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The generic syntax is:
ffmpeg [global options] [[infile options][`-i' infile]]... {[outfile options] outfile}...
|
ffmpeg is a very fast video and audio converter that can also grab from a live audio/video source. It can also convert between arbitrary sample rates and resize video on the fly with a high quality polyphase filter.
ffmpeg reads from an arbitrary number of input "files" (which can be regular
files, pipes, network streams, grabbing devices, etc.), specified by the
-i option, and writes to an arbitrary number of output "files", which are
specified by a plain output filename. Anything found on the command line which
cannot be interpreted as an option is considered to be an output filename.
Each input or output file can in principle contain any number of streams of
different types (video/audio/subtitle/attachment/data). Allowed number and/or
types of streams can be limited by the container format. Selecting, which
streams from which inputs go into output, is done either automatically or with
the -map option (see the Stream selection chapter).
To refer to input files in options, you must use their indices (0-based). E.g.
the first input file is 0, the second is 1 etc. Similarly, streams
within a file are referred to by their indices. E.g. 2:3 refers to the
fourth stream in the third input file. See also the Stream specifiers chapter.
As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file. Exceptions from this rule are the global options (e.g. verbosity level), which should be specified first.
Do not mix input and output files - first specify all input files, then all output files. Also do not mix options which belong to different files. All options apply ONLY to the next input or output file and are reset between files.
ffmpeg -i input.avi -b:v 64k output.avi |
ffmpeg -i input.avi -r 24 output.avi |
ffmpeg -r 1 -i input.m2v -r 24 output.avi |
The format option may be needed for raw input files.
By default ffmpeg includes only one stream of each type (video, audio, subtitle) present in the input files and adds them to each output file. It picks the "best" of each based upon the following criteria; for video it is the stream with the highest resolution, for audio the stream with the most channels, for subtitle it's the first subtitle stream. In the case where several streams of the same type rate equally, the lowest numbered stream is chosen.
You can disable some of those defaults by using -vn/-an/-sn options. For
full manual control, use the -map option, which disables the defaults just
described.
All the numerical options, if not specified otherwise, accept in input a string representing a number, which may contain one of the International System number postfixes, for example 'K', 'M', 'G'. If 'i' is appended after the postfix, powers of 2 are used instead of powers of 10. The 'B' postfix multiplies the value for 8, and can be appended after another postfix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Options which do not take arguments are boolean options, and set the corresponding value to true. They can be set to false by prefixing with "no" the option name, for example using "-nofoo" in the command line will set to false the boolean option with name "foo".
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. -codec:a:1 ac3 option contains
a:1 stream specifer, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
of them. E.g. the stream specifier in -b:a 128k matches all audio
streams.
An empty stream specifier matches all streams, for example -codec copy
or -codec: copy would copy all the streams without reencoding.
Possible forms of stream specifiers are:
Matches the stream with this index. E.g. -threads:1 4 would set the
thread count for the second stream to 4.
stream_type is one of: 'v' for video, 'a' for audio, 's' for subtitle, 'd' for data and 't' for attachments. If stream_index is given, then matches stream number stream_index of this type. Otherwise matches all streams of this type.
If stream_index is given, then matches stream number stream_index in program with id program_id. Otherwise matches all streams in this program.
Matches the stream by format-specific ID.
These options are shared amongst the av* tools.
Show license.
Show help.
Show version.
Show available formats.
The fields preceding the format names have the following meanings:
Decoding available
Encoding available
Show available codecs.
The fields preceding the codec names have the following meanings:
Decoding available
Encoding available
Video/audio/subtitle codec
Codec supports slices
Codec supports direct rendering
Codec can handle input truncated at random locations instead of only at frame boundaries
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Set the logging level used by the library. loglevel is a number or a string containing one of the following values:
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting
the environment variable AV_LOG_FORCE_COLOR.
The use of the environment variable NO_COLOR is deprecated and
will be dropped in a following FFmpeg version.
Dump full command line and console output to a file named
program-YYYYMMDD-HHMMSS.log in the current
directory.
This file can be useful for bug reports.
It also implies -loglevel verbose.
Note: setting the environment variable FFREPORT to any value has the
same effect.
Allows setting and clearing cpu flags. This option is intended for testing. Do not use it unless you know what you're doing.
ffmpeg -cpuflags -sse+mmx ... ffmpeg -cpuflags mmx ... ffmpeg -cpuflags 0 ... |
These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the `-help' option. They are separated into two categories:
These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.
These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.
For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the `id3v2_version' private option of the MP3 muxer:
ffmpeg -i input.flac -id3v2_version 3 out.mp3 |
All codec AVOptions are obviously per-stream, so the chapter on stream specifiers applies to them
Note `-nooption' syntax cannot be used for boolean AVOptions, use `-option 0'/`-option 1'.
Note2 old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.
set bitrate (in bits/s)
set bitrate (in bits/s)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to min/max bitrate. Lowering tolerance too much has an adverse effect on quality.
Possible values:
use four motion vector by macroblock (mpeg4)
use 1/4 pel motion compensation
use loop filter
use fixed qscale
use gmc
always try a mb with mv=<0,0>
use internal 2pass ratecontrol in first pass mode
use internal 2pass ratecontrol in second pass mode
only decode/encode grayscale
don't draw edges
error[?] variables will be set during encoding
normalize adaptive quantization
use interlaced dct
force low delay
place global headers in extradata instead of every keyframe
use only bitexact stuff (except (i)dct)
h263 advanced intra coding / mpeg4 ac prediction
Deprecated, use mpegvideo private options instead
Deprecated, use mpegvideo private options instead
interlaced motion estimation
closed gop
set motion estimation method
Possible values:
zero motion estimation (fastest)
full motion estimation (slowest)
EPZS motion estimation (default)
esa motion estimation (alias for full)
tesa motion estimation
dia motion estimation (alias for epzs)
log motion estimation
phods motion estimation
X1 motion estimation
hex motion estimation
umh motion estimation
iter motion estimation
set the group of picture size
set audio sampling rate (in Hz)
set number of audio channels
set cutoff bandwidth
video quantizer scale compression (VBR). Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
video quantizer scale blur (VBR)
min video quantizer scale (VBR)
max video quantizer scale (VBR)
max difference between the quantizer scale (VBR)
use 'frames' B frames
qp factor between p and b frames
ratecontrol method
strategy to choose between I/P/B-frames
rtp payload size in bytes
workaround not auto detected encoder bugs
Possible values:
some old lavc generated msmpeg4v3 files (no autodetection)
Xvid interlacing bug (autodetected if fourcc==XVIX)
(autodetected if fourcc==UMP4)
padding bug (autodetected)
illegal vlc bug (autodetected per fourcc)
old standard qpel (autodetected per fourcc/version)
direct-qpel-blocksize bug (autodetected per fourcc/version)
edge padding bug (autodetected per fourcc/version)
workaround various bugs in microsofts broken decoders
trancated frames
single coefficient elimination threshold for luminance (negative values also consider dc coefficient)
single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)
how strictly to follow the standards
Possible values:
strictly conform to a older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what consequences
allow unofficial extensions
allow non standardized experimental things
qp offset between P and B frames
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder shouldnt do as an error
use MPEG quantizers instead of H.263
how to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function)
experimental quantizer modulation
experimental quantizer modulation
Set rate control equation. When computing the expression, besides the standard functions defined in the section 'Expression Evaluation', the following functions are available: bits2qp(bits), qp2bits(qp). Also the following constants are available: iTex pTex tex mv fCode iCount mcVar var isI isP isB avgQP qComp avgIITex avgPITex avgPPTex avgBPTex avgTex.
Set max bitrate tolerance (in bits/s). Requires bufsize to be set.
Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.
set ratecontrol buffer size (in bits)
currently useless
qp factor between P and I frames
qp offset between P and I frames
initial complexity for 1-pass encoding
DCT algorithm
Possible values:
autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
compresses bright areas stronger than medium ones
temporal complexity masking
spatial complexity masking
inter masking
compresses dark areas stronger than medium ones
select IDCT implementation
Possible values:
floating point AAN IDCT
set error concealment strategy
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
prediction method
Possible values:
sample aspect ratio
print specific debug info
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
motion vector
error recognition
memory management control operations (H.264)
visualize quantization parameter (QP), lower QP are tinted greener
visualize block types
picture buffer allocations
threading operations
visualize motion vectors (MVs)
Possible values:
forward predicted MVs of P-frames
forward predicted MVs of B-frames
backward predicted MVs of B-frames
full pel me compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
sub pel me compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
macroblock compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
interlaced dct compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation
amount of motion predictors from the previous frame
pre motion estimation
pre motion estimation compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation pre-pass
sub pel motion estimation quality
limit motion vectors range (1023 for DivX player)
intra quant bias
inter quant bias
Possible values:
variable length coder / huffman coder
arithmetic coder
raw (no encoding)
run-length coder
deflate-based coder
context model
macroblock decision algorithm (high quality mode)
Possible values:
use mbcmp (default)
use fewest bits
use best rate distortion
scene change threshold
min lagrange factor (VBR)
max lagrange factor (VBR)
noise reduction
number of bits which should be loaded into the rc buffer before decoding starts
Possible values:
allow non spec compliant speedup tricks
Deprecated, use mpegvideo private options instead
skip bitstream encoding
place global headers at every keyframe instead of in extradata
Frame data might be split into multiple chunks
Show all frames before the first keyframe
Deprecated, use mpegvideo private options instead
deprecated, use mpegvideo private options instead
Possible values:
detect a good number of threads
motion estimaton threshold
macroblock threshold
intra_dc_precision
nsse weight
number of macroblock rows at the top which are skipped
number of macroblock rows at the bottom which are skipped
Possible values:
Possible values:
decode at 1= 1/2, 2=1/4, 3=1/8 resolutions
frame skip threshold
frame skip factor
frame skip exponent
frame skip compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
increases the quantizer for macroblocks close to borders
min macroblock lagrange factor (VBR)
max macroblock lagrange factor (VBR)
motion estimation bitrate penalty compensation (1.0 = 256)
Possible values:
Possible values:
Possible values:
refine the two motion vectors used in bidirectional macroblocks
downscales frames for dynamic B-frame decision
minimum interval between IDR-frames
reference frames to consider for motion compensation
chroma qp offset from luma
rate-distortion optimal quantization
multiplied by qscale for each frame and added to scene_change_score
adjusts sensitivity of b_frame_strategy 1
GOP timecode frame start number, in non drop frame format
set desired number of audio channels
Possible values:
Possible values:
set the log level offset
number of slices, used in parallelized encoding
select multithreading type
Possible values:
audio service type
Possible values:
Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
sample format audio decoders should prefer
Possible values:
8-bit unsigned integer
16-bit signed integer
32-bit signed integer
32-bit float
64-bit double
Possible values:
reduce buffering
set probing size
set packet size
Possible values:
ignore index
generate pts
do not fill in missing values that can be exactly calculated
disable AVParsers, this needs nofillin too
ignore dts
discard corrupted frames
try to interleave outputted packets by dts
dont merge side data
enable RTP MP4A-LATM payload
how many microseconds are analyzed to estimate duration
decryption key
max memory used for timestamp index (per stream)
max memory used for buffering real-time frames
print specific debug info
Possible values:
maximum muxing or demuxing delay in microseconds
number of frames used to probe fps
microseconds by which audio packets should be interleaved earlier
microseconds for each chunk
size in bytes for each chunk
set error detection flags (deprecated; use err_detect, save via avconv)
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder shouldnt do as an error
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
consider things that violate the spec and have not been seen in the wild as errors
consider all spec non compliancies as errors
consider things that a sane encoder shouldnt do as an error
Force input or output file format. The format is normally auto detected for input files and guessed from file extension for output files, so this option is not needed in most cases.
input file name
Overwrite output files without asking.
Do not overwrite output files but exit if file exists.
Select an encoder (when used before an output file) or a decoder (when used
before an input file) for one or more streams. codec is the name of a
decoder/encoder or a special value copy (output only) to indicate that
the stream is not to be re-encoded.
For example
ffmpeg -i INPUT -map 0 -c:v libx264 -c:a copy OUTPUT |
encodes all video streams with libx264 and copies all audio streams.
For each stream, the last matching c option is applied, so
ffmpeg -i INPUT -map 0 -c copy -c:v:1 libx264 -c:a:137 libvorbis OUTPUT |
will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis.
Stop writing the output after its duration reaches duration.
duration may be a number in seconds, or in hh:mm:ss[.xxx] form.
Set the file size limit, expressed in bytes.
When used as an input option (before -i), seeks in this input file to
position. When used as an output option (before an output filename),
decodes but discards input until the timestamps reach position. This is
slower, but more accurate.
position may be either in seconds or in hh:mm:ss[.xxx] form.
Set the input time offset in seconds.
[-]hh:mm:ss[.xxx] syntax is also supported.
The offset is added to the timestamps of the input files.
Specifying a positive offset means that the corresponding
streams are delayed by offset seconds.
Set the recording timestamp in the container. The syntax for time is:
now|([(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z]) |
If the value is "now" it takes the current time. Time is local time unless 'Z' or 'z' is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.
Set a metadata key/value pair.
An optional metadata_specifier may be given to set metadata
on streams or chapters. See -map_metadata documentation for
details.
This option overrides metadata set with -map_metadata. It is
also possible to delete metadata by using an empty value.
For example, for setting the title in the output file:
ffmpeg -i in.avi -metadata title="my title" out.flv |
To set the language of the first audio stream:
ffmpeg -i INPUT -metadata:s:a:1 language=eng OUTPUT |
Specify target file type (vcd, svcd, dvd, dv,
dv50). type may be prefixed with pal-, ntsc- or
film- to use the corresponding standard. All the format options
(bitrate, codecs, buffer sizes) are then set automatically. You can just type:
ffmpeg -i myfile.avi -target vcd /tmp/vcd.mpg |
Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:
ffmpeg -i myfile.avi -target vcd -bf 2 /tmp/vcd.mpg |
Set the number of data frames to record. This is an alias for -frames:d.
Stop writing to the stream after framecount frames.
Use fixed quality scale (VBR). The meaning of q is codec-dependent.
filter_graph is a description of the filter graph to apply to
the stream. Use -filters to show all the available filters
(including also sources and sinks).
See also the `-filter_complex' option if you want to create filter graphs with multiple inputs and/or outputs.
Specify the preset for matching stream(s).
Print encoding progress/statistics. On by default.
Print timestamp information. It is off by default. This option is mostly useful for testing and debugging purposes, and the output format may change from one version to another, so it should not be employed by portable scripts.
See also the option -fdebug ts.
Add an attachment to the output file. This is supported by a few formats
like Matroska for e.g. fonts used in rendering subtitles. Attachments
are implemented as a specific type of stream, so this option will add
a new stream to the file. It is then possible to use per-stream options
on this stream in the usual way. Attachment streams created with this
option will be created after all the other streams (i.e. those created
with -map or automatic mappings).
Note that for Matroska you also have to set the mimetype metadata tag:
ffmpeg -i INPUT -attach DejaVuSans.ttf -metadata:s:2 mimetype=application/x-truetype-font out.mkv |
(assuming that the attachment stream will be third in the output file).
Extract the matching attachment stream into a file named filename. If
filename is empty, then the value of the filename metadata tag
will be used.
E.g. to extract the first attachment to a file named 'out.ttf':
ffmpeg -dump_attachment:t:0 out.ttf INPUT |
To extract all attachments to files determined by the filename tag:
ffmpeg -dump_attachment:t "" INPUT |
Technical note - attachments are implemented as codec extradata, so this option can actually be used to extract extradata from any stream, not just attachments.
Set the number of video frames to record. This is an alias for -frames:v.
Set frame rate (Hz value, fraction or abbreviation), (default = 25). For output
streams implies -vsync cfr.
Set frame size. The format is `wxh' (default - same as source).
Set the video display aspect ratio specified by aspect.
aspect can be a floating point number string, or a string of the form num:den, where num and den are the numerator and denominator of the aspect ratio. For example "4:3", "16:9", "1.3333", and "1.7777" are valid argument values.
All the crop options have been removed. Use -vf crop=width:height:x:y instead.
All the pad options have been removed. Use -vf pad=width:height:x:y:color instead.
Disable video recording.
Set the video codec. This is an alias for -codec:v.
Use same quantizer as source (implies VBR).
Note that this is NOT SAME QUALITY. Do not use this option unless you know you need it.
Select the pass number (1 or 2). It is used to do two-pass video encoding. The statistics of the video are recorded in the first pass into a log file (see also the option -passlogfile), and in the second pass that log file is used to generate the video at the exact requested bitrate. On pass 1, you may just deactivate audio and set output to null, examples for Windows and Unix:
ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y NUL ffmpeg -i foo.mov -c:v libxvid -pass 1 -an -f rawvideo -y /dev/null |
Set two-pass log file name prefix to prefix, the default file name prefix is "ffmpeg2pass". The complete file name will be `PREFIX-N.log', where N is a number specific to the output stream
Note that this option is overwritten by a local option of the same name
when using -vcodec libx264. That option maps to the x264 option stats
which has a different syntax.
