read more about
-vbr onturns on variable bitrate. This is an option specific for libopus. Here are all options for libopus:
$ avconv -h full | grep opus -A 11 avconv version 9.11-6:9.11-3+b2, Copyright (c) 2000-2013 the Libav developers built on Apr 6 2014 17:45:45 with gcc 4.8 (Debian 4.8.2-16) libopus AVOptions: -application <int> E..A. Intended application type voip E..A. Favor improved speech intelligibility audio E..A. Favor faithfulness to the input lowdelay E..A. Restrict to only the lowest delay modes -frame_duration <float> E..A. Duration of a frame in milliseconds -packet_loss <int> E..A. Expected packet loss percentage -vbr <int> E..A. Variable bit rate mode off E..A. Use constant bit rate on E..A. Use variable bit rate constrained E..A. Use constrained VBR
file.opussets the output file.
Example 2: Grab the audio from a video file and encode it as opus
Take the second stream of the first input (
-map 0:1), which is the audio stream. Encode it with libopus at 100 kbit/s with variable bitrate on:
$ avconv -stats -i linuxactionshowep309-432p.mp4 -map 0:1 -c libopus -b 100k linuxactionshowep309-432p-audio-only.opus avconv version 9.11-6:9.11-3+b2, Copyright (c) 2000-2013 the Libav developers built on Apr 6 2014 17:45:45 with gcc 4.8 (Debian 4.8.2-16) Input #0, mov,mp4,m4a,3gp,3g2,mj2, from 'linuxactionshowep309-432p.mp4': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf55.33.100 Duration: 01:14:48.45, start: 0.042667, bitrate: 466 kb/s Stream #0.0(und): Video: h264 (High), yuv420p, 768x432 [PAR 1:1 DAR 16:9], 330 kb/s, 30 fps, 30 tbr, 15360 tbn, 60 tbc Stream #0.1(und): Audio: aac, 48000 Hz, stereo, fltp, 128 kb/s Output #0, ogg, to 'linuxactionshowep309-432p-audio-only.opus': Metadata: major_brand : isom minor_version : 512 compatible_brands: isomiso2avc1mp41 encoder : Lavf54.20.3 Stream #0.0(und): Audio: libopus, 48000 Hz, stereo, flt, 100 kb/s Stream mapping: Stream #0:1 -> #0:0 (aac -> libopus) Press ctrl-c to stop encoding size= 54360kB time=4488.47 bitrate= 99.2kbits/s video:0kB audio:53875kB global headers:0kB muxing overhead 0.900602%
With the package
$ mediainfo linuxactionshowep309-432p-audio-only.opus General Complete name : linuxactionshowep309-432p-audio-only.opus Format : OGG File size : 53.1 MiB Duration : 1h 14mn Overall bit rate : 99.2 Kbps Writing application : Lavf54.20.3 major_brand : isom minor_version : 512 compatible_brands : isomiso2avc1mp41 Audio ID : 2104437746 (0x7D6F2BF2) Format : Opus Duration : 1h 14mn Channel(s) : 2 channels Channel positions : Front: L R Sampling rate : 48.0 KHz Compression mode : Lossy Writing library : Lavf54.20.3
Opus on 12.04
On 12.04 (Precise), however, there are dependency problems with installing the opus codecs and tools, so I have found by far the best solution is the one that has become available very recently: compile the opus audio encoder and decoder as noted here, and build
ffmpeg with opus support by adding
--enable-opus to the configure options of
ffmpeg (as listed on the compilation guide).
I know that
ffmpeg is deprecated in Ubuntu in favour of
Libav, but compiling is a good way to get a fully functioning opus encoder/decoder integrated into
ffmpeg itself. You can then use it to convert files (first to wav) and then to
.opus. The documentation installed with libopus and ffmpeg will reveal all the options that can be used to convert files.
