The Opus audio codec looks like the best codec so far for compressing audio. It has recently become supported in the latest ffmpeg and VLC players. However, there is no documentation I can find on how to actually encode media with it. Can someone please direct me to said docs, preferably with specifics to ffmpeg flags and usage? I have a lot of audiobooks that are taking up far too much space and Opus looks to be the perfect format to keep them in.
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4You did not mention your OS. – llogan Dec 09 '12 at 17:48
3 Answers
ffmpeg -i <input> -c:a libopus -b:a bitrate <output>
The bitrate can be given like -b:a 96K for 96 kBit/s.
The ffmpeg documentation has a list of options and descriptions for libopus. Here you can set -vbr options or a different -compression_level.
Make sure you compiled ffmpeg with --enable-libopus!
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1This is great! I had figured out how to encode with ffmpeg, but didn't know about the -vbr and --compression_level flags. This is exactly what I was looking for! – MrDrMcCoy Dec 05 '13 at 20:36
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11This helped me. NB, `vbr on` and `compression_level 10` are defaults and can probably be omitted. https://ffmpeg.org/ffmpeg-codecs.html#Option-Mapping – Joshua Huber Oct 24 '17 at 18:12
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3Indeed, these are the default values, and in most cases they don't have to be modified. Also, note the bitrate is in bits/s, not the usual kbits/s. The default bitrate is 96000 (96 kbits/s), which is (of course arguably) a fine value. Sample command to summarize: `ffmpeg -i input.flac -acodec libopus -b:a 128000 output.opus` – Gras Double Aug 18 '18 at 15:14
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Here is how to use the 'application' parameter `-application voip` . As a new ffmpeg user I struggled a bit to find the right syntax. – tuxayo Apr 26 '19 at 20:16
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Is the `vbr` option broken? I have to specify the bitrate, otherwise, it complains that no bitrate is given and the lowest will be selected. `vbr` should decide on the bitrate itself. – Minh Nghĩa Dec 11 '19 at 11:20
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@MinhNghĩa vbr means it will try to get the best quality for a given bitrate by dynamically adjusting the bitrate. It still needs to be told what average bitrate it should target. – localhost Mar 17 '21 at 04:52
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@localhost Do you have a source for that? AFAIK, `vbr` mode for opus (and might be all other codec) uses a quality `q` slider, or something like that. The mode you describe is ABR. – Minh Nghĩa Mar 17 '21 at 10:56
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2@MinhNghĩa https://en.wikipedia.org/wiki/Variable_bitrate what you describe with `q` is `fixed quality` mode. I was describing `average bitrate` which I presume libopus is using here. They are both forms of `vbr`.Obviously in this answer we are not defining a `q` value, so it cannot be using `fixed quality` mode. I don't even see that libopus can operate in a `fixed quality` mode at all at http://ffmpeg.org/ffmpeg-codecs.html#libopus-1 . You must be thinking of a different codec. – localhost Mar 17 '21 at 22:16
Download Opus-tools
Encode:
opusenc --bitrate 64 What_A_Feeling.wav What_A_Feeling_64.opusDecode: (to play in any media player, useful if your media player does not support opus yet):
opusdec What_A_Feeling_64.opus What_A_Feeling_opus64.wav
(What_A_Feeling is a song name)
Detailed options displayed when running opusenc by itself:
Usage: opusenc [options] input_file output_file.opus
Encodes input_file using Opus.
It can read the WAV, AIFF, FLAC, Ogg/FLAC, or raw files.
General options:
-h, --help This help
-V, --version Version information
--quiet Quiet mode
input_file can be:
filename.wav file
- stdin
output_file can be:
filename.opus compressed file
- stdout
Encoding options:
--bitrate n.nnn Target bitrate in kbit/sec (6-256/channel)
--vbr Use variable bitrate encoding (default)
--cvbr Use constrained variable bitrate encoding
--hard-cbr Use hard constant bitrate encoding
--comp n Encoding complexity (0-10, default: 10 (slowest))
--framesize n Maximum frame size in milliseconds
(2.5, 5, 10, 20, 40, 60, default: 20)
--expect-loss Percentage packet loss to expect (default: 0)
--downmix-mono Downmix to mono
--downmix-stereo Downmix to stereo (if >2 channels)
--max-delay n Maximum container delay in milliseconds
(0-1000, default: 1000)
Diagnostic options:
--serial n Forces a specific stream serial number
--save-range file Saves 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
Metadata options:
--comment Add the given string as an extra comment
This may be used multiple times
--artist Author of this track
--title Title for this track
--album Album or collection this track belongs to
--date Date for this track
--genre Genre for this track
--picture Album art for this track
More than one --picture option can be specified.
