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.