Provided by: ffmpegfs_1.10-2_amd64 
NAME
ffmpegfs - mounts and transcodes a multitude of formats to one of the target formats on the fly
SYNOPSIS
ffmpegfs [OPTION]... IN_DIR OUT_DIR
DESCRIPTION
The ffmpegfs(1) command will mount the directory IN_DIR on OUT_DIR. Thereafter, accessing OUT_DIR will
show the contents of IN_DIR, with all supported media files transparently renamed and transcoded to one
of the supported target formats upon access.
Supported output formats:
• MP4 (MPEG-4)
• WebM
• OGG
• MOV (QuickTime File Format)
• Prores (a MOV container for Apple Prores video & PCM audio)
• Opus (audio only)
• MP3 (MPEG-2 Audio Layer III)
• WAV (Waveform Audio File Format)
• AIFF (Audio Interchange File Format)
• ALAC (Apple Lossless Audio Codec)
OPTIONS
Usage: ffmpegfs [OPTION]... IN_DIR OUT_DIR
Mount IN_DIR on OUT_DIR, converting audio/video files upon access.
Encoding options
--desttype=TYPE, -odesttype=TYPE
Select destination format. TYPE can currently be:
MP4, MP3, OGG, WEBM, MOV, ProRes, AIFF, ALAC, OPUS or WAV. To stream videos, MP4, OGG, WEBM or
MOV/ProRes must be selected.
To use the smart transcoding feature, specify a video and audio file type, separated by a "+" sign.
For example, --desttype=mov+aiff will convert video files to Apple Quicktime MOV and audio only files
to AIFF.
Default: mp4
--autocopy=OPTION, -oautocopy=OPTION
Select auto copy option, OPTION can be:
┌────────────┬───────────────────────────────────────┐
│OFF │ Never copy streams, transcode always. │
├────────────┼───────────────────────────────────────┤
│MATCH │ Copy stream if target supports codec. │
├────────────┼───────────────────────────────────────┤
│MATCHLIMIT │ Same as MATCH, only copy if target │
│ │ not larger, transcode otherwise. │
├────────────┼───────────────────────────────────────┤
│STRICT │ Copy stream if codec matches desired │
│ │ target, transcode otherwise. │
├────────────┼───────────────────────────────────────┤
│STRICTLIMIT │ Same as STRICT, only copy if target │
│ │ not larger, transcode otherwise. │
└────────────┴───────────────────────────────────────┘
This can speed up transcoding significantly as copying streams uses much less computing power as
compared to transcoding.
MATCH copies a stream if the target supports it, e.g. an AAC audio stream will be copied to MPEG
although ffmepeg’s target format is MP3 for this container. H264 would be copied to ProRes although
the result will be a regular MOV/MP4, not a ProRes file.
STRICT would convert AAC to MP3 for MPEG or H264 to ProRes for Prores files to strictly adhere to the
output format setting. This will create homogenous results which might prevent problems with picky
playback software.
Default: OFF
--profile=NAME, -oprofile=NAME
Set profile for target audience, NAME can be:
┌────────┬───────────────────────────────────┐
│NONE │ no profile │
├────────┼───────────────────────────────────┤
│FF:: │ optimise for Firefox │
├────────┼───────────────────────────────────┤
│EDGE │ optimise for MS Edge and Internet │
│ │ Explorer > 11 │
├────────┼───────────────────────────────────┤
│IE │ optimise for MS Edge and Internet │
│ │ Explorer ⇐ 11 │
├────────┼───────────────────────────────────┤
│CHROME │ Google Chrome │
├────────┼───────────────────────────────────┤
│SAFARI │ Apple Safari │
├────────┼───────────────────────────────────┤
│OPERA │ Opera │
├────────┼───────────────────────────────────┤
│MAXTHON │ Maxthon │
└────────┴───────────────────────────────────┘
Default: NONE
--level=NAME, -o level=NAME
Set level for output if available, NAME can be:
┌─────────┬─────────────────┐
│PROXY │ Proxy – apco │
├─────────┼─────────────────┤
│LT │ LT – apcs │
├─────────┼─────────────────┤
│STANDARD │ standard – apcn │
├─────────┼─────────────────┤
│HQ │ HQ - apch │
└─────────┴─────────────────┘
Default: HQ
Audio Options
--audiobitrate=BITRATE, -o audiobitrate=BITRATE
Audio encoding bitrate.
