Provided by: ffmpegfs_2.11-1build1_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 and video files upon access.
Encoding options
--desttype=TYPE, -odesttype=TYPE
Select the 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 not 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.
Defaults to: mp4
--audiocodec=TYPE, -oaudiocodec=TYPE
Select an 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.
Defaults to: The destination format’s default setting, as indicated by the first codec
name in the list.
--videocodec=TYPE, -ovideocodec=TYPE
Select a 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.
Defaults to: The destination format’s default setting, as indicated by the first codec
name in the list.
--autocopy=OPTION, -oautocopy=OPTION
Select the 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 would be a regular MOV or 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.
Defaults to: OFF
--recodesame=OPTION, -orecodesame=OPTION
Select recode to the same format option, OPTION can be:
┌────┬──────────────────────────────────┐
│NO │ Never recode to the same format. │
├────┼──────────────────────────────────┤
│YES │ Always recode to the same │
│ │ format. │
└────┴──────────────────────────────────┘
Defaults to: 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 the MP4 output format only, and is ignored for all other formats.
Defaults to: 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 the MP4 output format only, and is ignored for all other formats.
Defaults to: HQ
--extensions=LIST, -oextensions=LIST
Set a list of extra file extensions recognised as input files. LIST can contain one
or more entries, separated by commas.
Example: --extensions=xxx,abc,yxz,aaa
Set a list of extra file extensions recognised as input files. LIST can contain one
or more entries, separated by commas.Take care to select extensions that can actually
be converted. Specifying something like --extensions=txt would make FFmpegfs attempt
to transcode text files, resulting in error messages, making these files inaccessible.
Defaults to: Use the default set as defined by FFmpeg.
--hide_extensions=LIST, -ohide_extensions=LIST
Set a list of file extensions that should be hidden from the output. LIST can contain
one or more entries, separated by commas.
Example: --hide_extensions=jpg,png,cue to stop covers and cue sheets from showing up.
Defaults to: Show all files.
Audio Options
--audiobitrate=BITRATE, -o audiobitrate=BITRATE
Select the audio encoding bitrate.
Defaults to: 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
This limits the output sample rate to SAMPLERATE. If the source file sample rate is
higher, it will be downsampled automatically.
Typical values are 8000, 11025, 22050, 44100, 48000, 96000, and 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 but only 22.05 or 44.1 KHz is supported, 22.05
KHz will be set).
Set to 0 to keep the source rate.
Defaults to: 44.1 kHz
SAMPLERATE
can be defined as...
• In Hz: # or #Hz
• In kHz: #K or #KHz
--audiochannels=CHANNELS, -o audiochannels=CHANNELS
This 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.
Defaults to: 2 channels (stereo)
--audiosamplefmt=SAMPLEFMT, -o audiosamplefmt=SAMPLEFMT
This sets a 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
will be reported as a 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 │
└─────────────────┴─────────────────────────────────┘
Defaults to: 0 (Use the same as the source or the predefined format of the destination
if the source format is not possible.)
Video Options
--videobitrate=BITRATE, -o videobitrate=BITRATE
This sets the video encoding bit rate. Setting this too high or too low may cause
transcoding to fail.
Defaults to: 2 Mbit
mp4: May be specified as 500 to 25,000 kbps.
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
This 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.
Defaults to: keep source video height
--videowidth=WIDTH, -o videowidth=WIDTH
This 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.
Defaults to: keep source video width
--deinterlace, -o deinterlace
Deinterlace video if necessary while transcoding.
This may need a higher bit rate, but this will increase picture quality when streaming
via HTML5.
Defaults to: "no deinterlace"
HLS Options
--segment_duration, -o segment_duration
Set the duration of one video segment of the HLS stream. This argument is a floating
point value, e.g., it can be set to 2.5 for 2500 milliseconds.
Should normally be left as the default.
Note: This applies to the HLS output format only, and is ignored for all other
formats.
Defaults to: 10 seconds
--min_seek_time_diff, -o min_seek_time_diff
If the requested HLS segment is less than min_seek_time seconds away, discard the seek
request. The segment will be available very soon anyway, and that makes a re-transcode
necessary. Set to 0 to disable.
Should normally be left as the default.
Note: This applies to the HLS output format only, and is ignored for all other
formats.
Defaults to: 30 seconds
Hardware Acceleration Options
--hwaccel_enc=API, -o hwaccel_enc=API
Select the hardware acceleration API for encoding.