Either a FILENAME for the picture file or a more
complete SPECIFICATION form can be used. The
SPECIFICATION is a string whose parts are
separated by | (pipe) characters. Some parts may
be left empty to invoke default values. A
FILENAME is just shorthand for "||||FILENAME".
The format of SPECIFICATION is
[TYPE]|[MIME-TYPE]|[DESCRIPTION]|[WIDTHxHEIGHT
xDEPTH[/COLORS]]|FILENAME
TYPE is an optional number 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). There may only be
one picture each of type 1 and 2 in a file.
MIME-TYPE is optional. If left blank, it will be
detected from the file. For best compatibility
with players, use pictures with a MIME-TYPE of
image/jpeg or image/png. The MIME-TYPE can also
be --> to mean that FILENAME is actually a URL to
an image, though this use is discouraged. The
file at the URL will not be fetched. The URL
itself is stored in the metadata.
DESCRIPTION is optional. The default is an empty
string.
The next part specifies the resolution and color
information. If the MIME-TYPE is image/jpeg,
image/png, or image/gif, you can usually leave
this empty and they can be detected from the
file. Otherwise, you must specify the width in
pixels, height in pixels, and color depth in
bits-per-pixel. If the image has indexed colors
you should also specify the number of colors
used. If possible, these are checked against the
file for accuracy.
FILENAME is the path to the picture file to be
imported, or the URL if the MIME-TYPE is -->.
--padding n Extra bytes to reserve for metadata (default: 512)
--discard-comments Don't keep metadata when transcoding
--discard-pictures Don't keep pictures when transcoding
Input options:
--raw Raw input
--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 n 1 for bigendian, 0 for little (defaults to 0)
--ignorelength Always ignore the datalength in Wave headers
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Thanks, but I was hoping for something a bit more detailed. Are there variable bit rate options? What other flags can I pass to optimize the audio stream? – MrDrMcCoy Dec 09 '12 at 21:56
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2@nakedhitman The `--vbr` option is default. See the man page for opusenc. You can also encode with ffmpeg but it needs to be compiled with `--enable-libopus` after installing [libopus](http://www.opus-codec.org/downloads/). – llogan Dec 09 '12 at 23:10
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For getting an idea (not for batch conversion), an online version [like this](https://blog.rillke.com/opusenc.js/) might be helpful. In case one's plan is to convert an entire collection, opusinfo (also part of `opus-tools`) might tell, whether the chosen encoding tool performed well. – Rainer Rillke Sep 16 '17 at 23:11
The best Opus Codec commands in DOS/Windows Command would be:
For lib 1.3, smallest filesize (tape quality):
Opusenc a.wav a.opus --bitrate 24 --framesize 40 --discard-comments --discard-pictures
You will:
a- at 24kbits still have a 16kHz (32kHz stereo) sonic output, on a super small size. Any smaller, and Sonic output will drop to 11kHz
b- Increase framesize (save a few bits of data (about 6%), without perceivable loss in quality)
c- Discard unnecessary TAG info, and cover pictures that will only enlarge file size.
For lib 1.3, streaming (near CD quality) use '--bitrate 52' instead of '24'.
The default in 1.2 is 48kbits, but using 52 in 1.3, with 'framesize' set to 40, your file size is equal, but there are much less artifacts audible.
52 kbits with framesize 40, is arguably equal in quality to 160kbits MP3, at less than 1/3rd the size; while 48kbits on 1.2 equals 128kbits MP3, or only 2.5x smaller in size.
For near identical quality, you'll need between 80 to 96kbits. I hardly ever use this, as I want my audio as small as possible.
Over 96kbits, is only good for editing, and one can't audibly hear the difference between the original and the Opus encoded file.
That being said, I don't yet know how it all works in Linux.
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I got an error trying that one on both linux and windows: `Unrecognized option '-framesize'. Error splitting the argument list: Option not found`. ffmpeg version is 4.0.2 – sukhmel Feb 20 '19 at 08:55
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