Provided by: ffmpegfs_2.9-1_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:
┌───────┬─────────────────────┬────────┬───────────────┐
│ │ │ │ │
│Format │ Description │ Audio │ Video │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│AIFF │ Audio Interchange │ │ PCM 16 bit BE │
│ │ File Format │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│ALAC │ Apple Lossless │ │ ALAC │
│ │ Audio Codec │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│FLAC │ Free Lossless Audio │ │ FLAC │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│HLS │ HTTP Live Streaming │ H264 │ AAC │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│MOV │ QuickTime File │ H264 │ AAC │
│ │ Format │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│MP3 │ MPEG-2 Audio Layer │ │ MP3 │
│ │ III │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│MP4 │ MPEG-4 │ H264 │ AAC │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│OGG │ │ Theora │ Vorbis │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│MKV │ Matroska │ H264 │ AAC │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│Opus │ │ Opus │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│ProRes │ Apple ProRes │ ProRes │ PCM 16 bit LE │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│TS │ MPEG Transport │ H264 │ AAC │
│ │ Stream │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│WAV │ Waveform Audio File │ │ PCM 16 bit LE │
│ │ Format │ │ │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│WebM │ │ VP9 │ Opus │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│BMP │ Video to frameset │ │ BMP │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│JPG │ Video to frameset │ │ JPEG │
├───────┼─────────────────────┼────────┼───────────────┤
│ │ │ │ │
│PNG │ Video to frameset │ │ PNG │
└───────┴─────────────────────┴────────┴───────────────┘
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:
AIFF, ALAC, BMP, FLAC, HLS, JPG, MOV, MP3, MP4, MKV, OGG, Opus, PNG, ProRes, TS, WAV,
WebM.
To stream videos, MP4, TS, HLS, OGG, WEBM, MKV or MOV/PRORES must be selected.
To use HTTP Live Streaming, set HLS.
When a destination JPG, PNG or BMP is chosen, all frames of a video source file will
be presented in a virtual directory named after the source file. Audio will no be
available.
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
--extensions=LIST, -oextensions=LIST
Set list of extra file extensions recognised as input files. LIST can contain one or
more entries, separated by kommas.
Example: --extensions=xxx,abc,yxz,aaa
Take care to select extensions that actually can be converted. Specifying something
like --extensions=txt would make FFmpegfs attempt to transcode text files, resulting
in error messages, making these files inaccessible.
Default: Use default set as defined by FFmpeg.
--audiocodec=TYPE, -oaudiocodec=TYPE
Select audio codec. TYPE depends on the destination format and can currently be:
┌────────┬───────────────┐
│Formats │ Audio Codecs │
├────────┼───────────────┤
│MP4 │ AAC, MP3 │
├────────┼───────────────┤
│WebM │ OPUS, VORBIS │
├────────┼───────────────┤
│MOV │ AAC, AC3, MP3 │
├────────┼───────────────┤
│MKV │ AAC, AC3, MP3 │
├────────┼───────────────┤
│TS, HLS │ AAC, AC3, MP3 │
└────────┴───────────────┘
Other destination formats do not support other codecs than the default.
Default: The default setting for the destination format, see bold codec name in table
--videocodec=TYPE, -ovideocodec=TYPE
Select video codec. TYPE depends on the destination format and can currently be:
┌────────┬──────────────────────────┐
│Formats │ Video Codecs │
├────────┼──────────────────────────┤
│MP4 │ H264, H265, MPEG1, MPEG2 │
├────────┼──────────────────────────┤
│WebM │ VP9, VP8, AV1 │
├────────┼──────────────────────────┤
│MOV │ H264, H265, MPEG1, MPEG2 │
├────────┼──────────────────────────┤
│MKV │ H264, H265, MPEG1, MPEG2 │
├────────┼──────────────────────────┤
│TS, HLS │ H264, H265, MPEG1, MPEG2 │
└────────┴──────────────────────────┘
Other destination formats do not support other codecs than the default.
Default: The default setting for the destination format, see bold codec name in table
--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 FFmpeg’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
--recodesame=OPTION, -orecodesame=OPTION
Select recode to same format option, OPTION can be:
┌────┬───────────────────────────────┐
│NO │ Never recode to same format. │
├────┼───────────────────────────────┤
│YES │ Always recode to same format. │
└────┴───────────────────────────────┘
Default: NO
--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 │
└────────┴──────────────────────────┘
Note: Applies to MP4 output format only, ignored for all other formats.
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 │
└─────────┴─────────────────┘
Note: Applies to MP4 output format only, ignored for all other formats.
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
--audiochannels=CHANNELS, -o audiochannels=CHANNELS
Limits the number of output channels to CHANNELS. If the source has more channels, the
number will be reduced to this limit.
Typical values are 1, 2 or 6 (e.g. 5.1) channels.
If the target codec does not support the selected number of channels, transcoding may
fail.
Set to 0 to keep the number of channels.
Default: 2 channels (stereo)
--audiosamplefmt=SAMPLEFMT, -o audiosamplefmt=SAMPLEFMT
Set sample format. SAMPLEFMT can be:
0 to use the predefined setting, 8, 16, 32, 64 for integer format, F16, F32, F64 for
floating point
Not all formats are supported by all destination types, selecting an invalid format
for will be reported as command line error and a list of values printed.