Default: 128 kbit
Acceptable values for BITRATE:
mp4: 8, 16, 24, 32, 40, 48, 56, 64, 80, 96, 112, 128, 144, 160, 176, 192, 224, 256, 288, 320, 352,
384, 416 and 448 kbps.
mp3: For sampling frequencies of 32, 44.1, and 48 kHz, BITRATE can be among 32, 40, 48, 56, 64, 80,
96, 112, 128, 160, 192, 224, 256, and 320 kbps.
For sampling frequencies of 16, 22.05, and 24 kHz, BITRATE can be among 8, 16, 24, 32, 40, 48, 56,
64, 80, 96, 112, 128, 144, and 160 kbps.
When in doubt, it is recommended to choose a bitrate among 96, 112, 128, 160, 192, 224, 256, and 320
kbps.
BITRATE
can be defined as...
• n bit/s: # or #bps
• n kbit/s: #K or #Kbps
• n Mbit/s: #M or #Mbps
--audiosamplerate=SAMPLERATE, -o audiosamplerate=SAMPLERATE
Limits the output sample rate to SAMPLERATE. If the source file sample rate is more it will be
downsampled automatically.
Typical values are 8000, 11025, 22050, 44100, 48000, 96000, 192000.
If the target codec does not support the selected sample rate, the next matching rate will be chosen
(e.g. if 24K is selected ut only 22.05 or 44.1 KHz supported, 22.05 KHz will be set).
Set to 0 to keep source rate.
Default: 44.1 kHz
SAMPLERATE
can be defined as...
• In Hz: # or #Hz
• In kHz: #K or #KHz
Video Options
--videobitrate=BITRATE, -o videobitrate=BITRATE
Video encoding bit rate. Setting this too high or low may cause transcoding to fail.
Default: 2 Mbit
mp4: May be specified as 500 to 25000 kbit.
BITRATE
can be defined as...
• n bit/s: # or #bps
• n kbit/s: #K or #Kbps
• n Mbit/s: #M or #Mbps
--videoheight=HEIGHT, -o videoheight=HEIGHT
Sets the height of the transcoded video.
When the video is rescaled the aspect ratio is preserved if --width is not set at the same time.
Default: keep source video height
--videowidth=WIDTH, -o videowidth=WIDTH
Sets the width of the transcoded video.
When the video is rescaled the aspect ratio is preserved if --height is not set at the same time.
Default: keep source video width
--deinterlace, -o deinterlace
Deinterlace video if necessary while transcoding.
May need higher bit rate, but will increase picture quality when streaming via HTML5.
Default: no deinterlace
Album Arts
--noalbumarts, -o noalbumarts
Do not copy album arts into output file.
This will reduce the file size, may be useful when streaming via HTML5 when album arts are not used
anyway.
Default: add album arts
Virtual Script
--enablescript, -o enablescript
Add virtual index.php to every directory. It reads scripts/videotag.php from the ffmpegs binary
directory.
This can be very handy to test video playback. Of course, feel free to replace videotag.php with your
own script.
Default: Do not generate script file
--scriptfile, -o scriptfile
Set the name of the virtual script created in each directory.
Default: index.php
--scriptsource, -o scriptsource
Take a different source file.
Default: scripts/videotag.php
Cache Options
--expiry_time=TIME, -o expiry_time=TIME
Cache entries expire after TIME and will be deleted to save disk space.
Default: 1 week
--max_inactive_suspend=TIME, -o max_inactive_suspend=TIME
While being accessed the file is transcoded to the target format in the background. When the client
quits transcoding will continue until this time out. Transcoding is suspended until it is accessed
again, then transcoding will continue.
Default: 15 seconds
--max_inactive_abort=TIME, -o max_inactive_abort=TIME
While being accessed the file is transcoded in the background to the target format. When the client
quits transcoding will continue until this time out, then the transcoder thread quits.
Default: 30 seconds
--prebuffer_size=SIZE, -o prebuffer_size=SIZE
Files will be decoded until the buffer contains this much bytes allowing playback to start smoothly
without lags.
Set to 0 to disable pre-buffering.
Default: 100 KB
--max_cache_size=SIZE, -o max_cache_size=SIZE
Set the maximum diskspace used by the cache. If the cache would grow beyond this limit when a file is
transcoded, old entries will be deleted to keep the cache within the size limit.
Default: unlimited
--min_diskspace=SIZE, -o min_diskspace=SIZE
Set the required diskspace on the cachepath mount. If the remaining space would fall below SIZE when
a file is transcoded, old entries will be deleted to keep the diskspace within the limit.
Default: 0 (no minimum space)
--cachepath=DIR, -o cachepath=DIR
Sets the disk cache directory to DIR. Will be created if not existing. The user running ffmpegfs must
have write access to the location.