Set the ISO 639 language code (3 letters) of the current video stream.
filter_graph is a description of the filter graph to apply to
the input video.
Use the option "-filters" to show all the available filters (including
also sources and sinks). This is an alias for -filter:v.
Set pixel format. Use -pix_fmts to show all the supported
pixel formats.
If the selected pixel format can not be selected, ffmpeg will print a
warning and select the best pixel format supported by the encoder.
If pix_fmt is prefixed by a +, ffmpeg will exit with an error
if the requested pixel format can not be selected, and automatic conversions
inside filter graphs are disabled.
If pix_fmt is a single +, ffmpeg selects the same pixel format
as the input (or graph output) and automatic conversions are disabled.
Set SwScaler flags.
Discard threshold.
Rate control override for specific intervals, formatted as "int,int,int" list separated with slashes. Two first values are the beginning and end frame numbers, last one is quantizer to use if positive, or quality factor if negative.
Deinterlace pictures.
This option is deprecated since the deinterlacing is very low quality.
Use the yadif filter with -filter:v yadif.
Force interlacing support in encoder (MPEG-2 and MPEG-4 only). Use this option if your input file is interlaced and you want to keep the interlaced format for minimum losses. The alternative is to deinterlace the input stream with `-deinterlace', but deinterlacing introduces losses.
Calculate PSNR of compressed frames.
Dump video coding statistics to `vstats_HHMMSS.log'.
Dump video coding statistics to file.
top=1/bottom=0/auto=-1 field first
Intra_dc_precision.
Force video tag/fourcc. This is an alias for -tag:v.
Show QP histogram
Deprecated see -bsf
Force key frames at the specified timestamps, more precisely at the first frames after each specified time. This option can be useful to ensure that a seek point is present at a chapter mark or any other designated place in the output file. The timestamps must be specified in ascending order.
When doing stream copy, copy also non-key frames found at the beginning.
Set the number of audio frames to record. This is an alias for -frames:a.
Set the audio sampling frequency. For output streams it is set by default to the frequency of the corresponding input stream. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
Set the audio quality (codec-specific, VBR). This is an alias for -q:a.
Set the number of audio channels. For output streams it is set by default to the number of input audio channels. For input streams this option only makes sense for audio grabbing devices and raw demuxers and is mapped to the corresponding demuxer options.
Disable audio recording.
Set the audio codec. This is an alias for -codec:a.
Set the audio sample format. Use -sample_fmts to get a list
of supported sample formats.
filter_graph is a description of the filter graph to apply to
the input audio.
Use the option "-filters" to show all the available filters (including
also sources and sinks). This is an alias for -filter:a.
Force audio tag/fourcc. This is an alias for -tag:a.
Deprecated, see -bsf
Set the ISO 639 language code (3 letters) of the current subtitle stream.
Set the subtitle codec. This is an alias for -codec:s.
Disable subtitle recording.
Deprecated, see -bsf
Synchronize read on input.
Designate one or more input streams as a source for the output file. Each input stream is identified by the input file index input_file_id and the input stream index input_stream_id within the input file. Both indices start at 0. If specified, sync_file_id:stream_specifier sets which input stream is used as a presentation sync reference.
The first -map option on the command line specifies the
source for output stream 0, the second -map option specifies
the source for output stream 1, etc.
A - character before the stream identifier creates a "negative" mapping.
It disables matching streams from already created mappings.
An alternative [linklabel] form will map outputs from complex filter graphs (see the `-filter_complex' option) to the output file. linklabel must correspond to a defined output link label in the graph.
For example, to map ALL streams from the first input file to output
ffmpeg -i INPUT -map 0 output |
For example, if you have two audio streams in the first input file,
these streams are identified by "0:0" and "0:1". You can use
-map to select which streams to place in an output file. For
example:
ffmpeg -i INPUT -map 0:1 out.wav |
will map the input stream in `INPUT' identified by "0:1" to the (single) output stream in `out.wav'.
For example, to select the stream with index 2 from input file `a.mov' (specified by the identifier "0:2"), and stream with index 6 from input `b.mov' (specified by the identifier "1:6"), and copy them to the output file `out.mov':
ffmpeg -i a.mov -i b.mov -c copy -map 0:2 -map 1:6 out.mov |
To select all video and the third audio stream from an input file:
ffmpeg -i INPUT -map 0:v -map 0:a:2 OUTPUT |
To map all the streams except the second audio, use negative mappings
ffmpeg -i INPUT -map 0 -map -0:a:1 OUTPUT |
Note that using this option disables the default mappings for this output file.
Map an audio channel from a given input to an output. If output_file_id.stream_specifier is not set, the audio channel will be mapped on all the audio streams.
Using "-1" instead of input_file_id.stream_specifier.channel_id will map a muted channel.
For example, assuming INPUT is a stereo audio file, you can switch the two audio channels with the following command:
ffmpeg -i INPUT -map_channel 0.0.1 -map_channel 0.0.0 OUTPUT |
If you want to mute the first channel and keep the second:
ffmpeg -i INPUT -map_channel -1 -map_channel 0.0.1 OUTPUT |
The order of the "-map_channel" option specifies the order of the channels in the output stream. The output channel layout is guessed from the number of channels mapped (mono if one "-map_channel", stereo if two, etc.). Using "-ac" in combination of "-map_channel" makes the channel gain levels to be updated if input and output channel layouts don't match (for instance two "-map_channel" options and "-ac 6").
You can also extract each channel of an input to specific outputs; the following command extracts two channels of the INPUT audio stream (file 0, stream 0) to the respective OUTPUT_CH0 and OUTPUT_CH1 outputs:
ffmpeg -i INPUT -map_channel 0.0.0 OUTPUT_CH0 -map_channel 0.0.1 OUTPUT_CH1 |
The following example splits the channels of a stereo input into two separate streams, which are put into the same output file:
ffmpeg -i stereo.wav -map 0:0 -map 0:0 -map_channel 0.0.0:0.0 -map_channel 0.0.1:0.1 -y out.ogg |
Note that currently each output stream can only contain channels from a single input stream; you can't for example use "-map_channel" to pick multiple input audio channels contained in different streams (from the same or different files) and merge them into a single output stream. It is therefore not currently possible, for example, to turn two separate mono streams into a single stereo stream. However splitting a stereo stream into two single channel mono streams is possible.
If you need this feature, a possible workaround is to use the amerge filter. For example, if you need to merge a media (here `input.mkv') with 2 mono audio streams into one single stereo channel audio stream (and keep the video stream), you can use the following command:
ffmpeg -i input.mkv -f lavfi -i " amovie=input.mkv:si=1 [a1]; amovie=input.mkv:si=2 [a2]; [a1][a2] amerge" -c:a pcm_s16le -c:v copy output.mkv |
Set metadata information of the next output file from infile. Note that those are file indices (zero-based), not filenames. Optional metadata_spec_in/out parameters specify, which metadata to copy. A metadata specifier can have the following forms:
global metadata, i.e. metadata that applies to the whole file
per-stream metadata. stream_spec is a stream specifier as described in the Stream specifiers chapter. In an input metadata specifier, the first matching stream is copied from. In an output metadata specifier, all matching streams are copied to.
per-chapter metadata. chapter_index is the zero-based chapter index.
per-program metadata. program_index is the zero-based program index.
If metadata specifier is omitted, it defaults to global.
By default, global metadata is copied from the first input file, per-stream and per-chapter metadata is copied along with streams/chapters. These default mappings are disabled by creating any mapping of the relevant type. A negative file index can be used to create a dummy mapping that just disables automatic copying.
For example to copy metadata from the first stream of the input file to global metadata of the output file:
ffmpeg -i in.ogg -map_metadata 0:s:0 out.mp3 |
To do the reverse, i.e. copy global metadata to all audio streams:
ffmpeg -i in.mkv -map_metadata:s:a 0:g out.mkv |
Note that simple 0 would work as well in this example, since global
metadata is assumed by default.
Copy chapters from input file with index input_file_index to the next output file. If no chapter mapping is specified, then chapters are copied from the first input file with at least one chapter. Use a negative file index to disable any chapter copying.
Print specific debug info. category is a number or a string containing one of the following values:
picture buffer allocations
error recognition
macroblock (MB) type
memory management control operations (H.264)
motion vector
picture info
per-block quantization parameter (QP)
rate control
threading operations
visualize block types
visualize quantization parameter (QP), lower QP are tinted greener
Show benchmarking information at the end of an encode. Shows CPU time used and maximum memory consumption. Maximum memory consumption is not supported on all systems, it will usually display as 0 if not supported.
Show benchmarking information during the encode. Shows CPU time used in various steps (audio/video encode/decode).
Exit after ffmpeg has been running for duration seconds.
Dump each input packet to stderr.
When dumping packets, also dump the payload.
Read input at native frame rate. Mainly used to simulate a grab device.
Loop over the input stream. Currently it works only for image streams. This option is used for automatic FFserver testing. This option is deprecated, use -loop 1.
Repeatedly loop output for formats that support looping such as animated GIF (0 will loop the output infinitely). This option is deprecated, use -loop.
Video sync method. For compatibility reasons old values can be specified as numbers. Newly added values will have to be specified as strings always.
Each frame is passed with its timestamp from the demuxer to the muxer.
Frames will be duplicated and dropped to achieve exactly the requested constant framerate.
Frames are passed through with their timestamp or dropped so as to prevent 2 frames from having the same timestamp.
As passthrough but destroys all timestamps, making the muxer generate fresh timestamps based on frame-rate.
Chooses between 1 and 2 depending on muxer capabilities. This is the default method.
With -map you can select from which stream the timestamps should be taken. You can leave either video or audio unchanged and sync the remaining stream(s) to the unchanged one.
Audio sync method. "Stretches/squeezes" the audio stream to match the timestamps,
the parameter is the maximum samples per second by which the audio is changed.
-async 1 is a special case where only the start of the audio stream is corrected
without any later correction.
This option has been deprecated. Use the asyncts audio filter instead.
Copy timestamps from input to output.
Specify how to set the encoder timebase when stream copying. mode is an integer numeric value, and can assume one of the following values:
Use the demuxer timebase.
The time base is copied to the output encoder from the corresponding input demuxer. This is sometimes required to avoid non monotonically increasing timestamps when copying video streams with variable frame rate.
Use the decoder timebase.
The time base is copied to the output encoder from the corresponding input decoder.
Try to make the choice automatically, in order to generate a sane output.
Default value is -1.
Finish encoding when the shortest input stream ends.
Timestamp discontinuity delta threshold.
Set the maximum demux-decode delay.
Set the initial demux-decode delay.
Assign a new stream-id value to an output stream. This option should be specified prior to the output filename to which it applies. For the situation where multiple output files exist, a streamid may be reassigned to a different value.
For example, to set the stream 0 PID to 33 and the stream 1 PID to 36 for an output mpegts file:
ffmpeg -i infile -streamid 0:33 -streamid 1:36 out.ts |
Set bitstream filters for matching streams. bistream_filters is
a comma-separated list of bitstream filters. Use the -bsfs option
to get the list of bitstream filters.
ffmpeg -i h264.mp4 -c:v copy -bsf:v h264_mp4toannexb -an out.h264 |
ffmpeg -i file.mov -an -vn -bsf:s mov2textsub -c:s copy -f rawvideo sub.txt |
Force a tag/fourcc for matching streams.
Specify Timecode for writing. SEP is ':' for non drop timecode and ';' (or '.') for drop.
ffmpeg -i input.mpg -timecode 01:02:03.04 -r 30000/1001 -s ntsc output.mpg |
Define a complex filter graph, i.e. one with arbitrary number of inputs and/or outputs. For simple graphs - those with one input and one output of the same type - see the `-filter' options. filtergraph is a description of the filter graph, as described in Filtergraph syntax.
Input link labels must refer to input streams using the
[file_index:stream_specifier] syntax (i.e. the same as `-map'
uses). If stream_specifier matches multiple streams, the first one will be
used. An unlabeled input will be connected to the first unused input stream of
the matching type.
Output link labels are referred to with `-map'. Unlabeled outputs are added to the first output file.
For example, to overlay an image over video
ffmpeg -i video.mkv -i image.png -filter_complex '[0:v][1:v]overlay[out]' -map '[out]' out.mkv |
Here [0:v] refers to the first video stream in the first input file,
which is linked to the first (main) input of the overlay filter. Similarly the
first video stream in the second input is linked to the second (overlay) input
of overlay.
Assuming there is only one video stream in each input file, we can omit input labels, so the above is equivalent to
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay[out]' -map '[out]' out.mkv |
Furthermore we can omit the output label and the single output from the filter graph will be added to the output file automatically, so we can simply write
ffmpeg -i video.mkv -i image.png -filter_complex 'overlay' out.mkv |
A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which would be awkward to specify on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Check the `presets' directory in the FFmpeg source tree for examples.
Preset files are specified with the vpre, apre,
spre, and fpre options. The fpre option takes the
filename of the preset instead of a preset name as input and can be
used for any kind of codec. For the vpre, apre, and
spre options, the options specified in a preset file are
applied to the currently selected codec of the same type as the preset
option.
The argument passed to the vpre, apre, and spre
preset options identifies the preset file to use according to the
following rules:
First ffmpeg searches for a file named arg.ffpreset in the
directories `$FFMPEG_DATADIR' (if set), and `$HOME/.ffmpeg', and in
the datadir defined at configuration time (usually `PREFIX/share/ffmpeg')
or in a `ffpresets' folder along the executable on win32,
in that order. For example, if the argument is libx264-max, it will
search for the file `libx264-max.ffpreset'.
If no such file is found, then ffmpeg will search for a file named
codec_name-arg.ffpreset in the above-mentioned
directories, where codec_name is the name of the codec to which
the preset file options will be applied. For example, if you select
the video codec with -vcodec libx264 and use -vpre max,
then it will search for the file `libx264-max.ffpreset'.
ffmpeg -g 3 -r 3 -t 10 -b:v 50k -s qcif -f rv10 /tmp/b.rm |
A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which can be specified also on the command line. Lines starting with the hash ('#') character are ignored and are used to provide comments. Empty lines are also ignored. Check the `presets' directory in the FFmpeg source tree for examples.
Preset files are specified with the pre option, this option takes a
preset name as input. FFmpeg searches for a file named preset_name.avpreset in
the directories `$AVCONV_DATADIR' (if set), and `$HOME/.ffmpeg', and in
the data directory defined at configuration time (usually `$PREFIX/share/ffmpeg')
in that order. For example, if the argument is libx264-max, it will
search for the file `libx264-max.avpreset'.
If you specify the input format and device then ffmpeg can grab video and audio directly.
ffmpeg -f oss -i /dev/dsp -f video4linux2 -i /dev/video0 /tmp/out.mpg |
Or with an ALSA audio source (mono input, card id 1) instead of OSS:
ffmpeg -f alsa -ac 1 -i hw:1 -f video4linux2 -i /dev/video0 /tmp/out.mpg |
Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as xawtv by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.
Grab the X11 display with ffmpeg via
ffmpeg -f x11grab -s cif -r 25 -i :0.0 /tmp/out.mpg |
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.
ffmpeg -f x11grab -s cif -r 25 -i :0.0+10,20 /tmp/out.mpg |
0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x-offset and 20 the y-offset for the grabbing.
Any supported file format and protocol can serve as input to ffmpeg:
Examples:
ffmpeg -i /tmp/test%d.Y /tmp/out.mpg |
It will use the files:
/tmp/test0.Y, /tmp/test0.U, /tmp/test0.V, /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc... |
The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the `-s' option if ffmpeg cannot guess it.
ffmpeg -i /tmp/test.yuv /tmp/out.avi |
test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.
ffmpeg -i mydivx.avi hugefile.yuv |
ffmpeg -i /tmp/a.wav -s 640x480 -i /tmp/a.yuv /tmp/a.mpg |
Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.
ffmpeg -i /tmp/a.wav -ar 22050 /tmp/a.mp2 |
Converts a.wav to MPEG audio at 22050 Hz sample rate.
ffmpeg -i /tmp/a.wav -map 0:a -b:a 64k /tmp/a.mp2 -map 0:a -b:a 128k /tmp/b.mp2 |
Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. '-map file:index' specifies which input stream is used for each output stream, in the order of the definition of output streams.
ffmpeg -i snatch_1.vob -f avi -c:v mpeg4 -b:v 800k -g 300 -bf 2 -c:a libmp3lame -b:a 128k snatch.avi |
This is a typical DVD ripping example; the input is a VOB file, the
output an AVI file with MPEG-4 video and MP3 audio. Note that in this
command we use B-frames so the MPEG-4 stream is DivX5 compatible, and
GOP size is 300 which means one intra frame every 10 seconds for 29.97fps
input video. Furthermore, the audio stream is MP3-encoded so you need
to enable LAME support by passing --enable-libmp3lame to configure.