Defaults to: 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 a codec and, optionally, a profile for hardware decoding. The option can be
repeated to block several codecs.
Defaults to: 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 from the codec 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
than one device is available.
Note: This only applies to VAAPI hardware acceleration; all other types are ignored.
Defaults to: empty (use default device).
Example: /dev/dri/renderD128
--hwaccel_dec=API, -o hwaccel_dec=API
Select the hardware acceleration API for decoding.
Defaults to: 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
than one device is available.
Note: This only applies to VAAPI hardware acceleration; all other types are ignored.
Defaults to: empty (use default device)
Example: /dev/dri/renderD128
Album Arts
--noalbumarts, -o noalbumarts
Do not copy album art into the output file.
This will reduce the file size and may be useful when streaming via HTML5 when album
art is not used anyway.
Defaults to: add album arts
Virtual Script
--enablescript, -o enablescript
Add a virtual index.php to every directory. It reads scripts/videotag.php from the
FFmpegfs binary directory.
This can be very handy for testing video playback. Of course, feel free to replace
videotag.php with your own script.
Defaults to: Do not generate script file
--scriptfile, -o scriptfile
Set the name of the virtual script created in each directory.
Defaults to: index.php
--scriptsource, -o scriptsource
Use a different source file.
Defaults to: scripts/videotag.php
Cache Options
--expiry_time=TIME, -o expiry_time=TIME
Cache entries expire after TIME and will be deleted to save disc space.
Defaults to: 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.
Defaults to: 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.
Defaults to: 30 seconds
--prebuffer_size=SIZE, -o prebuffer_size=SIZE
Files will be decoded until the buffer contains this many bytes, allowing playback to
start smoothly without lags.
Set to 0 to disable pre-buffering.
Defaults to: 100 KB
--max_cache_size=SIZE, -o max_cache_size=SIZE
Set the maximum diskspace used by the cache. If the cache grows beyond this limit when
a file is transcoded, old entries will be deleted to keep the cache within the size
limit.
Defaults to: unlimited
--min_diskspace=SIZE, -o min_diskspace=SIZE
Set the required diskspace on the cachepath mount. If the remaining space falls below
SIZE when a file is transcoded, old entries will be deleted to keep the diskspace
within the limit.
Defaults to: 0 (no minimum space)
--cachepath=DIR, -o cachepath=DIR
Sets the disc cache directory to DIR. If it does not already exist, it will be
created. The user running FFmpegfs must have write access to the location.
--disable_cache, -o disable_cache
Disable the cache functionality completely.
Defaults to: 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 it 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 one will always do the maintenance.
Defaults to: 1 hour
--prune_cache
Prune the cache immediately according to the above settings at application start up.
--clear_cache, -o clear_cache
On startup, clear the cache. 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 the number of CPU cores. Should be left as the default.
Defaults to: 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 still creating a valid file. When this option is set, transcoding will stop
with an error.
Defaults to: Ignore errors
--min_dvd_chapter_duration=SECONDS, -o min_dvd_chapter_duration=SECONDS
This ignores DVD chapters shorter than SECONDS. To disable, set to 0. This avoids
transcoding errors for DVD chapters too short to detect its streams.
Defaults to: 1 second
--win_smb_fix, -o win_smb_fix
Windows seems to access the files on Samba drives starting at the last 64K segment
when the file is opened. Setting --win_smb_fix=1 will ignore these attempts (not
decode the file up to this point).
Defaults to: 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 is 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 programme runs. It implies -f.
-f
Run in the foreground instead of detaching from the terminal.
-h, --help
Print usage information.
-V, --version
Output version information.
-c, --capabilities
Output FFmpeg capabilities: a list of the system’s 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 yet, 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 file size 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 disc buffer and can be accessed very quickly.
Transcoding is done in an extra thread, so if other processes 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 the timeout, transcoding will continue. If not, it will stop, and the chunk created
so far will be discarded to save disc 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 programmes will read a file from
start to finish. Future enhancements may provide true random seeking (but if this is
feasible, it is not yet clear 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: The 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 about 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.
The reason is that the start block of an MP4 contains a field with the size of the
compressed data section. Suffice it 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 seeking 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 starting 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", which 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 a few) 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 could
start.
The data part is divided into chunks of about 1 second each, each with its own header, so
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 here:
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 OBJECTIVES
• Create a Windows version
FILES
/usr/local/bin/ffmpegfs, /etc/fstab
AUTHORS
This fork with FFmpeg support has been 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.