--disable_cache, -o disable_cache
Disable the cache functionality.
Default: enabled
--cache_maintenance=TIME, -o cache_maintenance=TIME
Starts cache maintenance in TIME intervals. This will enforce the expery_time, max_cache_size and
min_diskspace settings. Do not set too low as this can slow down transcoding.
Only one ffmpegfs process will do the maintenance by becoming the master. If that process exits,
another will take over so that always one will do the maintenance.
Default: 1 hour
--prune_cache
Prune cache immediately according to the above settings.
--clear_cache, -o clear_cache
Clear cache on startup. All previously recoded files will be deleted.
TIME
can be defined as...
• Seconds: #
• Minutes: #m
• Hours: #h
• Days: #d
• Weeks: #w
SIZE
can be defined as...
• In bytes: # or #B
• In KBytes: #K or #KB
• In MBytes: #M or #MB
• In GBytes: #G or #GB
• In TBytes: #T or #TB
Other
--max_threads=COUNT, -o max_threads=COUNT
Limit concurrent transcoder threads. Set to 0 for unlimited threads. Recommended values are up to 16
times number of CPU cores.
Default: 16 times number of detected cpu cores
--decoding_errors, -o decoding_errors
Decoding errors are normally ignored, leaving bloopers and hiccups in encoded audio or video but yet
creating a valid file. When this option is set, transcoding will stop with an error.
Default: Ignore errors
--min_dvd_chapter_duration=SECONDS, -o min_dvd_chapter_duration=SECONDS
Ignores DVD chapters shorter than SECONDS. Set to 0 to disable. This avoids transcoding errors for
DVD chapters too short to detect its streams.
Default: 1 second
--win_smb_fix, -o win_smb_fix
Windows seems to access the files on Samba drives starting at the last 64K segment simply when the
file is opened. Setting --win_smb_fix=1 will ignore these attempts (not decode the file up to this
point).
Default: off
Logging
--log_maxlevel=LEVEL, -o log_maxlevel=LEVEL
Maximum level of messages to log, either ERROR, WARNING, INFO, DEBUG or TRACE. Defaults to INFO, and
always set to DEBUG in debug mode.
Note that the other log flags must also be set to enable logging.
--log_stderr, -o log_stderr
Enable outputting logging messages to stderr. Automatically enabled in debug mode.
--log_syslog, -o log_syslog
Enable outputting logging messages to syslog.
--logfile=FILE, -o logfile=FILE
File to output log messages to. By default, no file will be written.
General/FUSE options
-d, -o debug
Enable debug output. This will result in a large quantity of diagnostic information being printed to
stderr as the program runs. It implies -f.
-f
Run in foreground instead of detaching from the terminal.
-h, --help
Print usage information.
-V, --version
Output version information.
-s
Force single-threaded operation.
USAGE
Mount your filesystem like this:
ffmpegfs [--audiobitrate bitrate] [--videobitrate bitrate] musicdir mountpoint [-o fuse_options]
For example,
ffmpegfs --audiobitrate 256K -videobitrate 2000000 /mnt/music /mnt/ffmpegfs -o allow_other,ro
In recent versions of FUSE and FFmpegfs, the same can be achieved with the following entry in /etc/fstab:
ffmpegfs#/mnt/music /mnt/ffmpegfs fuse allow_other,ro,audiobitrate=256K,videobitrate=2000000 0 0
Another (more modern) form of this command:
/mnt/music /mnt/ffmpegfs fuse.ffmpegfs allow_other,ro,audiobitrate=256K,videobitrate=2000000 0 0
At this point files like /mnt/music/**.flac and /mnt/music/**.ogg will show up as /mnt/ffmpegfs/**.mp4.
Note that the "allow_other" option by default can only be used by root. You must either run FFmpegfs as
root or better add a "user_allow_other" key to /etc/fuse.conf.
"allow_other" is required to allow any user access to the mount, by default this is only possible for the
user who launched FFmpegfs.
HOW IT WORKS
When a file is opened, the decoder and encoder are initialised and the file metadata is read. At this
time the final filesize can be determined approximately. This works well for mp3 output files, but only
fair to good for mp4.
As the file is read, it is transcoded into an internal per-file buffer. This buffer continues to grow
while the file is being read until the whole file is transcoded in memory. Once decoded the file is kept
in a disk buffer and can be accessed very fast.
Transcoding is done in an extra thread, so if other processes should access the same file they will share
the same transcoded data, saving CPU time. If the first process abandons the file before its end,
transconding will continue for some time. If the file is accessed again before the timeout, transcoding
will go on, if not it stops and the chunk created so far discarded to save disk space.