The mapping is particularly useful for DVD transcoding
to get the desired audio language.
NOTE: To see the supported input formats, use ffmpeg -formats.
For extracting images from a video:
ffmpeg -i foo.avi -r 1 -s WxH -f image2 foo-%03d.jpeg |
This will extract one video frame per second from the video and will output them in files named `foo-001.jpeg', `foo-002.jpeg', etc. Images will be rescaled to fit the new WxH values.
If you want to extract just a limited number of frames, you can use the above command in combination with the -vframes or -t option, or in combination with -ss to start extracting from a certain point in time.
For creating a video from many images:
ffmpeg -f image2 -i foo-%03d.jpeg -r 12 -s WxH foo.avi |
The syntax foo-%03d.jpeg specifies to use a decimal number
composed of three digits padded with zeroes to express the sequence
number. It is the same syntax supported by the C printf function, but
only formats accepting a normal integer are suitable.
When importing an image sequence, -i also supports expanding shell-like
wildcard patterns (globbing) internally. To lower the chance of interfering
with your actual file names and the shell's glob expansion, you are required
to activate glob meta characters by prefixing them with a single %
character, like in foo-%*.jpeg, foo-%?%?%?.jpeg or
foo-00%[234%]%*.jpeg.
If your filename actually contains a character sequence of a % character
followed by a glob character, you must double the % character to escape
it. Imagine your files begin with %?-foo-, then you could use a glob
pattern like %%?-foo-%*.jpeg. For input patterns that could be both a
printf or a glob pattern, ffmpeg will assume it is a glob pattern.
ffmpeg -i test1.avi -i test2.avi -map 0.3 -map 0.2 -map 0.1 -map 0.0 -c copy test12.nut |
The resulting output file `test12.avi' will contain first four streams from the input file in reverse order.
ffmpeg -i myfile.avi -b 4000k -minrate 4000k -maxrate 4000k -bufsize 1835k out.m2v |
ffmpeg -i src.ext -lmax 21*QP2LAMBDA dst.ext |
When evaluating specific formats, FFmpeg uses internal library parsing functions, shared by the tools. This section documents the syntax of some of these formats.
The accepted syntax is:
[(YYYY-MM-DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...]]])|(HHMMSS[.m...]]]))[Z] now |
If the value is "now" it takes the current time.
Time is local time unless Z is appended, in which case it is interpreted as UTC. If the year-month-day part is not specified it takes the current year-month-day.
The accepted syntax is:
[-]HH:MM:SS[.m...] [-]S+[.m...] |
HH expresses the number of hours, MM the number a of minutes and SS the number of seconds.
Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation.
The following abbreviations are recognized:
128x96
176x144
352x288
704x576
1408x1152
160x120
320x240
640x480
800x600
1024x768
1600x1200
2048x1536
1280x1024
2560x2048
5120x4096
852x480
1366x768
1600x1024
1920x1200
2560x1600
3200x2048
3840x2400
6400x4096
7680x4800
320x200
640x350
852x480
1280x720
1920x1080
Specify the frame rate of a video, expressed as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation.
The following abbreviations are recognized:
30000/1001
25/1
30000/1
25/1
30000/1
25/1
24/1
24000/1
A ratio can be expressed as an expression, or in the form numerator:denominator.
Note that a ratio with infinite (1/0) or negative value is considered valid, so you should check on the returned value if you want to exclude those values.
The undefined value can be expressed using the "0:0" string.
It can be the name of a color (case insensitive match) or a [0x|#]RRGGBB[AA] sequence, possibly followed by "" and a string representing the alpha component.
The alpha component may be a string composed by "0x" followed by an hexadecimal number or a decimal number between 0.0 and 1.0, which represents the opacity value (0x00/0.0 means completely transparent, 0xff/1.0 completely opaque). If the alpha component is not specified then 0xff is assumed.
The string "random" will result in a random color.
When evaluating an arithmetic expression, FFmpeg uses an internal formula evaluator, implemented through the `libavutil/eval.h' interface.
An expression may contain unary, binary operators, constants, and functions.
Two expressions expr1 and expr2 can be combined to form another expression "expr1;expr2". expr1 and expr2 are evaluated in turn, and the new expression evaluates to the value of expr2.
The following binary operators are available: +, -,
*, /, ^.
The following unary operators are available: +, -.
The following functions are available:
Return 1.0 if x is NAN, 0.0 otherwise.
Allow to store the value of the expression expr in an internal variable. var specifies the number of the variable where to store the value, and it is a value ranging from 0 to 9. The function returns the value stored in the internal variable. Note, Variables are currently not shared between expressions.
Allow to load the value of the internal variable with number var, which was previously stored with st(var, expr). The function returns the loaded value.
Evaluate expression expr while the expression cond is non-zero, and returns the value of the last expr evaluation, or NAN if cond was always false.
Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".
Round the value of expression expr downwards to the nearest integer. For example, "floor(-1.5)" is "-2.0".
Round the value of expression expr towards zero to the nearest integer. For example, "trunc(-1.5)" is "-1.0".
Compute the square root of expr. This is equivalent to "(expr)^.5".
Return 1.0 if expr is zero, 0.0 otherwise.
Compute the power of x elevated y, it is equivalent to "(x)^(y)".
Return a pseudo random value between 0.0 and 1.0. x is the index of the internal variable which will be used to save the seed/state.
This function is similar to the C function with the same name; it returns "sqrt(x*x + y*y)", the length of the hypotenuse of a right triangle with sides of length x and y, or the distance of the point (x, y) from the origin.
Return the greatest common divisor of x and y. If both x and y are 0 or either or both are less than zero then behavior is undefined.
Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.
Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.
Evaluate a taylor series at x. expr represents the LD(id)-th derivates of f(x) at 0. If id is not specified then 0 is assumed. note, when you have the derivatives at y instead of 0 taylor(expr, x-y) can be used When the series does not converge the results are undefined.
Finds x where f(x)=0 in the interval 0..max. f() must be continuous or the result is undefined.
The following constants are available:
area of the unit disc, approximately 3.14
exp(1) (Euler's number), approximately 2.718
golden ratio (1+sqrt(5))/2, approximately 1.618
Assuming that an expression is considered "true" if it has a non-zero value, note that:
* works like AND
+ works like OR
and the construct:
if A then B else C |
is equivalent to
if(A,B) + ifnot(A,C) |
In your C code, you can extend the list of unary and binary functions, and define recognized constants, so that they are available for your expressions.
The evaluator also recognizes the International System number postfixes. If 'i' is appended after the postfix, powers of 2 are used instead of powers of 10. The 'B' postfix multiplies the value for 8, and can be appended after another postfix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as postfix.
Follows the list of available International System postfixes, with indication of the corresponding powers of 10 and of 2.
-24 / -80
-21 / -70
-18 / -60
-15 / -50
-12 / -40
-9 / -30
-6 / -20
-3 / -10
-2
-1
2
3 / 10
3 / 10
6 / 20
9 / 30
12 / 40
15 / 40
18 / 50
21 / 60
24 / 70
Decoders are configured elements in FFmpeg which allow the decoding of multimedia streams.
When you configure your FFmpeg build, all the supported native decoders
are enabled by default. Decoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available decoders using the configure option --list-decoders.
You can disable all the decoders with the configure option
--disable-decoders and selectively enable / disable single decoders
with the options --enable-decoder=DECODER /
--disable-decoder=DECODER.
The option -codecs of the ff* tools will display the list of
enabled decoders.
A description of some of the currently available video decoders follows.
Raw video decoder.
This decoder decodes rawvideo streams.
Specify the assumed field type of the input video.
the video is assumed to be progressive (default)
bottom-field-first is assumed
top-field-first is assumed
Internal wave synthetizer.
This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.
libcelt decoder wrapper
libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec.
Requires the presence of the libcelt headers and library during configuration.
You need to explicitly configure the build with --enable-libcelt.
libgsm decoder wrapper
libgsm allows libavcodec to decode the GSM full rate audio codec. Requires
the presence of the libgsm headers and library during configuration. You need
to explicitly configure the build with --enable-libgsm.
This decoder supports both the ordinary GSM and the Microsoft variant.
libilbc decoder wrapper
libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC)
audio codec. Requires the presence of the libilbc headers and library during
configuration. You need to explicitly configure the build with
--enable-libilbc.
The following option is supported by the libilbc wrapper.
Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).
libopencore-amrnb decoder wrapper
libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate
Narrowband audio codec. Using it requires the presence of the
libopencore-amrnb headers and library during configuration. You need to
explicitly configure the build with --enable-libopencore-amrnb.
An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.
libopencore-amrwb decoder wrapper.
libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate
Wideband audio codec. Using it requires the presence of the
libopencore-amrwb headers and library during configuration. You need to
explicitly configure the build with --enable-libopencore-amrwb.
An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.
Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.
When you configure your FFmpeg build, all the supported native encoders
are enabled by default. Encoders requiring an external library must be enabled
manually via the corresponding --enable-lib option. You can list all
available encoders using the configure option --list-encoders.
You can disable all the encoders with the configure option
--disable-encoders and selectively enable / disable single encoders
with the options --enable-encoder=ENCODER /
--disable-encoder=ENCODER.
The option -codecs of the ff* tools will display the list of
enabled encoders.
A description of some of the currently available audio encoders follows.
AC-3 audio encoders.
These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).
The ac3 encoder uses floating-point math, while the ac3_fixed
encoder only uses fixed-point integer math. This does not mean that one is
always faster, just that one or the other may be better suited to a
particular system. The floating-point encoder will generally produce better
quality audio for a given bitrate. The ac3_fixed encoder is not the
default codec for any of the output formats, so it must be specified explicitly
using the option -acodec ac3_fixed in order to use it.
The AC-3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below.
These parameters are described in detail in several publicly-available documents.
Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.
The metadata values set at initialization will be used for every frame in the stream. (default)
Metadata values can be changed before encoding each frame.
Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:
Apply -3dB gain
Apply -4.5dB gain (default)
Apply -6dB gain
Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:
Apply -3dB gain
Apply -6dB gain (default)
Silence Surround Channel(s)
Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
Mixing Level. Specifies peak sound pressure level (SPL) in the production
environment when the mix was mastered. Valid values are 80 to 111, or -1 for
unknown or not indicated. The default value is -1, but that value cannot be
used if the Audio Production Information is written to the bitstream. Therefore,
if the room_type option is not the default value, the mixing_level
option must not be -1.
Room Type. Describes the equalization used during the final mixing session at
the studio or on the dubbing stage. A large room is a dubbing stage with the
industry standard X-curve equalization; a small room has flat equalization.
This field will not be written to the bitstream if both the mixing_level
option and the room_type option have the default values.
Not Indicated (default)
Large Room
Small Room
Copyright Indicator. Specifies whether a copyright exists for this audio.
No Copyright Exists (default)
Copyright Exists
Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital 100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of -31dB will result in no volume level change, relative to the source volume, during audio reproduction. Valid values are whole numbers in the range -31 to -1, with -31 being the default.
Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean the encoder will actually apply Dolby Surround processing.
Not Indicated (default)
Not Dolby Surround Encoded
Dolby Surround Encoded
Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.
Not Original Source
Original Source (default)
The extended bitstream options are part of the Alternate Bit Stream Syntax as
specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts.
If any one parameter in a group is specified, all values in that group will be
written to the bitstream. Default values are used for those that are written
but have not been specified. If the mixing levels are written, the decoder
will use these values instead of the ones specified in the center_mixlev
and surround_mixlev options if it supports the Alternate Bit Stream
Syntax.
Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.
Not Indicated (default)
Lt/Rt Downmix Preferred
Lo/Ro Downmix Preferred
Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lt/Rt mode.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode.
Apply +3dB gain
Apply +1.5dB gain
Apply 0dB gain
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain (default)
Apply -6.0dB gain
Silence Center Channel
Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode.
Apply -1.5dB gain
Apply -3.0dB gain
Apply -4.5dB gain
Apply -6.0dB gain (default)
Silence Surround Channel(s)
Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.
Not Indicated (default)
Dolby Surround EX Off
Dolby Surround EX On
Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually apply Dolby Headphone processing.
Not Indicated (default)
Dolby Headphone Off
Dolby Headphone On
A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.
Standard A/D Converter (default)
HDCD A/D Converter
Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC-3 feature that increases quality by selectively encoding the left/right channels as mid/side. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes.
These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point.
Enables/Disables use of channel coupling, which is an optional AC-3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.
Selected by Encoder (default)
Disable Channel Coupling
Enable Channel Coupling
Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled.
Selected by Encoder (default)
LAME (Lame Ain't an MP3 Encoder) MP3 encoder wrapper
Requires the presence of the libmp3lame headers and library during
configuration. You need to explicitly configure the build with
--enable-libmp3lame.
The following options are supported by the libmp3lame wrapper, the LAME-equivalent options follow the FFmpeg ones.
FFmpeg | LAME |
b | b
FFmpeg |
q | V Quality setting for VBR. |
compression_level | q Algorithm quality. Valid options are integers from 0-9. |
reservoir | N.A. Enable use of bit reservoir. LAME has this enabled by default. |
joint_stereo | -m j Enables the the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo. |
OpenCORE Adaptive Multi-Rate Narrowband encoder.
Requires the presence of the libopencore-amrnb headers and library during
configuration. You need to explicitly configure the build with
--enable-libopencore-amrnb --enable-version3.
This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting `strict' to `unofficial' or lower.
Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
VisualOn AAC encoder
Requires the presence of the libvo-aacenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-aacenc --enable-version3.
The VisualOn AAC encoder only support encoding AAC-LC and up to 2 channels. It is also CBR-only. It is considered to be worse than the native experimental FFmpeg AAC encoder.
Bitrate.
VisualOn Adaptive Multi-Rate Wideband encoder
Requires the presence of the libvo-amrwbenc headers and library during
configuration. You need to explicitly configure the build with
--enable-libvo-amrwbenc --enable-version3.
This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting `strict' to `unofficial' or lower.
Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.
Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).
A description of some of the currently available video encoders follows.
VP8 format supported through libvpx.
Requires the presence of the libvpx headers and library during configuration.
You need to explicitly configure the build with --enable-libvpx.
Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
g_threads
g_profile
rc_target_bitrate
kf_max_dist
kf_min_dist
rc_min_quantizer
rc_max_quantizer
rc_buf_sz
(bufsize * 1000 / vb)
rc_buf_optimal_sz
(bufsize * 1000 / vb * 5 / 6)
rc_buf_initial_sz
(rc_init_occupancy * 1000 / vb)
rc_undershoot_pct
rc_dropframe_thresh
rc_2pass_vbr_bias_pct
rc_2pass_vbr_maxsection_pct
(maxrate * 100 / vb)
rc_2pass_vbr_minsection_pct
(minrate * 100 / vb)
VPX_CBR
(minrate == maxrate == vb)
VPX_CQ, VP8E_SET_CQ_LEVEL
VPX_DL_BEST_QUALITY
VPX_DL_GOOD_QUALITY
VPX_DL_REALTIME
VP8E_SET_CPUUSED
VP8E_SET_NOISE_SENSITIVITY
VP8E_SET_STATIC_THRESHOLD
VP8E_SET_TOKEN_PARTITIONS
VP8E_SET_ENABLEAUTOALTREF
VP8E_SET_ARNR_MAXFRAMES
VP8E_SET_ARNR_TYPE
VP8E_SET_ARNR_STRENGTH
g_lag_in_frames
g_error_resilient
For more information about libvpx see: http://www.webmproject.org/
H.264 / AVC / MPEG-4 AVC / MPEG-4 part 10 format supported through libx264.
Requires the presence of the libx264 headers and library during
configuration. You need to explicitly configure the build with
--enable-libx264.
Set the encoding preset.
Tune the encoding params.
Use fast settings when encoding first pass, default value is 1.
Set profile restrictions.
Specify level (as defined by Annex A). Deprecated in favor of x264opts.
Specify filename for 2 pass stats. Deprecated in favor of x264opts (see stats libx264 option).
Specify Weighted prediction for P-frames. Deprecated in favor of x264opts (see weightp libx264 option).
Allow to set any x264 option, see x264 -fullhelp for a list.
options is a list of key=value couples separated by ":".
For example to specify libx264 encoding options with ffmpeg:
ffmpeg -i foo.mpg -vcodec libx264 -x264opts keyint=123:min-keyint=20 -an out.mkv |
For more information about libx264 and the supported options see: http://www.videolan.org/developers/x264.html
Xvid MPEG-4 Part 2 encoder wrapper.
This encoder requires the presence of the libxvidcore headers and library
during configuration. You need to explicitly configure the build with
--enable-libxvid --enable-gpl.