┌─────────────────┬─────────────────────────────────┐
│Container Format │ Sample Format │
├─────────────────┼─────────────────────────────────┤
│AIFF │ 0, 16, 32 │
├─────────────────┼─────────────────────────────────┤
│ALAC │ 0, 16, 24 │
├─────────────────┼─────────────────────────────────┤
│WAV │ 0, 8, 16, 32, 64, F16, F32, F64 │
├─────────────────┼─────────────────────────────────┤
│FLAC │ 0, 16, 24 │
└─────────────────┴─────────────────────────────────┘
Default: 0 (Use same as source or the predefined format of the destination if source
format is not possible).
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
HLS Options
--segment_duration, -o segment_duration
Set duration of one video segment of HLS stream. This argument is a floating point
value, e.g can be set to 2.5 for 2500 milliseconds.
Should normally be left at default.
Note: Applies to HLS output format only, ignored for all other formats.
Default: 10 seconds
--min_seek_time_diff, -o min_seek_time_diff
If requested HLS segment is less than min_seek_time seconds away, discard seek
request. The segment will be available very soon anyway, and that seek makes a
re-transcode necessary. Set to 0 to disable.
Should normally be left at default.
Note: Applies to HLS output format only, ignored for all other formats.
Default: 30 seconds
Hardware Acceleration Options
--hwaccel_enc=API, -o hwaccel_enc=API
Select the hardware acceleration API for encoding.
Default: NONE (no acceleration)
API
can be defined as...
· NONE: use software encoder
· VAAPI: Video Acceleration API (VA-API)
· OMX: OpenMAX (Open Media Acceleration)
--hwaccel_dec_blocked=CODEC[:PROFILE[:PROFILE]], -o
hwaccel_dec_blocked=CODEC:[:PROFILE[:PROFILE]]
Block codec and optionally profile from hardware decoding. Option can be repeated to
block several codecs
Default: no codecs blocked
CODEC
can be defined as...
· H263: H.263
· H264: H.264
· HEVC: H.265 / HEVC
· MPEG2: MPEG-2 video
· MPEG4: MPEG-4 video
· VC1: SMPTE VC-1
· VP8: Google VP9
· VP9: Google VP9
· WMV3: Windows Media Video 9
PROFILE
can optionally be added to block a certain profile only.
Example: VP9:0 blocks Google VP profile 0
Example: H264:1:33 blocks H.264 profile 1 and 33
--hwaccel_enc_device=DEVICE, -o hwaccel_enc_device=DEVICE
Select the hardware acceleration device. May be required for VAAPI, especially if more
the one device is available.
Note: Applies to VAAPI hardware acceleration only, ignored for all other types.
Default: empty (use default device)
Example: /dev/dri/renderD128
--hwaccel_dec=API, -o hwaccel_dec=API
Select the hardware acceleration API for decoding.
Default: NONE (no acceleration)
API
can be defined as...
· NONE: use software decoder
· VAAPI: Video Acceleration API (VA-API)
· MMAL: Multimedia Abstraction Layer by Broadcom
--hwaccel_dec_device=DEVICE, -o hwaccel_dec_device=DEVICE
Select the hardware acceleration device. May be required for VAAPI, especially if more
the one device is available.
Note: Applies to VAAPI hardware acceleration only, ignored for all other types.
Default: empty (use default device)
Example: /dev/dri/renderD128
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
FFmpegfs 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 completely.
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 at application start up.
--clear_cache, -o clear_cache
Clear cache on start up. All previously transcoded 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. Should be left at default.
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: on
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.
-c, --capabilities
Output FFmpeg capabilities: list on the system available codecs.
-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/{empty}*.flac and /mnt/music/{empty}*.ogg will show up
as /mnt/ffmpegfs/{empty}*.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 permit 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, AIFF or WAV output files, but only fair to good for MP4 or WebM because the actual
size heavily depends on the content encoded.
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 all processes close the file
before its end, transcoding will continue for some time. If the file is accessed again
before timeout, transcoding will continue, 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.
WAV: A pro format WAV header will be created with estimates of the WAV file size. This
header will be replaced when the file is finished. It does not seem necessary, though, as
most modern players obviously ignore this information and play the file anyway.
ABOUT OUTPUT FORMATS
A few words to the supported output formats. There is not much to say about the MP3 output
as these are regular constant bitrate (CBR) MP3 files with no strings attached. They
should play well in any modern player.
MP4 files are special, though, as regular MP4s are 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 file seek done to the
beginning, and the size atom updated.
For a continuous live stream, that size will never be known. For our transcoded files one
would have to wait for the whole file to be recoded to get that value. 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. Also, there is only one big data block, a fact that
hampers random seek inside the contents without having the complete data section.
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 aforementioned meta 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 active (ISMV,
faststart, separate_moof/empty_moov to name them) which are not implemented in older
versions of FFmpeg (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 encoding it as a whole before this is
possible. Encoding it completely would mean it would take some time before playback can
start.
The data part is divided into chunks of about 1 second length, each with its own header,
thus it is possible to fill in the size fields early enough.
As a draw back not all players support the format, or play it with strange side effects.
VLC plays the file, but updates the time display every few seconds only. When streamed
over HTML5 video tags, sometimes there will be no total time shown, but that is OK, as
long as the file plays. Playback cannot be positioned past the current playback position,
only backwards.
But that’s the price of starting playback fast.
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 little bit of C and mostly C++11. It uses the following
libraries:
· FUSE
FFmpeg home pages:
· FFmpeg
FUTURE PLANS
· Create a windows version
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).
Many 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) 2017-2022 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) 2017-2022 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.