Seeking within a file will cause the file to be transcoded up to the seek point (if not already done).
This is not usually a problem since most programs will read a file from start to finish. Future
enhancements may provide true random seeking (But if this is feasible is yet unclear due to restrictions
to positioning inside compressed streams).
mp3: ID3 version 2.4 and 1.1 tags are created from the comments in the source file. They are located at
the start and end of the file respectively.
mp4: Same applies to meta atoms in mp4 containers.
mp3 target only: A special optimisation is made so that applications which scan for id3v1 tags do not
have to wait for the whole file to be transcoded before reading the tag. This dramatically speeds up such
applications.
SUPPORTED OUTPUT FORMATS
A few words to the supported output formats which are mp3 and mp4 currently. There is not much to say
about the mp3 output as these are regular mp3 files with no strings attached. They should play well in
any modern player.
The mp4 files created are special, though, as mp4 is not quite suited for live streaming. Reason being
that the start block of an mp4 contains a field with the size of the compressed data section. Suffice to
say that this field cannot be filled in until the size is known, which means compression must be
completed first, a seek done to the beginning, and the size atom updated.
Alas, for a continuous live stream, that size will never be known or for our transcoded files one would
have to wait for the whole file to be recoded. If that was not enough some important pieces of
information are located at the end of the file, including meta tags with artist, album, etc.
Subsequently many applications will go to the end of an mp4 to read important information before going
back to the head of the file and start playing. This will break the whole transcode-on-demand idea of
FFmpegfs.
To get around the restriction several extensions have been developed, one of which is called "faststart"
that relocates the afformentioned data from the end to the beginning of the mp4. Additionally, the size
field can be left empty (0). isml (smooth live streaming) is another extension.
For direct to stream transcoding several new features in mp4 need to be used (ISMV, faststart,
separate_moof/empty_moov to name them) which are not implemented in older versions (or if available, not
working properly).
By default faststart files will be created with an empty size field so that the file can be started to be
written out at once instead of decoding it as a whole before this is possible. That would mean it would
take some time before playback can start.
The data part is divided into chunks of about 5 seconds length each, this allowing to fill in the size
fields early enough.
As a draw back not all players support the format, or play back with strange side effects. VLC plays the
file, but updates the time display every 5 seconds only. When streamed over HTML5 video tags, there will
be no total time shown, but that is OK, as it is yet unknown. The playback cannot be positioned past the
current playback position, only backwards.
But that’s the price of starting playback, fast.
So there is a lot of work to be put into mp4 support, still.
The output format must be selectable for the desired audience, for streaming or opening the files
locally, for example.
DEVELOPMENT
FFmpegfs uses Git for revision control. You can obtain the full repository with:
git clone https://github.com/nschlia/ffmpegfs.git
FFmpegfs is written in a mixture of C and C++ and uses the following libraries:
• FUSE
If using the FFmpeg support (Libav works as well, but FFmpeg is recommended):
• FFmpeg or Libav
FUTURE PLANS
• Create a windows version
• Add DVD/Bluray support
FILES
/usr/local/bin/ffmpegfs, /etc/fstab
AUTHORS
This fork with FFmpeg support is maintained by Norbert Schlia since 2017.
Based on work by K. Henriksson (from 2008 to 2017) and the original author David Collett (from 2006 to
2008).
Much thanks to them for the original work!
LICENSE
This program can be distributed under the terms of the GNU GPL version 3 or later. It can be found online
or in the COPYING file.
This file and other documentation files can be distributed under the terms of the GNU Free Documentation
License 1.3 or later. It can be found online or in the COPYING.DOC file.
FFMPEG LICENSE
FFmpeg is licensed under the GNU Lesser General Public License (LGPL) version 2.1 or later. However,
FFmpeg incorporates several optional parts and optimizations that are covered by the GNU General Public
License (GPL) version 2 or later. If those parts get used the GPL applies to all of FFmpeg.
See https://www.ffmpeg.org/legal.html for details.
COPYRIGHT
This fork with FFmpeg support copyright (C) 2017-2020 Norbert Schlia.
Based on work copyright (C) 2006-2008 David Collett, 2008-2013 K. Henriksson.
Much thanks to them for the original work!
This is free software: you are free to change and redistribute it under the terms of the GNU General
Public License (GPL) version 3 or later.
This manual is copyright (C) 2010-2011 K. Henriksson and (C) 2017-2020 by N. Schlia and may be
distributed under the GNU Free Documentation License (GFDL) 1.3 or later with no invariant sections, or
alternatively under the GNU General Public License (GPL) version 3 or later.