The native mpeg4 encoder supports the MPEG-4 Part 2 format, so
users can encode to this format without this library.
The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. The other shared options which are not listed have no effect for the libxvid encoder.
Set specific encoding flags. Possible values:
Use four motion vector by macroblock.
Enable high quality AC prediction.
Only encode grayscale.
Enable the use of global motion compensation (GMC).
Enable quarter-pixel motion compensation.
Enable closed GOP.
Place global headers in extradata instead of every keyframe.
Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:
Use no motion estimation (default).
Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. `x1' and `log' are aliases for `phods'.
Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma planes.
Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.
Set macroblock decision algorithm. Possible values in the increasing order of quality:
Use macroblock comparing function algorithm (default).
Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.
Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern.
Demuxers are configured elements in FFmpeg which allow to read the multimedia streams from a particular type of file.
When you configure your FFmpeg build, all the supported demuxers are enabled by default. You can list all available ones using the configure option "-list-demuxers".
You can disable all the demuxers using the configure option "-disable-demuxers", and selectively enable a single demuxer with the option "-enable-demuxer=DEMUXER", or disable it with the option "-disable-demuxer=DEMUXER".
The option "-formats" of the ff* tools will display the list of enabled demuxers.
The description of some of the currently available demuxers follows.
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The pattern may contain the string "%d" or "%0Nd", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character '%' can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between 0 and 4, all the following numbers must be sequential. This limitation may be hopefully fixed.
The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files.
For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form `img-001.bmp', `img-002.bmp', ..., `img-010.bmp', etc.; the pattern "i%%m%%g-%d.jpg" will match a sequence of filenames of the form `i%m%g-1.jpg', `i%m%g-2.jpg', ..., `i%m%g-10.jpg', etc.
The size, the pixel format, and the format of each image must be the same for all the files in the sequence.
The following example shows how to use ffmpeg for creating a
video from the images in the file sequence `img-001.jpeg',
`img-002.jpeg', ..., assuming an input frame rate of 10 frames per
second:
ffmpeg -i 'img-%03d.jpeg' -r 10 out.mkv |
Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a single image file `img.jpeg' you can employ the command:
ffmpeg -i img.jpeg img.png |
Apple HTTP Live Streaming demuxer.
This demuxer presents all AVStreams from all variant streams. The id field is set to the bitrate variant index number. By setting the discard flags on AVStreams (by pressing 'a' or 'v' in ffplay), the caller can decide which variant streams to actually receive. The total bitrate of the variant that the stream belongs to is available in a metadata key named "variant_bitrate".
SBaGen script demuxer.
This demuxer reads the script language used by SBaGen http://uazu.net/sbagen/ to generate binaural beats sessions. A SBG script looks like that:
-SE a: 300-2.5/3 440+4.5/0 b: 300-2.5/0 440+4.5/3 off: - NOW == a +0:07:00 == b +0:14:00 == a +0:21:00 == b +0:30:00 off |
A SBG script can mix absolute and relative timestamps. If the script uses either only absolute timestamps (including the script start time) or only relative ones, then its layout is fixed, and the conversion is straightforward. On the other hand, if the script mixes both kind of timestamps, then the NOW reference for relative timestamps will be taken from the current time of day at the time the script is read, and the script layout will be frozen according to that reference. That means that if the script is directly played, the actual times will match the absolute timestamps up to the sound controller's clock accuracy, but if the user somehow pauses the playback or seeks, all times will be shifted accordingly.
Muxers are configured elements in FFmpeg which allow writing multimedia streams to a particular type of file.
When you configure your FFmpeg build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option --list-muxers.
You can disable all the muxers with the configure option
--disable-muxers and selectively enable / disable single muxers
with the options --enable-muxer=MUXER /
--disable-muxer=MUXER.
The option -formats of the ff* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
Audio Interchange File Format muxer.
It accepts the following options:
Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).
Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file `out.crc':
ffmpeg -i INPUT -f crc out.crc |
You can print the CRC to stdout with the command:
ffmpeg -i INPUT -f crc - |
You can select the output format of each frame with ffmpeg by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - |
See also the framecrc muxer.
Per-packet CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a line for each audio and video packet of the form:
stream_index, packet_dts, packet_pts, packet_duration, packet_size, 0xCRC |
CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the packet.
For example to compute the CRC of the audio and video frames in `INPUT', converted to raw audio and video packets, and store it in the file `out.crc':
ffmpeg -i INPUT -f framecrc out.crc |
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framecrc - |
With ffmpeg, you can select the output format to which the
audio and video frames are encoded before computing the CRC for each
packet by specifying the audio and video codec. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
ffmpeg -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - |
See also the crc muxer.
Per-packet MD5 testing format.
This muxer computes and prints the MD5 hash for each audio and video packet. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash.
The output of the muxer consists of a line for each audio and video packet of the form:
stream_index, packet_dts, packet_pts, packet_duration, packet_size, MD5 |
MD5 is a hexadecimal number representing the computed MD5 hash for the packet.
For example to compute the MD5 of the audio and video frames in `INPUT', converted to raw audio and video packets, and store it in the file `out.md5':
ffmpeg -i INPUT -f framemd5 out.md5 |
To print the information to stdout, use the command:
ffmpeg -i INPUT -f framemd5 - |
See also the md5 muxer.
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character '%' can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.
The pattern may contain a suffix which is used to automatically determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form `img-001.bmp', `img-002.bmp', ..., `img-010.bmp', etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form `img%-1.jpg', `img%-2.jpg', ..., `img%-10.jpg', etc.
The following example shows how to use ffmpeg for creating a
sequence of files `img-001.jpeg', `img-002.jpeg', ...,
taking one image every second from the input video:
ffmpeg -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg' |
Note that with ffmpeg, if the format is not specified with the
-f option and the output filename specifies an image file
format, the image2 muxer is automatically selected, so the previous
command can be written as:
ffmpeg -i in.avi -vsync 1 -r 1 'img-%03d.jpeg' |
Note also that the pattern must not necessarily contain "%d" or "%0Nd", for example to create a single image file `img.jpeg' from the input video you can employ the command:
ffmpeg -i in.avi -f image2 -frames:v 1 img.jpeg |
The image muxer supports the .Y.U.V image file format. This format is special in that that each image frame consists of three files, for each of the YUV420P components. To read or write this image file format, specify the name of the '.Y' file. The muxer will automatically open the '.U' and '.V' files as required.
MD5 testing format.
This muxer computes and prints the MD5 hash of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the hash.
The output of the muxer consists of a single line of the form: MD5=MD5, where MD5 is a hexadecimal number representing the computed MD5 hash.
For example to compute the MD5 hash of the input converted to raw audio and video, and store it in the file `out.md5':
ffmpeg -i INPUT -f md5 out.md5 |
You can print the MD5 to stdout with the command:
ffmpeg -i INPUT -f md5 - |
See also the framemd5 muxer.
The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4
file has all the metadata about all packets stored in one location
(written at the end of the file, it can be moved to the start for
better playback using the qt-faststart tool). A fragmented
file consists of a number of fragments, where packets and metadata
about these packets are stored together. Writing a fragmented
file has the advantage that the file is decodable even if the
writing is interrupted (while a normal MOV/MP4 is undecodable if
it is not properly finished), and it requires less memory when writing
very long files (since writing normal MOV/MP4 files stores info about
every single packet in memory until the file is closed). The downside
is that it is less compatible with other applications.
Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments:
Reserves space for the moov atom at the beginning of the file instead of placing the moov atom at the end. If the space reserved is insufficient, muxing will fail.
Start a new fragment at each video keyframe.
Create fragments that are duration microseconds long.
Create fragments that contain up to size bytes of payload data.
Allow the caller to manually choose when to cut fragments, by
calling av_write_frame(ctx, NULL) to write a fragment with
the packets written so far. (This is only useful with other
applications integrating libavformat, not from ffmpeg.)
Don't create fragments that are shorter than duration microseconds long.
If more than one condition is specified, fragments are cut when
one of the specified conditions is fulfilled. The exception to this is
-min_frag_duration, which has to be fulfilled for any of the other
conditions to apply.
Additionally, the way the output file is written can be adjusted through a few other options:
Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration.
Files written with this option set do not work in QuickTime. This option is implicitly set when writing ismv (Smooth Streaming) files.
Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example:
ffmpeg -re <normal input/transcoding options> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) |
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
Set the original_network_id (default 0x0001). This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path Original_Network_ID, Transport_Stream_ID.
Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB.
Set the service_id (default 0x0001) also known as program in DVB.
Set the first PID for PMT (default 0x1000, max 0x1f00).
Set the first PID for data packets (default 0x0100, max 0x0f00).
The recognized metadata settings in mpegts muxer are service_provider
and service_name. If they are not set the default for
service_provider is "FFmpeg" and the default for
service_name is "Service01".
ffmpeg -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
|
Null muxer.
This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes.
For example to benchmark decoding with ffmpeg you can use the
command:
ffmpeg -benchmark -i INPUT -f null out.null |
Note that the above command does not read or write the `out.null'
file, but specifying the output file is required by the ffmpeg
syntax.
Alternatively you can write the command as:
ffmpeg -benchmark -i INPUT -f null - |
Matroska container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
Name provided to a single track
Specifies the language of the track in the Matroska languages form
Stereo 3D video layout of two views in a single video track
video is not stereo
Both views are arranged side by side, Left-eye view is on the left
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
Each view is constituted by a row based interleaving, Right-eye view is first row
Each view is constituted by a row based interleaving, Left-eye view is first row
Both views are arranged in a column based interleaving manner, Right-eye view is first column
Both views are arranged in a column based interleaving manner, Left-eye view is first column
All frames are in anaglyph format viewable through red-cyan filters
Both views are arranged side by side, Right-eye view is on the left
All frames are in anaglyph format viewable through green-magenta filters
Both eyes laced in one Block, Left-eye view is first
Both eyes laced in one Block, Right-eye view is first
For example a 3D WebM clip can be created using the following command line:
ffmpeg -i sample_left_right_clip.mpg -an -c:v libvpx -metadata stereo_mode=left_right -y stereo_clip.webm |
Basic stream segmenter.
The segmenter muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to image2.
Every segment starts with a video keyframe, if a video stream is present. The segment muxer works best with a single constant frame rate video.
Optionally it can generate a flat list of the created segments, one segment per line.
Override the inner container format, by default it is guessed by the filename extension.
Set segment duration to t seconds.
Generate also a listfile named name.
Overwrite the listfile once it reaches size entries.
Wrap around segment index once it reaches limit.
ffmpeg -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut |
The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and
optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the
id3v2_version option controls which one is used. The legacy ID3v1 tag is
not written by default, but may be enabled with the write_id3v1 option.
For seekable output the muxer also writes a Xing frame at the beginning, which contains the number of frames in the file. It is useful for computing duration of VBR files.
The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures are supplied to the muxer in form of a video stream with a single packet. There can be any number of those streams, each will correspond to a single APIC frame. The stream metadata tags title and comment map to APIC description and picture type respectively. See http://id3.org/id3v2.4.0-frames for allowed picture types.
Note that the APIC frames must be written at the beginning, so the muxer will buffer the audio frames until it gets all the pictures. It is therefore advised to provide the pictures as soon as possible to avoid excessive buffering.
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
ffmpeg -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 |
Attach a picture to an mp3:
ffmpeg -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 |
Input devices are configured elements in FFmpeg which allow to access the data coming from a multimedia device attached to your system.
When you configure your FFmpeg build, all the supported input devices are enabled by default. You can list all available ones using the configure option "-list-indevs".
You can disable all the input devices using the configure option "-disable-indevs", and selectively enable an input device using the option "-enable-indev=INDEV", or you can disable a particular input device using the option "-disable-indev=INDEV".
The option "-formats" of the ff* tools will display the list of supported input devices (amongst the demuxers).
A description of the currently available input devices follows.
ALSA (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound installed on your system.
This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
hw:CARD[,DEV[,SUBDEV]] |
where the DEV and SUBDEV components are optional.
The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any).
To see the list of cards currently recognized by your system check the files `/proc/asound/cards' and `/proc/asound/devices'.
For example to capture with ffmpeg from an ALSA device with
card id 0, you may run the command:
ffmpeg -f alsa -i hw:0 alsaout.wav |
For more information see: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html
BSD video input device.
Windows DirectShow input device.
DirectShow support is enabled when FFmpeg is built with the mingw-w64 project. Currently only audio and video devices are supported.
Multiple devices may be opened as separate inputs, but they may also be opened on the same input, which should improve synchronism between them.
The input name should be in the format:
TYPE=NAME[:TYPE=NAME] |
where TYPE can be either audio or video, and NAME is the device's name.
If no options are specified, the device's defaults are used. If the device does not support the requested options, it will fail to open.
Set the video size in the captured video.
Set the framerate in the captured video.
Set the sample rate (in Hz) of the captured audio.
Set the sample size (in bits) of the captured audio.
Set the number of channels in the captured audio.
If set to `true', print a list of devices and exit.
If set to `true', print a list of selected device's options and exit.
Set video device number for devices with same name (starts at 0, defaults to 0).
Set audio device number for devices with same name (starts at 0, defaults to 0).
$ ffmpeg -list_devices true -f dshow -i dummy |
$ ffmpeg -f dshow -i video="Camera" |
$ ffmpeg -f dshow -video_device_number 1 -i video="Camera" |
$ ffmpeg -f dshow -i video="Camera":audio="Microphone" |
$ ffmpeg -list_options true -f dshow -i video="Camera" |
Linux DV 1394 input device.
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually `/dev/fb0'.
For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device `/dev/fb0' with
ffmpeg:
ffmpeg -f fbdev -r 10 -i /dev/fb0 out.avi |
You can take a single screenshot image with the command:
ffmpeg -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg |
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
JACK input device.
To enable this input device during configuration you need libjack installed on your system.
A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the FFmpeg input device.
Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the jack_connect
and jack_disconnect programs, or do it through a graphical interface,
for example with qjackctl.
To list the JACK clients and their properties you can invoke the command
jack_lsp.
Follows an example which shows how to capture a JACK readable client
with ffmpeg.
# Create a JACK writable client with name "ffmpeg". $ ffmpeg -f jack -i ffmpeg -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 # List the current JACK clients. $ jack_lsp -c system:capture_1 system:capture_2 system:playback_1 system:playback_2 ffmpeg:input_1 metro:120_bpm # Connect metro to the ffmpeg writable client. $ jack_connect metro:120_bpm ffmpeg:input_1 |
For more information read: http://jackaudio.org/
Libavfilter input virtual device.
This input device reads data from the open output pads of a libavfilter filtergraph.
For each filtergraph open output, the input device will create a corresponding stream which is mapped to the generated output. Currently only video data is supported. The filtergraph is specified through the option `graph'.
Specify the filtergraph to use as input. Each video open output must be labelled by a unique string of the form "outN", where N is a number starting from 0 corresponding to the mapped input stream generated by the device. The first unlabelled output is automatically assigned to the "out0" label, but all the others need to be specified explicitly.
If not specified defaults to the filename specified for the input device.
ffplay:
ffplay -f lavfi -graph "color=pink [out0]" dummy |
ffplay -f lavfi color=pink |
ffplay -f lavfi -graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3 |
ffplay:
ffplay -f lavfi "amovie=test.wav" |
ffplay:
ffplay -f lavfi "movie=test.avi[out0];amovie=test.wav[out1]" |
IIDC1394 input device, based on libdc1394 and libraw1394.
The OpenAL input device provides audio capture on all systems with a working OpenAL 1.1 implementation.
To enable this input device during configuration, you need OpenAL
headers and libraries installed on your system, and need to configure
FFmpeg with --enable-openal.
OpenAL headers and libraries should be provided as part of your OpenAL
implementation, or as an additional download (an SDK). Depending on your
installation you may need to specify additional flags via the
--extra-cflags and --extra-ldflags for allowing the build
system to locate the OpenAL headers and libraries.
An incomplete list of OpenAL implementations follows:
The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See http://openal.org/.
Portable, open source (LGPL) software implementation. Includes backends for the most common sound APIs on the Windows, Linux, Solaris, and BSD operating systems. See http://kcat.strangesoft.net/openal.html.
OpenAL is part of Core Audio, the official Mac OS X Audio interface. See http://developer.apple.com/technologies/mac/audio-and-video.html
This device allows to capture from an audio input device handled through OpenAL.
You need to specify the name of the device to capture in the provided filename. If the empty string is provided, the device will automatically select the default device. You can get the list of the supported devices by using the option list_devices.
Set the number of channels in the captured audio. Only the values `1' (monaural) and `2' (stereo) are currently supported. Defaults to `2'.
Set the sample size (in bits) of the captured audio. Only the values `8' and `16' are currently supported. Defaults to `16'.
Set the sample rate (in Hz) of the captured audio. Defaults to `44.1k'.