When converting files with
ffmpeg after compilation, you must specify
-acodec libopus or
ffmpeg will not use the opus codec:
ffmpeg -i pc.wav -ar 48000 -ac 2 -acodec libopus -ab 256k man.opus
You can then test the file created with
There's no need to reproduce the guide here in its entirety, but it's worth noting one or two things:
You should first install the dependencies as listed (I omit
yasmfrom the list: see my second point):
sudo apt-get -y install autoconf build-essential checkinstall git libass-dev libfaac-dev libgpac-dev libjack-jackd2-dev libmp3lame-dev libopencore-amrnb-dev libopencore-amrwb-dev librtmp-dev libsdl1.2-dev libtheora-dev libtool libva-dev libvdpau-dev libvorbis-dev libx11-dev libxext-dev libxfixes-dev pkg-config texi2html zlib1g-dev
There is one issue that should be pointed out: the git build seems to want
yasm-1.2, and that is not available, so you have to compile the source from the official site, but it is simple. Just remove any installed versions of
yasm, then unpack the downloaded archive,
cdto the folder, run
./configure && makeand then
sudo checkinstall. If any other builds require the earlier version, you can just remove this version and install the repository version.
It is necessary to remove any existing
fdk-aacpackages before you start compiling.
It is critical that you compile and install
opusbefore you build
ffmpeg, as those libraries will be used in the build.
Do not forget to add
--enable-opusto the configure options when you run the
The version of opus compiled was 1.1alpha, so you may need to re-compile the opus library and ffmpeg in the future again when a new version is released.
You can use
ffplayto play any opus files you create.
That's how I do it:
- First, open a terminal in the same directory where your audio files are.
- Then, type this command:
$ opusenc --bitrate 320 --max-delay 10 "18 - Soul Asylum - Runaway Train (Album Version).flac" "18 - Soul Asylum - Runaway Train (Album Version).opus"
$ opusenc --bitrate 510 --max-delay 10 "18 - Soul Asylum - Runaway Train (Album Version).flac" "18 - Soul Asylum - Runaway Train (Album Version).opus"
No need to specify
--maxdelay 10 option because
opusenc do this by default.
Console Output for this file conversion (
Encoding using libopus 1.1.2 (audio) ----------------------------------------------------- Input: 44.1kHz 2 channels Output: 2 channels (2 coupled) 20ms packets, 320kbit/sec VBR Preskip: 356 Encoding complete ----------------------------------------------------- Encoded: 4 minutes and 22.4 seconds Runtime: 8 seconds (32.8x realtime) Wrote: 10955530 bytes, 13120 packets, 13124 pages Bitrate: 317.691kbit/s (without overhead) Instant rates: 1.2kbit/s to 510.4kbit/s (3 to 1276 bytes per packet) Overhead: 4.89% (container+metadata)
It's super fast! Less than 8 seconds with a complexity of 10 (Encoding computational complexity (0-10, default: 10). Zero gives the fastest encodes but lower quality, while 10 gives the highest quality but slower encoding) and a maximum delay time of 10ms (Maximum container delay in milliseconds (0-1000, default: 1000)), so if you skip time in a song, the clipping effect will have a duration of 10ms so it is inperceptible (try with 1000 and hear the difference skipping time with your mouse). Bitrate is VBR by default. 320kbps worked for me so is optional, play with this number:
--bitrate N.nnn => Target bitrate in kbit/sec (6-256 per channel)
By the way, encoding from MP3 to OPUS is not a good idea, it's not going to sound better, their compression algorithms are way too different. But from FLAC or WAV or any other Lossless Audio Format, that's another story.
Note: To encode another file, just press the Up Arrow in the same terminal to call the last command and change the name of the input and output files.
If you are looking for a ffmpeg/avconv GUI, maybe TraGtor is what you need.
If only the mp3 to opus route is needed,
mpg123 can do the decoding to wav/pcm.
mpg123 -w - input.mp3 | opusenc - output.opus
For the unfamiliar the dash "-" functions as stdout on the left to be piped to opusencs stdin on the right.
Of course ffmpeg is excellent for general media conversion and editing, but its install size and usual distribution dependencies also have a bigger footprint.
- Find a CD (or flac with CDEMU) with the same content as mp3.
Open Audex and add a new profile called Opus, add Command pattern;
opusenc $i --comment="TRACKNUMBER="$trackno"" --artist "$artist" --album "$title" --title "$ttitle" --date "$date" --picture "$cover" $o
- let it rip!