If set to `true', print a list of devices and exit. Defaults to `false'.
Print the list of OpenAL supported devices and exit:
$ ffmpeg -list_devices true -f openal -i dummy out.ogg |
Capture from the OpenAL device `DR-BT101 via PulseAudio':
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out.ogg |
Capture from the default device (note the empty string " as filename):
$ ffmpeg -f openal -i '' out.ogg |
Capture from two devices simultaneously, writing to two different files,
within the same ffmpeg command:
$ ffmpeg -f openal -i 'DR-BT101 via PulseAudio' out1.ogg -f openal -i 'ALSA Default' out2.ogg |
Note: not all OpenAL implementations support multiple simultaneous capture - try the latest OpenAL Soft if the above does not work.
Open Sound System input device.
The filename to provide to the input device is the device node representing the OSS input device, and is usually set to `/dev/dsp'.
For example to grab from `/dev/dsp' using ffmpeg use the
command:
ffmpeg -f oss -i /dev/dsp /tmp/oss.wav |
For more information about OSS see: http://manuals.opensound.com/usersguide/dsp.html
pulseaudio input device.
To enable this input device during configuration you need libpulse-simple installed in your system.
The filename to provide to the input device is a source device or the string "default"
To list the pulse source devices and their properties you can invoke
the command pactl list sources.
ffmpeg -f pulse -i default /tmp/pulse.wav |
The syntax is:
-server server name |
Connects to a specific server.
The syntax is:
-name application name |
Specify the application name pulse will use when showing active clients, by default it is the LIBAVFORMAT_IDENT string
The syntax is:
-stream_name stream name |
Specify the stream name pulse will use when showing active streams, by default it is "record"
The syntax is:
-sample_rate samplerate |
Specify the samplerate in Hz, by default 48kHz is used.
The syntax is:
-channels N |
Specify the channels in use, by default 2 (stereo) is set.
The syntax is:
-frame_size bytes |
Specify the number of byte per frame, by default it is set to 1024.
The syntax is:
-fragment_size bytes |
Specify the minimal buffering fragment in pulseaudio, it will affect the audio latency. By default it is unset.
sndio input device.
To enable this input device during configuration you need libsndio installed on your system.
The filename to provide to the input device is the device node representing the sndio input device, and is usually set to `/dev/audio0'.
For example to grab from `/dev/audio0' using ffmpeg use the
command:
ffmpeg -f sndio -i /dev/audio0 /tmp/oss.wav |
Video4Linux2 input video device.
The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind `/dev/videoN', where N is a number associated to the device.
Video4Linux2 devices usually support a limited set of
widthxheight sizes and framerates. You can check which are
supported using -list_formats all for Video4Linux2 devices.
Some usage examples of the video4linux2 devices with ffmpeg and ffplay:
The time base for the timestamps is 1 microsecond. Depending on the kernel version and configuration, the timestamps may be derived from the real time clock (origin at the Unix Epoch) or the monotonic clock (origin usually at boot time, unaffected by NTP or manual changes to the clock). The `-timestamps abs' or `-ts abs' option can be used to force conversion into the real time clock.
Note that if FFmpeg is build with v4l-utils support ("-enable-libv4l2" option), it will always be used.
# Grab and show the input of a video4linux2 device. ffplay -f video4linux2 -framerate 30 -video_size hd720 /dev/video0 # Grab and record the input of a video4linux2 device, leave the framerate and size as previously set. ffmpeg -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg |
"v4l" and "v4l2" can be used as aliases for the respective "video4linux" and "video4linux2".
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from 0 to 9. You may use "list" as filename to print a list of drivers. Any other filename will be interpreted as device number 0.
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
[hostname]:display_number.screen_number[+x_offset,y_offset] |
hostname:display_number.screen_number specifies the
X11 display name of the screen to grab from. hostname can be
omitted, and defaults to "localhost". The environment variable
DISPLAY contains the default display name.
x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the dpyinfo program for getting basic information about the
properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from `:0.0' using ffmpeg:
ffmpeg -f x11grab -r 25 -s cif -i :0.0 out.mpg # Grab at position 10,20. ffmpeg -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg |
The syntax is:
-follow_mouse centered|PIXELS |
When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within PIXELS (greater than zero) to the edge of region.
For example:
ffmpeg -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg # Follows only when the mouse pointer reaches within 100 pixels to edge ffmpeg -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg |
The syntax is:
-show_region 1 |
If show_region AVOption is specified with 1, then the grabbing region will be indicated on screen. With this option, it's easy to know what is being grabbed if only a portion of the screen is grabbed.
For example:
ffmpeg -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg # With follow_mouse ffmpeg -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg |
Output devices are configured elements in FFmpeg which allow to write multimedia data to an output device attached to your system.
When you configure your FFmpeg build, all the supported output devices are enabled by default. You can list all available ones using the configure option "-list-outdevs".
You can disable all the output devices using the configure option "-disable-outdevs", and selectively enable an output device using the option "-enable-outdev=OUTDEV", or you can disable a particular input device using the option "-disable-outdev=OUTDEV".
The option "-formats" of the ff* tools will display the list of enabled output devices (amongst the muxers).
A description of the currently available output devices follows.
ALSA (Advanced Linux Sound Architecture) output device.
OSS (Open Sound System) output device.
SDL (Simple DirectMedia Layer) output device.
This output devices allows to show a video stream in an SDL window. Only one SDL window is allowed per application, so you can have only one instance of this output device in an application.
To enable this output device you need libsdl installed on your system when configuring your build.
For more information about SDL, check: http://www.libsdl.org/
Set the SDL window title, if not specified default to the filename specified for the output device.
Set the name of the iconified SDL window, if not specified it is set to the same value of window_title.
Set the SDL window size, can be a string of the form widthxheight or a video size abbreviation. If not specified it defaults to the size of the input video.
The following command shows the ffmpeg output is an
SDL window, forcing its size to the qcif format:
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt yuv420p -window_size qcif -f sdl "SDL output" |
sndio audio output device.
Protocols are configured elements in FFmpeg which allow to access resources which require the use of a particular protocol.
When you configure your FFmpeg build, all the supported protocols are enabled by default. You can list all available ones using the configure option "-list-protocols".
You can disable all the protocols using the configure option "-disable-protocols", and selectively enable a protocol using the option "-enable-protocol=PROTOCOL", or you can disable a particular protocol using the option "-disable-protocol=PROTOCOL".
The option "-protocols" of the ff* tools will display the list of supported protocols.
A description of the currently available protocols follows.
Read BluRay playlist.
The accepted options are:
BluRay angle
Start chapter (1...N)
Playlist to read (BDMV/PLAYLIST/?????.mpls)
Examples:
Read longest playlist from BluRay mounted to /mnt/bluray:
bluray:/mnt/bluray |
Read angle 2 of playlist 4 from BluRay mounted to /mnt/bluray, start from chapter 2:
-playlist 4 -angle 2 -chapter 2 bluray:/mnt/bluray |
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were a unique resource.
A URL accepted by this protocol has the syntax:
concat:URL1|URL2|...|URLN |
where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.
For example to read a sequence of files `split1.mpeg',
`split2.mpeg', `split3.mpeg' with ffplay use the
command:
ffplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg |
Note that you may need to escape the character "|" which is special for many shells.
File access protocol.
Allow to read from or read to a file.
For example to read from a file `input.mpeg' with ffmpeg
use the command:
ffmpeg -i file:input.mpeg output.mpeg |
The ff* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.mpeg".
Gopher protocol.
Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either "file" or "http".
hls+http://host/path/to/remote/resource.m3u8 hls+file://path/to/local/resource.m3u8 |
Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.
HTTP (Hyper Text Transfer Protocol).
MMS (Microsoft Media Server) protocol over TCP.
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
mmsh://server[:port][/app][/playpath] |
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.
Some examples follow.
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5. ffmpeg -i input.flv -f avi -y md5:output.avi.md5 # Write the MD5 hash of the encoded AVI file to stdout. ffmpeg -i input.flv -f avi -y md5: |
Note that some formats (typically MOV) require the output protocol to be seekable, so they will fail with the MD5 output protocol.
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
pipe:[number] |
number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.
For example to read from stdin with ffmpeg:
cat test.wav | ffmpeg -i pipe:0 # ...this is the same as... cat test.wav | ffmpeg -i pipe: |
For writing to stdout with ffmpeg:
ffmpeg -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... ffmpeg -i test.wav -f avi pipe: | cat > test.avi |
Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.
Real-Time Messaging Protocol.
The Real-Time Messaging Protocol (RTMP) is used for streaming multimedia content across a TCP/IP network.
The required syntax is:
rtmp://server[:port][/app][/playpath] |
The accepted parameters are:
The address of the RTMP server.
The number of the TCP port to use (by default is 1935).
It is the name of the application to access. It usually corresponds to the path where the application is installed on the RTMP server (e.g. `/ondemand/', `/flash/live/', etc.).
It is the path or name of the resource to play with reference to the application specified in app, may be prefixed by "mp4:".
For example to read with ffplay a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
ffplay rtmp://myserver/vod/sample |
Real-Time Messaging Protocol and its variants supported through librtmp.
Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "-enable-librtmp". If enabled this will replace the native RTMP protocol.
This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
rtmp_proto://server[:port][/app][/playpath] options |
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
ffmpeg:
ffmpeg -re -i myfile -f flv rtmp://myserver/live/mystream |
To play the same stream using ffplay:
ffplay "rtmp://myserver/live/mystream live=1" |
Real-Time Protocol.
RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock's RTSP server).
The required syntax for a RTSP url is:
rtsp://hostname[:port]/path |
The following options (set on the ffmpeg/ffplay command
line, or set in code via AVOptions or in avformat_open_input),
are supported:
Flags for rtsp_transport:
Use UDP as lower transport protocol.
Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
Use UDP multicast as lower transport protocol.
Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the tcp and udp options are supported.
Flags for rtsp_flags:
Accept packets only from negotiated peer address and port.
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). This
can be disabled by setting the maximum demuxing delay to zero (via
the max_delay field of AVFormatContext).
When watching multi-bitrate Real-RTSP streams with ffplay, the
streams to display can be chosen with -vst n and
-ast n for video and audio respectively, and can be switched
on the fly by pressing v and a.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
ffplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 |
To watch a stream tunneled over HTTP:
ffplay -rtsp_transport http rtsp://server/video.mp4 |
To send a stream in realtime to a RTSP server, for others to watch:
ffmpeg -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp |
Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port.
The syntax for a SAP url given to the muxer is:
sap://destination[:port][?options] |
The RTP packets are sent to destination on port port,
or to port 5004 if no port is specified.
options is a &-separated list. The following options
are supported:
Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.
Specify the port to send the announcements on, defaults to 9875 if not specified.
Specify the time to live value for the announcements and RTP packets, defaults to 255.
If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
ffmpeg -re -i input -f sap sap://224.0.0.255?same_port=1 |
Similarly, for watching in ffplay:
ffmpeg -re -i input -f sap sap://224.0.0.255 |
And for watching in ffplay, over IPv6:
ffmpeg -re -i input -f sap sap://[ff0e::1:2:3:4] |
The syntax for a SAP url given to the demuxer is:
sap://[address][:port] |
address is the multicast address to listen for announcements on, if omitted, the default 224.2.127.254 (sap.mcast.net) is used. port is the port that is listened on, 9875 if omitted.
The demuxers listens for announcements on the given address and port. Once an announcement is received, it tries to receive that particular stream.
Example command lines follow.
To play back the first stream announced on the normal SAP multicast address:
ffplay sap:// |
To play back the first stream announced on one the default IPv6 SAP multicast address:
ffplay sap://[ff0e::2:7ffe] |
Trasmission Control Protocol.
The required syntax for a TCP url is:
tcp://hostname:port[?options] |
Listen for an incoming connection
ffmpeg -i input -f format tcp://hostname:port?listen ffplay tcp://hostname:port |
User Datagram Protocol.
The required syntax for a UDP url is:
udp://hostname:port[?options] |
options contains a list of &-seperated options of the form key=val. Follow the list of supported options.
set the UDP buffer size in bytes
override the local UDP port to bind with
Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.
set the size in bytes of UDP packets
explicitly allow or disallow reusing UDP sockets
set the time to live value (for multicast only)
Initialize the UDP socket with connect(). In this case, the
destination address can't be changed with ff_udp_set_remote_url later.
If the destination address isn't known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
Some usage examples of the udp protocol with ffmpeg follow.
To stream over UDP to a remote endpoint:
ffmpeg -i input -f format udp://hostname:port |
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
ffmpeg -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535 |
To receive over UDP from a remote endpoint:
ffmpeg -i udp://[multicast-address]:port |
When you configure your FFmpeg build, all the supported bitstream
filters are enabled by default. You can list all available ones using
the configure option --list-bsfs.
You can disable all the bitstream filters using the configure option
--disable-bsfs, and selectively enable any bitstream filter using
the option --enable-bsf=BSF, or you can disable a particular
bitstream filter using the option --disable-bsf=BSF.
The option -bsfs of the ff* tools will display the list of
all the supported bitstream filters included in your build.
Below is a description of the currently available bitstream filters.
Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).
This is required by some streaming formats, typically the MPEG-2 transport stream format ("mpegts").
For example to remux an MP4 file containing an H.264 stream to mpegts
format with ffmpeg, you can use the command:
ffmpeg -i INPUT.mp4 -codec copy -bsf:v h264_mp4toannexb OUTPUT.ts |
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.
MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by
ffmpeg -i ../some_mjpeg.avi -c:v copy frames_%d.jpg |
Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml:
Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that "MJPEG, or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed - and *omitted* - Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won't have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec."
This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.
ffmpeg -i mjpeg-movie.avi -c:v copy -bsf:v mjpeg2jpeg frame_%d.jpg exiftran -i -9 frame*.jpg ffmpeg -i frame_%d.jpg -c:v copy rotated.avi |
A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple links between a pair of filters. Each link has one input pad on one side connecting it to one filter from which it takes its input, and one output pad on the other side connecting it to the one filter accepting its output.
Each filter in a filtergraph is an instance of a filter class registered in the application, which defines the features and the number of input and output pads of the filter.
A filter with no input pads is called a "source", a filter with no output pads is called a "sink".
A filtergraph can be represented using a textual representation, which is
recognized by the `-filter'/`-vf' and `-filter_complex'
options in ffmpeg and `-vf' in ffplay, and by the
avfilter_graph_parse()/avfilter_graph_parse2() function defined in
`libavfilter/avfiltergraph.h'.
A filterchain consists of a sequence of connected filters, each one connected to the previous one in the sequence. A filterchain is represented by a list of ","-separated filter descriptions.
A filtergraph consists of a sequence of filterchains. A sequence of filterchains is represented by a list of ";"-separated filterchain descriptions.
A filter is represented by a string of the form: [in_link_1]...[in_link_N]filter_name=arguments[out_link_1]...[out_link_M]
filter_name is the name of the filter class of which the described filter is an instance of, and has to be the name of one of the filter classes registered in the program. The name of the filter class is optionally followed by a string "=arguments".
arguments is a string which contains the parameters used to initialize the filter instance, and are described in the filter descriptions below.
The list of arguments can be quoted using the character "'" as initial and ending mark, and the character '\' for escaping the characters within the quoted text; otherwise the argument string is considered terminated when the next special character (belonging to the set "[]=;,") is encountered.
The name and arguments of the filter are optionally preceded and followed by a list of link labels. A link label allows to name a link and associate it to a filter output or input pad. The preceding labels in_link_1 ... in_link_N, are associated to the filter input pads, the following labels out_link_1 ... out_link_M, are associated to the output pads.
When two link labels with the same name are found in the filtergraph, a link between the corresponding input and output pad is created.
If an output pad is not labelled, it is linked by default to the first unlabelled input pad of the next filter in the filterchain. For example in the filterchain:
nullsrc, split[L1], [L2]overlay, nullsink |
the split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled.
In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected.
Libavfilter will automatically insert scale filters where format
conversion is required. It is possible to specify swscale flags
for those automatically inserted scalers by prepending
sws_flags=flags;
to the filtergraph description.
Follows a BNF description for the filtergraph syntax:
NAME ::= sequence of alphanumeric characters and '_' LINKLABEL ::= "[" NAME "]" LINKLABELS ::= LINKLABEL [LINKLABELS] FILTER_ARGUMENTS ::= sequence of chars (eventually quoted) FILTER ::= [LINKNAMES] NAME ["=" ARGUMENTS] [LINKNAMES] FILTERCHAIN ::= FILTER [,FILTERCHAIN] FILTERGRAPH ::= [sws_flags=flags;] FILTERCHAIN [;FILTERGRAPH] |
When you configure your FFmpeg build, you can disable any of the
existing filters using --disable-filters.
The configure output will show the audio filters included in your
build.
Below is a description of the currently available audio filters.
Convert the input audio format to the specified formats.
The filter accepts a string of the form: "sample_format:channel_layout".
sample_format specifies the sample format, and can be a string or the corresponding numeric value defined in `libavutil/samplefmt.h'. Use 'p' suffix for a planar sample format.
channel_layout specifies the channel layout, and can be a string or the corresponding number value defined in `libavutil/audioconvert.h'.
The special parameter "auto", signifies that the filter will automatically select the output format depending on the output filter.
Some examples follow.
aconvert=fltp:stereo |
aconvert=u8:auto |
Convert the input audio to one of the specified formats. The framework will negotiate the most appropriate format to minimize conversions.
The filter accepts the following named parameters:
A comma-separated list of requested sample formats.
A comma-separated list of requested sample rates.
A comma-separated list of requested channel layouts.
If a parameter is omitted, all values are allowed.
For example to force the output to either unsigned 8-bit or signed 16-bit stereo:
aformat=sample_fmts\=u8\,s16:channel_layouts\=stereo |
Merge two audio streams into a single multi-channel stream.
This filter does not need any argument.
If the channel layouts of the inputs are disjoint, and therefore compatible, the channel layout of the output will be set accordingly and the channels will be reordered as necessary. If the channel layouts of the inputs are not disjoint, the output will have all the channels of the first input then all the channels of the second input, in that order, and the channel layout of the output will be the default value corresponding to the total number of channels.
For example, if the first input is in 2.1 (FL+FR+LF) and the second input is FC+BL+BR, then the output will be in 5.1, with the channels in the following order: a1, a2, b1, a3, b2, b3 (a1 is the first channel of the first input, b1 is the first channel of the second input).
On the other hand, if both input are in stereo, the output channels will be in the default order: a1, a2, b1, b2, and the channel layout will be arbitrarily set to 4.0, which may or may not be the expected value.
Both inputs must have the same sample rate, and format.
If inputs do not have the same duration, the output will stop with the shortest.
Example: merge two mono files into a stereo stream:
amovie=left.wav [l] ; amovie=right.mp3 [r] ; [l] [r] amerge |
If you need to do multiple merges (for instance multiple mono audio streams in a single video media), you can do:
ffmpeg -f lavfi -i " amovie=input.mkv:si=0 [a0]; amovie=input.mkv:si=1 [a1]; amovie=input.mkv:si=2 [a2]; amovie=input.mkv:si=3 [a3]; amovie=input.mkv:si=4 [a4]; amovie=input.mkv:si=5 [a5]; [a0][a1] amerge [x0]; [x0][a2] amerge [x1]; [x1][a3] amerge [x2]; [x2][a4] amerge [x3]; [x3][a5] amerge" -c:a pcm_s16le output.mkv |
Mixes multiple audio inputs into a single output.
For example
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUT |
will mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds.
The filter accepts the following named parameters:
Number of inputs. If unspecified, it defaults to 2.
How to determine the end-of-stream.
Duration of longest input. (default)
Duration of shortest input.
Duration of first input.
Transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
Pass the audio source unchanged to the output.
Resample the input audio to the specified sample rate.
The filter accepts exactly one parameter, the output sample rate. If not specified then the filter will automatically convert between its input and output sample rates.
For example, to resample the input audio to 44100Hz:
aresample=44100 |
Show a line containing various information for each input audio frame. The input audio is not modified.
The shown line contains a sequence of key/value pairs of the form key:value.
A description of each shown parameter follows:
sequential number of the input frame, starting from 0
presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad, and is usually 1/sample_rate.
presentation TimeStamp of the input frame, expressed as a number of seconds
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
sample format name
channel layout description
number of samples (per each channel) contained in the filtered frame
sample rate for the audio frame
Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame
Adler-32 checksum (printed in hexadecimal) for each input frame plane, expressed in the form "[c0 c1 c2 c3 c4 c5 c6 c7]"
Split input audio into several identical outputs.
The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2.
For example:
[in] asplit [out0][out1] |
will create two separate outputs from the same input.
To create 3 or more outputs, you need to specify the number of outputs, like in:
[in] asplit=3 [out0][out1][out2] |
ffmpeg -i INPUT -filter_complex asplit=5 OUTPUT |
will create 5 copies of the input audio.
Forward two audio streams and control the order the buffers are forwarded.
The argument to the filter is an expression deciding which stream should be forwarded next: if the result is negative, the first stream is forwarded; if the result is positive or zero, the second stream is forwarded. It can use the following variables:
number of buffers forwarded so far on each stream
number of samples forwarded so far on each stream
current timestamp of each stream
The default value is t1-t2, which means to always forward the stream
that has a smaller timestamp.
Example: stress-test amerge by randomly sending buffers on the wrong
input, while avoiding too much of a desynchronization:
amovie=file.ogg [a] ; amovie=file.mp3 [b] ; [a] [b] astreamsync=(2*random(1))-1+tanh(5*(t1-t2)) [a2] [b2] ; [a2] [b2] amerge |
Make audio easier to listen to on headphones.
This filter adds `cues' to 44.1kHz stereo (i.e. audio CD format) audio so that when listened to on headphones the stereo image is moved from inside your head (standard for headphones) to outside and in front of the listener (standard for speakers).
Ported from SoX.
Mix channels with specific gain levels. The filter accepts the output channel layout followed by a set of channels definitions.
This filter is also designed to remap efficiently the channels of an audio stream.
The filter accepts parameters of the form: "l:outdef:outdef:..."
output channel layout or number of channels
output channel specification, of the form: "out_name=[gain*]in_name[+[gain*]in_name...]"
output channel to define, either a channel name (FL, FR, etc.) or a channel number (c0, c1, etc.)
multiplicative coefficient for the channel, 1 leaving the volume unchanged
input channel to use, see out_name for details; it is not possible to mix named and numbered input channels
If the `=' in a channel specification is replaced by `<', then the gains for that specification will be renormalized so that the total is 1, thus avoiding clipping noise.
For example, if you want to down-mix from stereo to mono, but with a bigger factor for the left channel:
pan=1:c0=0.9*c0+0.1*c1 |
A customized down-mix to stereo that works automatically for 3-, 4-, 5- and 7-channels surround:
pan=stereo: FL < FL + 0.5*FC + 0.6*BL + 0.6*SL : FR < FR + 0.5*FC + 0.6*BR + 0.6*SR |
Note that ffmpeg integrates a default down-mix (and up-mix) system
that should be preferred (see "-ac" option) unless you have very specific
needs.
The channel remapping will be effective if, and only if:
If all these conditions are satisfied, the filter will notify the user ("Pure channel mapping detected"), and use an optimized and lossless method to do the remapping.
For example, if you have a 5.1 source and want a stereo audio stream by dropping the extra channels:
pan="stereo: c0=FL : c1=FR" |
Given the same source, you can also switch front left and front right channels and keep the input channel layout:
pan="5.1: c0=c1 : c1=c0 : c2=c2 : c3=c3 : c4=c4 : c5=c5" |
If the input is a stereo audio stream, you can mute the front left channel (and still keep the stereo channel layout) with:
pan="stereo:c1=c1" |
Still with a stereo audio stream input, you can copy the right channel in both front left and right:
pan="stereo: c0=FR : c1=FR" |
Detect silence in an audio stream.
This filter logs a message when it detects that the input audio volume is less or equal to a noise tolerance value for a duration greater or equal to the minimum detected noise duration.
The printed times and duration are expressed in seconds.
Set silence duration until notification (default is 2 seconds).
Set noise tolerance. Can be specified in dB (in case "dB" is appended to the specified value) or amplitude ratio. Default is -60dB, or 0.001.
Detect 5 seconds of silence with -50dB noise tolerance:
silencedetect=n=-50dB:d=5 |
Complete example with ffmpeg to detect silence with 0.0001 noise
tolerance in `silence.mp3':
ffmpeg -f lavfi -i amovie=silence.mp3,silencedetect=noise=0.0001 -f null - |
Adjust the input audio volume.
The filter accepts exactly one parameter vol, which expresses how the audio volume will be increased or decreased.
Output values are clipped to the maximum value.
If vol is expressed as a decimal number, the output audio volume is given by the relation:
output_volume = vol * input_volume |
If vol is expressed as a decimal number followed by the string "dB", the value represents the requested change in decibels of the input audio power, and the output audio volume is given by the relation:
output_volume = 10^(vol/20) * input_volume |
Otherwise vol is considered an expression and its evaluated value is used for computing the output audio volume according to the first relation.
Default value for vol is 1.0.
volume=0.5 |
The above example is equivalent to:
volume=1/2 |
volume=-12dB |
Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed.
The filter accepts the following named parameters:
Enable stretching/squeezing the data to make it match the timestamps.
Minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples.
Maximum compensation in samples per second.
Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly.
Below is a description of the currently available audio sources.
Buffer audio frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in `libavfilter/asrc_abuffer.h'.
It accepts the following mandatory parameters: sample_rate:sample_fmt:channel_layout
The sample rate of the incoming audio buffers.
The sample format of the incoming audio buffers. Either a sample format name or its corresponging integer representation from the enum AVSampleFormat in `libavutil/samplefmt.h'
The channel layout of the incoming audio buffers. Either a channel layout name from channel_layout_map in `libavutil/audioconvert.c' or its corresponding integer representation from the AV_CH_LAYOUT_* macros in `libavutil/audioconvert.h'
For example:
abuffer=44100:s16p:stereo |
will instruct the source to accept planar 16bit signed stereo at 44100Hz. Since the sample format with name "s16p" corresponds to the number 6 and the "stereo" channel layout corresponds to the value 0x3, this is equivalent to:
abuffer=44100:6:0x3 |
Generate an audio signal specified by an expression.
This source accepts in input one or more expressions (one for each channel), which are evaluated and used to generate a corresponding audio signal.
It accepts the syntax: exprs[::options]. exprs is a list of expressions separated by ":", one for each separate channel. In case the channel_layout is not specified, the selected channel layout depends on the number of provided expressions.
options is an optional sequence of key=value pairs, separated by ":".
The description of the accepted options follows.
Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.
Set the minimum duration of the sourced audio. See the function
av_parse_time() for the accepted format.
Note that the resulting duration may be greater than the specified
duration, as the generated audio is always cut at the end of a
complete frame.
If not specified, or the expressed duration is negative, the audio is supposed to be generated forever.
Set the number of samples per channel per each output frame, default to 1024.
Specify the sample rate, default to 44100.
Each expression in exprs can contain the following constants:
number of the evaluated sample, starting from 0
time of the evaluated sample expressed in seconds, starting from 0
sample rate
aevalsrc=0 |
aevalsrc="sin(440*2*PI*t)::s=8000" |
aevalsrc="sin(420*2*PI*t):cos(430*2*PI*t)::c=FC|BC" |
aevalsrc="-2+random(0)" |
aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)" |
aevalsrc="0.1*sin(2*PI*(360-2.5/2)*t) : 0.1*sin(2*PI*(360+2.5/2)*t)" |
Read an audio stream from a movie container.
It accepts the syntax: movie_name[:options] where movie_name is the name of the resource to read (not necessarily a file but also a device or a stream accessed through some protocol), and options is an optional sequence of key=value pairs, separated by ":".
The description of the accepted options follows.
Specify the format assumed for the movie to read, and can be either the name of a container or an input device. If not specified the format is guessed from movie_name or by probing.
Specify the seek point in seconds, the frames will be output
starting from this seek point, the parameter is evaluated with
av_strtod so the numerical value may be suffixed by an IS
postfix. Default value is "0".
Specify the index of the audio stream to read. If the value is -1, the best suited audio stream will be automatically selected. Default value is "-1".
Null audio source, return unprocessed audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools, or as the source for filters which ignore the input data (for example the sox synth filter).
It accepts an optional sequence of key=value pairs, separated by ":".
The description of the accepted options follows.
Specify the sample rate, and defaults to 44100.
Specify the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is "stereo".
Check the channel_layout_map definition in `libavcodec/audioconvert.c' for the mapping between strings and channel layout values.
Set the number of samples per requested frames.
Follow some examples:
# set the sample rate to 48000 Hz and the channel layout to AV_CH_LAYOUT_MONO. anullsrc=r=48000:cl=4 # same as anullsrc=r=48000:cl=mono |
Buffer audio frames, and make them available to the filter chain.
This source is not intended to be part of user-supplied graph descriptions but for insertion by calling programs through the interface defined in `libavfilter/buffersrc.h'.
It accepts the following named parameters:
Timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
Audio sample rate.
Name of the sample format, as returned by av_get_sample_fmt_name().
Channel layout of the audio data, in the form that can be accepted by
av_get_channel_layout().
All the parameters need to be explicitly defined.
Below is a description of the currently available audio sinks.
Buffer audio frames, and make them available to the end of filter chain.
This sink is mainly intended for programmatic use, in particular through the interface defined in `libavfilter/buffersink.h'.
It requires a pointer to an AVABufferSinkContext structure, which
defines the incoming buffers' formats, to be passed as the opaque
parameter to avfilter_init_filter for initialization.
Null audio sink, do absolutely nothing with the input audio. It is mainly useful as a template and to be employed in analysis / debugging tools.
This sink is intended for programmatic use. Frames that arrive on this sink can be retrieved by the calling program using the interface defined in `libavfilter/buffersink.h'.
This filter accepts no parameters.
When you configure your FFmpeg build, you can disable any of the
existing filters using --disable-filters.
The configure output will show the video filters included in your
build.
Below is a description of the currently available video filters.
Draw ASS (Advanced Substation Alpha) subtitles on top of input video using the libass library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libass.
This filter accepts the syntax: ass_filename[:options], where ass_filename is the filename of the ASS file to read, and options is an optional sequence of key=value pairs, separated by ":".
A description of the accepted options follows.
Specifies the size of the original video, the video for which the ASS file was composed. Due to a misdesign in ASS aspect ratio arithmetic, this is necessary to correctly scale the fonts if the aspect ratio has been changed.
For example, to render the file `sub.ass' on top of the input video, use the command:
ass=sub.ass |
Compute the bounding box for the non-black pixels in the input frame luminance plane.
This filter computes the bounding box containing all the pixels with a luminance value greater than the minimum allowed value. The parameters describing the bounding box are printed on the filter log.
Detect video intervals that are (almost) completely black. Can be useful to detect chapter transitions, commercials, or invalid recordings. Output lines contains the time for the start, end and duration of the detected black interval expressed in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
This filter accepts a list of options in the form of key=value pairs separated by ":". A description of the accepted options follows.
Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number.
Default value is 2.0.
Set the threshold for considering a picture "black". Express the minimum value for the ratio:
nb_black_pixels / nb_pixels |
for which a picture is considered black. Default value is 0.98.
Set the threshold for considering a pixel "black".
The threshold expresses the maximum pixel luminance value for which a pixel is considered "black". The provided value is scaled according to the following equation:
absolute_threshold = luminance_minimum_value + pixel_black_th * luminance_range_size |
luminance_range_size and luminance_minimum_value depend on the input video format, the range is [0-255] for YUV full-range formats and [16-235] for YUV non full-range formats.
Default value is 0.10.
The following example sets the maximum pixel threshold to the minimum value, and detects only black intervals of 2 or more seconds:
blackdetect=d=2:pix_th=0.00 |
Detect frames that are (almost) completely black. Can be useful to detect chapter transitions or commercials. Output lines consist of the frame number of the detected frame, the percentage of blackness, the position in the file if known or -1 and the timestamp in seconds.
In order to display the output lines, you need to set the loglevel at least to the AV_LOG_INFO value.
The filter accepts the syntax:
blackframe[=amount:[threshold]] |
amount is the percentage of the pixels that have to be below the threshold, and defaults to 98.
threshold is the threshold below which a pixel value is considered black, and defaults to 32.
Apply boxblur algorithm to the input video.
This filter accepts the parameters: luma_radius:luma_power:chroma_radius:chroma_power:alpha_radius:alpha_power
Chroma and alpha parameters are optional, if not specified they default to the corresponding values set for luma_radius and luma_power.
luma_radius, chroma_radius, and alpha_radius represent the radius in pixels of the box used for blurring the corresponding input plane. They are expressions, and can contain the following constants:
the input width and height in pixels
the input chroma image width and height in pixels
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
The radius must be a non-negative number, and must not be greater than
the value of the expression min(w,h)/2 for the luma and alpha planes,
and of min(cw,ch)/2 for the chroma planes.
luma_power, chroma_power, and alpha_power represent how many times the boxblur filter is applied to the corresponding plane.
Some examples follow:
boxblur=2:1 |
boxblur=2:1:0:0:0:0 |
boxblur=min(h\,w)/10:1:min(cw\,ch)/10:1 |
The colormatrix filter allows conversion between any of the following color space: BT.709 (bt709), BT.601 (bt601), SMPTE-240M (smpte240m) and FCC (fcc).
The syntax of the parameters is source:destination:
colormatrix=bt601:smpte240m |
Copy the input source unchanged to the output. Mainly useful for testing purposes.
Crop the input video to out_w:out_h:x:y:keep_aspect
The keep_aspect parameter is optional, if specified and set to a non-zero value will force the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.
The out_w, out_h, x, y parameters are expressions containing the following constants:
the computed values for x and y. They are evaluated for each new frame.
the input width and height
same as in_w and in_h
the output (cropped) width and height
same as out_w and out_h
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the number of input frame, starting from 0
the position in the file of the input frame, NAN if unknown
timestamp expressed in seconds, NAN if the input timestamp is unknown
The out_w and out_h parameters specify the expressions for the width and height of the output (cropped) video. They are evaluated just at the configuration of the filter.
The default value of out_w is "in_w", and the default value of out_h is "in_h".
The expression for out_w may depend on the value of out_h, and the expression for out_h may depend on out_w, but they cannot depend on x and y, as x and y are evaluated after out_w and out_h.
The x and y parameters specify the expressions for the position of the top-left corner of the output (non-cropped) area. They are evaluated for each frame. If the evaluated value is not valid, it is approximated to the nearest valid value.
The default value of x is "(in_w-out_w)/2", and the default value for y is "(in_h-out_h)/2", which set the cropped area at the center of the input image.
The expression for x may depend on y, and the expression for y may depend on x.
Follow some examples:
# crop the central input area with size 100x100 crop=100:100 # crop the central input area with size 2/3 of the input video "crop=2/3*in_w:2/3*in_h" # crop the input video central square crop=in_h # delimit the rectangle with the top-left corner placed at position # 100:100 and the right-bottom corner corresponding to the right-bottom # corner of the input image. crop=in_w-100:in_h-100:100:100 # crop 10 pixels from the left and right borders, and 20 pixels from # the top and bottom borders "crop=in_w-2*10:in_h-2*20" # keep only the bottom right quarter of the input image "crop=in_w/2:in_h/2:in_w/2:in_h/2" # crop height for getting Greek harmony "crop=in_w:1/PHI*in_w" # trembling effect "crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(n/10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(n/7)" # erratic camera effect depending on timestamp "crop=in_w/2:in_h/2:(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):(in_h-out_h)/2 +((in_h-out_h)/2)*sin(t*13)" # set x depending on the value of y "crop=in_w/2:in_h/2:y:10+10*sin(n/10)" |
Auto-detect crop size.
Calculate necessary cropping parameters and prints the recommended parameters through the logging system. The detected dimensions correspond to the non-black area of the input video.
It accepts the syntax:
cropdetect[=limit[:round[:reset]]] |
Threshold, which can be optionally specified from nothing (0) to everything (255), defaults to 24.
Value which the width/height should be divisible by, defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs.
Counter that determines after how many frames cropdetect will reset the previously detected largest video area and start over to detect the current optimal crop area. Defaults to 0.
This can be useful when channel logos distort the video area. 0 indicates never reset and return the largest area encountered during playback.
Suppress a TV station logo by a simple interpolation of the surrounding pixels. Just set a rectangle covering the logo and watch it disappear (and sometimes something even uglier appear - your mileage may vary).
The filter accepts parameters as a string of the form "x:y:w:h:band", or as a list of key=value pairs, separated by ":".
The description of the accepted parameters follows.
Specify the top left corner coordinates of the logo. They must be specified.
Specify the width and height of the logo to clear. They must be specified.
Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 4.
When set to 1, a green rectangle is drawn on the screen to simplify finding the right x, y, w, h parameters, and band is set to 4. The default value is 0.
Some examples follow.
delogo=0:0:100:77:10 |
delogo=x=0:y=0:w=100:h=77:band=10 |
Attempt to fix small changes in horizontal and/or vertical shift. This filter helps remove camera shake from hand-holding a camera, bumping a tripod, moving on a vehicle, etc.
The filter accepts parameters as a string of the form "x:y:w:h:rx:ry:edge:blocksize:contrast:search:filename"
A description of the accepted parameters follows.
Specify a rectangular area where to limit the search for motion vectors. If desired the search for motion vectors can be limited to a rectangular area of the frame defined by its top left corner, width and height. These parameters have the same meaning as the drawbox filter which can be used to visualise the position of the bounding box.
This is useful when simultaneous movement of subjects within the frame might be confused for camera motion by the motion vector search.
If any or all of x, y, w and h are set to -1 then the full frame is used. This allows later options to be set without specifying the bounding box for the motion vector search.
Default - search the whole frame.
Specify the maximum extent of movement in x and y directions in the range 0-64 pixels. Default 16.
Specify how to generate pixels to fill blanks at the edge of the frame. An integer from 0 to 3 as follows:
Fill zeroes at blank locations
Original image at blank locations
Extruded edge value at blank locations
Mirrored edge at blank locations
The default setting is mirror edge at blank locations.
Specify the blocksize to use for motion search. Range 4-128 pixels, default 8.
Specify the contrast threshold for blocks. Only blocks with more than the specified contrast (difference between darkest and lightest pixels) will be considered. Range 1-255, default 125.
Specify the search strategy 0 = exhaustive search, 1 = less exhaustive search. Default - exhaustive search.
If set then a detailed log of the motion search is written to the specified file.
Draw a colored box on the input image.
It accepts the syntax:
drawbox=x:y:width:height:color |
Specify the top left corner coordinates of the box. Default to 0.
Specify the width and height of the box, if 0 they are interpreted as the input width and height. Default to 0.
Specify the color of the box to write, it can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence.
Follow some examples:
# draw a black box around the edge of the input image drawbox # draw a box with color red and an opacity of 50% drawbox=10:20:200:60:red@0.5" |
Draw text string or text from specified file on top of video using the libfreetype library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libfreetype.
The filter also recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime().
The filter accepts parameters as a list of key=value pairs, separated by ":".
The description of the accepted parameters follows.
Used to draw a box around text using background color. Value should be either 1 (enable) or 0 (disable). The default value of box is 0.
The color to be used for drawing box around text. Either a string (e.g. "yellow") or in 0xRRGGBB[AA] format (e.g. "0xff00ff"), possibly followed by an alpha specifier. The default value of boxcolor is "white".
Set an expression which specifies if the text should be drawn. If the expression evaluates to 0, the text is not drawn. This is useful for specifying that the text should be drawn only when specific conditions are met.
Default value is "1".
See below for the list of accepted constants and functions.
If true, check and fix text coords to avoid clipping.
The color to be used for drawing fonts. Either a string (e.g. "red") or in 0xRRGGBB[AA] format (e.g. "0xff000033"), possibly followed by an alpha specifier. The default value of fontcolor is "black".
The font file to be used for drawing text. Path must be included. This parameter is mandatory.
The font size to be used for drawing text. The default value of fontsize is 16.
Flags to be used for loading the fonts.
The flags map the corresponding flags supported by libfreetype, and are a combination of the following values:
Default value is "render".
For more information consult the documentation for the FT_LOAD_* libfreetype flags.
The color to be used for drawing a shadow behind the drawn text. It can be a color name (e.g. "yellow") or a string in the 0xRRGGBB[AA] form (e.g. "0xff00ff"), possibly followed by an alpha specifier. The default value of shadowcolor is "black".
The x and y offsets for the text shadow position with respect to the position of the text. They can be either positive or negative values. Default value for both is "0".
The size in number of spaces to use for rendering the tab. Default value is 4.
Set the initial timecode representation in "hh:mm:ss[:;.]ff" format. It can be used with or without text parameter. timecode_rate option must be specified.
Set the timecode frame rate (timecode only).
The text string to be drawn. The text must be a sequence of UTF-8 encoded characters. This parameter is mandatory if no file is specified with the parameter textfile.
A text file containing text to be drawn. The text must be a sequence of UTF-8 encoded characters.
This parameter is mandatory if no text string is specified with the parameter text.
If both text and textfile are specified, an error is thrown.
The expressions which specify the offsets where text will be drawn within the video frame. They are relative to the top/left border of the output image.
The default value of x and y is "0".
See below for the list of accepted constants and functions.
The parameters for x and y are expressions containing the following constants and functions:
input display aspect ratio, it is the same as (w / h) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
the height of each text line
the input height
the input width
the maximum distance from the baseline to the highest/upper grid coordinate used to place a glyph outline point, for all the rendered glyphs. It is a positive value, due to the grid's orientation with the Y axis upwards.
the maximum distance from the baseline to the lowest grid coordinate used to place a glyph outline point, for all the rendered glyphs. This is a negative value, due to the grid's orientation, with the Y axis upwards.
maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent - descent.
maximum glyph width, that is the maximum width for all the glyphs contained in the rendered text
the number of input frame, starting from 0
return a random number included between min and max
input sample aspect ratio
timestamp expressed in seconds, NAN if the input timestamp is unknown
the height of the rendered text
the width of the rendered text
the x and y offset coordinates where the text is drawn.
These parameters allow the x and y expressions to refer
each other, so you can for example specify y=x/dar.
If libavfilter was built with --enable-fontconfig, then
`fontfile' can be a fontconfig pattern or omitted.
Some examples follow.
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'" |
drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text':\
x=100: y=50: fontsize=24: fontcolor=yellow@0.2: box=1: boxcolor=red@0.2"
|
Note that the double quotes are not necessary if spaces are not used within the parameter list.
drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w-text_w)/2:y=(h-text_h-line_h)/2" |
drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h-line_h:x=-50*t" |
drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h-20*t" |
drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w-max_glyph_w)/2:y=h/2-ascent" |
drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:draw=lt(mod(t\\,3)\\,1):text='blink'" |
drawtext='fontfile=Linux Libertine O-40\\:style=Semibold:text=FFmpeg' |
For more information about libfreetype, check: http://www.freetype.org/.
For more information about fontconfig, check: http://freedesktop.org/software/fontconfig/fontconfig-user.html.
Apply fade-in/out effect to input video.
It accepts the parameters: type:start_frame:nb_frames[:options]
type specifies if the effect type, can be either "in" for fade-in, or "out" for a fade-out effect.
start_frame specifies the number of the start frame for starting to apply the fade effect.
nb_frames specifies the number of frames for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be completely black.
options is an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows.
See type.
See start_frame.
See nb_frames.
If set to 1, fade only alpha channel, if one exists on the input. Default value is 0.
A few usage examples follow, usable too as test scenarios.
# fade in first 30 frames of video fade=in:0:30 # fade out last 45 frames of a 200-frame video fade=out:155:45 # fade in first 25 frames and fade out last 25 frames of a 1000-frame video fade=in:0:25, fade=out:975:25 # make first 5 frames black, then fade in from frame 5-24 fade=in:5:20 # fade in alpha over first 25 frames of video fade=in:0:25:alpha=1 |
Transform the field order of the input video.
It accepts one parameter which specifies the required field order that the input interlaced video will be transformed to. The parameter can assume one of the following values:
output bottom field first
output top field first
Default value is "tff".
Transformation is achieved by shifting the picture content up or down by one line, and filling the remaining line with appropriate picture content. This method is consistent with most broadcast field order converters.
If the input video is not flagged as being interlaced, or it is already flagged as being of the required output field order then this filter does not alter the incoming video.
This filter is very useful when converting to or from PAL DV material, which is bottom field first.
For example:
ffmpeg -i in.vob -vf "fieldorder=bff" out.dv |
Buffer input images and send them when they are requested.
This filter is mainly useful when auto-inserted by the libavfilter framework.
The filter does not take parameters.
Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is supported for the input to the next filter.
The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24".
Some examples follow:
# convert the input video to the format "yuv420p" format=yuv420p # convert the input video to any of the formats in the list format=yuv420p:yuv444p:yuv410p |
Convert the video to specified constant framerate by duplicating or dropping frames as necessary.
This filter accepts the following named parameters:
Desired output framerate.
Apply a frei0r effect to the input video.
To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with --enable-frei0r.
The filter supports the syntax:
filter_name[{:|=}param1:param2:...:paramN]
|
filter_name is the name to the frei0r effect to load. If the
environment variable FREI0R_PATH is defined, the frei0r effect
is searched in each one of the directories specified by the colon
separated list in FREIOR_PATH, otherwise in the standard frei0r
paths, which are in this order: `HOME/.frei0r-1/lib/',
`/usr/local/lib/frei0r-1/', `/usr/lib/frei0r-1/'.
param1, param2, ... , paramN specify the parameters for the frei0r effect.
A frei0r effect parameter can be a boolean (whose values are specified
with "y" and "n"), a double, a color (specified by the syntax
R/G/B, R, G, and B being float
numbers from 0.0 to 1.0) or by an av_parse_color() color
description), a position (specified by the syntax X/Y,
X and Y being float numbers) and a string.
The number and kind of parameters depend on the loaded effect. If an effect parameter is not specified the default value is set.
Some examples follow:
# apply the distort0r effect, set the first two double parameters frei0r=distort0r:0.5:0.01 # apply the colordistance effect, takes a color as first parameter frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 # apply the perspective effect, specify the top left and top right # image positions frei0r=perspective:0.2/0.2:0.8/0.2 |
For more information see: http://piksel.org/frei0r
Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8bit color depth. Interpolate the gradients that should go where the bands are, and dither them.
This filter is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands.
The filter takes two optional parameters, separated by ':': strength:radius
strength is the maximum amount by which the filter will change any one pixel. Also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 255, default value is 1.2, out-of-range values will be clipped to the valid range.
radius is the neighborhood to fit the gradient to. A larger radius makes for smoother gradients, but also prevents the filter from modifying the pixels near detailed regions. Acceptable values are 8-32, default value is 16, out-of-range values will be clipped to the valid range.
# default parameters gradfun=1.2:16 # omitting radius gradfun=1.2 |
Flip the input video horizontally.
For example to horizontally flip the input video with ffmpeg:
ffmpeg -i in.avi -vf "hflip" out.avi |
High precision/quality 3d denoise filter. This filter aims to reduce image noise producing smooth images and making still images really still. It should enhance compressibility.
It accepts the following optional parameters: luma_spatial:chroma_spatial:luma_tmp:chroma_tmp
a non-negative float number which specifies spatial luma strength, defaults to 4.0
a non-negative float number which specifies spatial chroma strength, defaults to 3.0*luma_spatial/4.0
a float number which specifies luma temporal strength, defaults to 6.0*luma_spatial/4.0
a float number which specifies chroma temporal strength, defaults to luma_tmp*chroma_spatial/luma_spatial
Interlaceing detect filter. This filter tries to detect if the input is interlaced or progressive. Top or bottom field first.
Compute a look-up table for binding each pixel component input value to an output value, and apply it to input video.
lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video.
These filters accept in input a ":"-separated list of options, which specify the expressions used for computing the lookup table for the corresponding pixel component values.
The lut filter requires either YUV or RGB pixel formats in input, and accepts the options:
first pixel component
second pixel component
third pixel component
fourth pixel component, corresponds to the alpha component
The exact component associated to each option depends on the format in input.
The lutrgb filter requires RGB pixel formats in input, and accepts the options:
red component
green component
blue component
alpha component
The lutyuv filter requires YUV pixel formats in input, and accepts the options:
Y/luminance component
U/Cb component
V/Cr component
alpha component
The expressions can contain the following constants and functions:
the input width and height
input value for the pixel component
the input value clipped in the minval-maxval range
maximum value for the pixel component
minimum value for the pixel component
the negated value for the pixel component value clipped in the minval-maxval range , it corresponds to the expression "maxval-clipval+minval"
the computed value in val clipped in the minval-maxval range
the computed gamma correction value of the pixel component value clipped in the minval-maxval range, corresponds to the expression "pow((clipval-minval)/(maxval-minval)\,gamma)*(maxval-minval)+minval"
All expressions default to "val".
Some examples follow:
# negate input video lutrgb="r=maxval+minval-val:g=maxval+minval-val:b=maxval+minval-val" lutyuv="y=maxval+minval-val:u=maxval+minval-val:v=maxval+minval-val" # the above is the same as lutrgb="r=negval:g=negval:b=negval" lutyuv="y=negval:u=negval:v=negval" # negate luminance lutyuv=y=negval # remove chroma components, turns the video into a graytone image lutyuv="u=128:v=128" # apply a luma burning effect lutyuv="y=2*val" # remove green and blue components lutrgb="g=0:b=0" # set a constant alpha channel value on input format=rgba,lutrgb=a="maxval-minval/2" # correct luminance gamma by a 0.5 factor lutyuv=y=gammaval(0.5) |
Apply an MPlayer filter to the input video.
This filter provides a wrapper around most of the filters of MPlayer/MEncoder.
This wrapper is considered experimental. Some of the wrapped filters may not work properly and we may drop support for them, as they will be implemented natively into FFmpeg. Thus you should avoid depending on them when writing portable scripts.
The filters accepts the parameters: filter_name[:=]filter_params
filter_name is the name of a supported MPlayer filter, filter_params is a string containing the parameters accepted by the named filter.
The list of the currently supported filters follows:
The parameter syntax and behavior for the listed filters are the same of the corresponding MPlayer filters. For detailed instructions check the "VIDEO FILTERS" section in the MPlayer manual.
Some examples follow:
# adjust gamma, brightness, contrast mp=eq2=1.0:2:0.5 # tweak hue and saturation mp=hue=100:-10 |
See also mplayer(1), http://www.mplayerhq.hu/.
Negate input video.
This filter accepts an integer in input, if non-zero it negates the alpha component (if available). The default value in input is 0.
Force libavfilter not to use any of the specified pixel formats for the input to the next filter.
The filter accepts a list of pixel format names, separated by ":", for example "yuv420p:monow:rgb24".
Some examples follow:
# force libavfilter to use a format different from "yuv420p" for the # input to the vflip filter noformat=yuv420p,vflip # convert the input video to any of the formats not contained in the list noformat=yuv420p:yuv444p:yuv410p |
Pass the video source unchanged to the output.
Apply video transform using libopencv.
To enable this filter install libopencv library and headers and
configure FFmpeg with --enable-libopencv.
The filter takes the parameters: filter_name{:=}filter_params.
filter_name is the name of the libopencv filter to apply.
filter_params specifies the parameters to pass to the libopencv filter. If not specified the default values are assumed.
Refer to the official libopencv documentation for more precise information: http://opencv.willowgarage.com/documentation/c/image_filtering.html
Follows the list of supported libopencv filters.
Dilate an image by using a specific structuring element.
This filter corresponds to the libopencv function cvDilate.
It accepts the parameters: struct_el:nb_iterations.
struct_el represents a structuring element, and has the syntax: colsxrows+anchor_xxanchor_y/shape
cols and rows represent the number of columns and rows of the structuring element, anchor_x and anchor_y the anchor point, and shape the shape for the structuring element, and can be one of the values "rect", "cross", "ellipse", "custom".
If the value for shape is "custom", it must be followed by a string of the form "=filename". The file with name filename is assumed to represent a binary image, with each printable character corresponding to a bright pixel. When a custom shape is used, cols and rows are ignored, the number or columns and rows of the read file are assumed instead.
The default value for struct_el is "3x3+0x0/rect".
nb_iterations specifies the number of times the transform is applied to the image, and defaults to 1.
Follow some example:
# use the default values ocv=dilate # dilate using a structuring element with a 5x5 cross, iterate two times ocv=dilate=5x5+2x2/cross:2 # read the shape from the file diamond.shape, iterate two times # the file diamond.shape may contain a pattern of characters like this: # * # *** # ***** # *** # * # the specified cols and rows are ignored (but not the anchor point coordinates) ocv=0x0+2x2/custom=diamond.shape:2 |
Erode an image by using a specific structuring element.
This filter corresponds to the libopencv function cvErode.
The filter accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter.
Smooth the input video.
The filter takes the following parameters: type:param1:param2:param3:param4.
type is the type of smooth filter to apply, and can be one of the following values: "blur", "blur_no_scale", "median", "gaussian", "bilateral". The default value is "gaussian".
param1, param2, param3, and param4 are parameters whose meanings depend on smooth type. param1 and param2 accept integer positive values or 0, param3 and param4 accept float values.
The default value for param1 is 3, the default value for the other parameters is 0.
These parameters correspond to the parameters assigned to the
libopencv function cvSmooth.
Overlay one video on top of another.
It takes two inputs and one output, the first input is the "main" video on which the second input is overlayed.
It accepts the parameters: x:y[:options].
x is the x coordinate of the overlayed video on the main video, y is the y coordinate. x and y are expressions containing the following parameters:
main input width and height
same as main_w and main_h
overlay input width and height
same as overlay_w and overlay_h
options is an optional list of key=value pairs, separated by ":".
The description of the accepted options follows.
If set to 1, force the filter to accept inputs in the RGB color space. Default value is 0.
Be aware that frames are taken from each input video in timestamp order, hence, if their initial timestamps differ, it is a a good idea to pass the two inputs through a setpts=PTS-STARTPTS filter to have them begin in the same zero timestamp, as it does the example for the movie filter.
Follow some examples:
# draw the overlay at 10 pixels from the bottom right # corner of the main video. overlay=main_w-overlay_w-10:main_h-overlay_h-10 # insert a transparent PNG logo in the bottom left corner of the input ffmpeg -i input -i logo -filter_complex 'overlay=10:main_h-overlay_h-10' output # insert 2 different transparent PNG logos (second logo on bottom # right corner): ffmpeg -i input -i logo1 -i logo2 -filter_complex 'overlay=10:H-h-10,overlay=W-w-10:H-h-10' output # add a transparent color layer on top of the main video, # WxH specifies the size of the main input to the overlay filter color=red.3:WxH [over]; [in][over] overlay [out] |
You can chain together more overlays but the efficiency of such approach is yet to be tested.
Add paddings to the input image, and places the original input at the given coordinates x, y.
It accepts the following parameters: width:height:x:y:color.
The parameters width, height, x, and y are expressions containing the following constants:
the input video width and height
same as in_w and in_h
the output width and height, that is the size of the padded area as specified by the width and height expressions
same as out_w and out_h
x and y offsets as specified by the x and y expressions, or NAN if not yet specified
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
Follows the description of the accepted parameters.
Specify the size of the output image with the paddings added. If the value for width or height is 0, the corresponding input size is used for the output.
The width expression can reference the value set by the height expression, and vice versa.
The default value of width and height is 0.
Specify the offsets where to place the input image in the padded area with respect to the top/left border of the output image.
The x expression can reference the value set by the y expression, and vice versa.
The default value of x and y is 0.
Specify the color of the padded area, it can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence.
The default value of color is "black".
Some examples follow:
# Add paddings with color "violet" to the input video. Output video # size is 640x480, the top-left corner of the input video is placed at # column 0, row 40. pad=640:480:0:40:violet # pad the input to get an output with dimensions increased bt 3/2, # and put the input video at the center of the padded area pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" # pad the input to get a squared output with size equal to the maximum # value between the input width and height, and put the input video at # the center of the padded area pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" # pad the input to get a final w/h ratio of 16:9 pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" # for anamorphic video, in order to set the output display aspect ratio, # it is necessary to use sar in the expression, according to the relation: # (ih * X / ih) * sar = output_dar # X = output_dar / sar pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" # double output size and put the input video in the bottom-right # corner of the output padded area pad="2*iw:2*ih:ow-iw:oh-ih" |
Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video.
For example:
format=monow, pixdesctest |
can be used to test the monowhite pixel format descriptor definition.
Suppress a TV station logo, using an image file to determine which pixels comprise the logo. It works by filling in the pixels that comprise the logo with neighboring pixels.
This filter requires one argument which specifies the filter bitmap file, which can be any image format supported by libavformat. The width and height of the image file must match those of the video stream being processed.
Pixels in the provided bitmap image with a value of zero are not considered part of the logo, non-zero pixels are considered part of the logo. If you use white (255) for the logo and black (0) for the rest, you will be safe. For making the filter bitmap, it is recommended to take a screen capture of a black frame with the logo visible, and then using a threshold filter followed by the erode filter once or twice.
If needed, little splotches can be fixed manually. Remember that if logo pixels are not covered, the filter quality will be much reduced. Marking too many pixels as part of the logo does not hurt as much, but it will increase the amount of blurring needed to cover over the image and will destroy more information than necessary, and extra pixels will slow things down on a large logo.
Scale the input video to width:height[:interl={1|-1}] and/or convert the image format.
The scale filter forces the output display aspect ratio to be the same of the input, by changing the output sample aspect ratio.
The parameters width and height are expressions containing the following constants:
the input width and height
same as in_w and in_h
the output (cropped) width and height
same as out_w and out_h
same as iw / ih
input sample aspect ratio
input display aspect ratio, it is the same as (iw / ih) * sar
horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
If the input image format is different from the format requested by the next filter, the scale filter will convert the input to the requested format.
If the value for width or height is 0, the respective input size is used for the output.
If the value for width or height is -1, the scale filter will use, for the respective output size, a value that maintains the aspect ratio of the input image.
The default value of width and height is 0.
Valid values for the optional parameter interl are:
force interlaced aware scaling
select interlaced aware scaling depending on whether the source frames are flagged as interlaced or not
Unless interl is set to one of the above options, interlaced scaling will not be used.
Some examples follow:
# scale the input video to a size of 200x100. scale=200:100 # scale the input to 2x scale=2*iw:2*ih # the above is the same as scale=2*in_w:2*in_h # scale the input to 2x with forced interlaced scaling scale=2*iw:2*ih:interl=1 # scale the input to half size scale=iw/2:ih/2 # increase the width, and set the height to the same size scale=3/2*iw:ow # seek for Greek harmony scale=iw:1/PHI*iw scale=ih*PHI:ih # increase the height, and set the width to 3/2 of the height scale=3/2*oh:3/5*ih # increase the size, but make the size a multiple of the chroma scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" # increase the width to a maximum of 500 pixels, keep the same input aspect ratio scale='min(500\, iw*3/2):-1' |
Select frames to pass in output.
It accepts in input an expression, which is evaluated for each input frame. If the expression is evaluated to a non-zero value, the frame is selected and passed to the output, otherwise it is discarded.
The expression can contain the following constants:
the sequential number of the filtered frame, starting from 0
the sequential number of the selected frame, starting from 0
the sequential number of the last selected frame, NAN if undefined
timebase of the input timestamps
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units, NAN if undefined
the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
the PTS of the previously filtered video frame, NAN if undefined
the PTS of the last previously filtered video frame, NAN if undefined
the PTS of the last previously selected video frame, NAN if undefined
the PTS of the first video frame in the video, NAN if undefined
the time of the first video frame in the video, NAN if undefined
the type of the filtered frame, can assume one of the following values:
the frame interlace type, can assume one of the following values:
the frame is progressive (not interlaced)
the frame is top-field-first
the frame is bottom-field-first
1 if the filtered frame is a key-frame, 0 otherwise
the position in the file of the filtered frame, -1 if the information is not available (e.g. for synthetic video)
The default value of the select expression is "1".
Some examples follow:
# select all frames in input select # the above is the same as: select=1 # skip all frames: select=0 # select only I-frames select='eq(pict_type\,I)' # select one frame every 100 select='not(mod(n\,100))' # select only frames contained in the 10-20 time interval select='gte(t\,10)*lte(t\,20)' # select only I frames contained in the 10-20 time interval select='gte(t\,10)*lte(t\,20)*eq(pict_type\,I)' # select frames with a minimum distance of 10 seconds select='isnan(prev_selected_t)+gte(t-prev_selected_t\,10)' |
The setdar filter sets the Display Aspect Ratio for the filter
output video.
This is done by changing the specified Sample (aka Pixel) Aspect Ratio, according to the following equation:
DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR |
Keep in mind that the setdar filter does not modify the pixel
dimensions of the video frame. Also the display aspect ratio set by
this filter may be changed by later filters in the filterchain,
e.g. in case of scaling or if another "setdar" or a "setsar" filter is
applied.
The setsar filter sets the Sample (aka Pixel) Aspect Ratio for
the filter output video.
Note that as a consequence of the application of this filter, the output display aspect ratio will change according to the equation above.
Keep in mind that the sample aspect ratio set by the setsar
filter may be changed by later filters in the filterchain, e.g. if
another "setsar" or a "setdar" filter is applied.
The setdar and setsar filters accept a parameter string
which represents the wanted aspect ratio. The parameter can
be a floating point number string, an expression, or a string of the form
num:den, where num and den are the numerator
and denominator of the aspect ratio. If the parameter is not
specified, it is assumed the value "0:1".
For example to change the display aspect ratio to 16:9, specify:
setdar=16:9 |
The example above is equivalent to:
setdar=1.77777 |
To change the sample aspect ratio to 10:11, specify:
setsar=10:11 |
Force field for the output video frame.
The setfield filter marks the interlace type field for the
output frames. It does not change the input frame, but only sets the
corresponding property, which affects how the frame is treated by
following filters (e.g. fieldorder or yadif).
It accepts a string parameter, which can assume the following values:
Keep the same field property.
Mark the frame as bottom-field-first.
Mark the frame as top-field-first.
Mark the frame as progressive.
Change the PTS (presentation timestamp) of the input video frames.
Accept in input an expression evaluated through the eval API, which can contain the following constants:
the presentation timestamp in input
the count of the input frame, starting from 0.
the PTS of the first video frame
tell if the current frame is interlaced
original position in the file of the frame, or undefined if undefined for the current frame
previous input PTS
previous output PTS
Some examples follow:
# start counting PTS from zero setpts=PTS-STARTPTS # fast motion setpts=0.5*PTS # slow motion setpts=2.0*PTS # fixed rate 25 fps setpts=N/(25*TB) # fixed rate 25 fps with some jitter setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' |
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration.
It accepts in input an arithmetic expression representing a rational. The expression can contain the constants "AVTB" (the default timebase), and "intb" (the input timebase).
The default value for the input is "intb".
Follow some examples.
# set the timebase to 1/25 settb=1/25 # set the timebase to 1/10 settb=0.1 #set the timebase to 1001/1000 settb=1+0.001 #set the timebase to 2*intb settb=2*intb #set the default timebase value settb=AVTB |
Show a line containing various information for each input video frame. The input video is not modified.
The shown line contains a sequence of key/value pairs of the form key:value.
A description of each shown parameter follows:
sequential number of the input frame, starting from 0
Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.
Presentation TimeStamp of the input frame, expressed as a number of seconds
position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic video)
pixel format name
sample aspect ratio of the input frame, expressed in the form num/den
size of the input frame, expressed in the form widthxheight
interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first)
1 if the frame is a key frame, 0 otherwise
picture type of the input frame ("I" for an I-frame, "P" for a
P-frame, "B" for a B-frame, "?" for unknown type).
Check also the documentation of the AVPictureType enum and of
the av_get_picture_type_char function defined in
`libavutil/avutil.h'.
Adler-32 checksum (printed in hexadecimal) of all the planes of the input frame
Adler-32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]"
Pass the images of input video on to next video filter as multiple slices.
ffmpeg -i in.avi -vf "slicify=32" out.avi |
The filter accepts the slice height as parameter. If the parameter is not specified it will use the default value of 16.
Adding this in the beginning of filter chains should make filtering faster due to better use of the memory cache.
Split input video into several identical outputs.
The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2.
For example
ffmpeg -i INPUT -filter_complex split=5 OUTPUT |
will create 5 copies of the input video.
For example:
[in] split [splitout1][splitout2]; [splitout1] crop=100:100:0:0 [cropout]; [splitout2] pad=200:200:100:100 [padout]; |
will create two separate outputs from the same input, one cropped and one padded.
Scale the input by 2x and smooth using the Super2xSaI (Scale and Interpolate) pixel art scaling algorithm.
Useful for enlarging pixel art images without reducing sharpness.
Swap U & V plane.
Select the most representative frame in a given sequence of consecutive frames.
It accepts as argument the frames batch size to analyze (default N=100); in a set of N frames, the filter will pick one of them, and then handle the next batch of N frames until the end.
Since the filter keeps track of the whole frames sequence, a bigger N value will result in a higher memory usage, so a high value is not recommended.
The following example extract one picture each 50 frames:
thumbnail=50 |
Complete example of a thumbnail creation with ffmpeg:
ffmpeg -i in.avi -vf thumbnail,scale=300:200 -frames:v 1 out.png |
Tile several successive frames together.
It accepts as argument the tile size (i.e. the number of lines and columns) in the form "wxh".
For example, produce 8×8 PNG tiles of all keyframes (`-skip_frame nokey') in a movie:
ffmpeg -skip_frame nokey -i file.avi -vf 'scale=128:72,tile=8x8' -an -vsync 0 keyframes%03d.png |
The `-vsync 0' is necessary to prevent ffmpeg from
duplicating each output frame to accomodate the originally detected frame
rate.
Perform various types of temporal field interlacing.
Frames are counted starting from 1, so the first input frame is considered odd.
This filter accepts a single parameter specifying the mode. Available modes are:
Move odd frames into the upper field, even into the lower field, generating a double height frame at half framerate.
Only output even frames, odd frames are dropped, generating a frame with unchanged height at half framerate.
Only output odd frames, even frames are dropped, generating a frame with unchanged height at half framerate.
Expand each frame to full height, but pad alternate lines with black, generating a frame with double height at the same input framerate.
Interleave the upper field from odd frames with the lower field from even frames, generating a frame with unchanged height at half framerate.
Interleave the lower field from odd frames with the upper field from even frames, generating a frame with unchanged height at half framerate.
Double frame rate with unchanged height. Frames are inserted each containing the second temporal field from the previous input frame and the first temporal field from the next input frame. This mode relies on the top_field_first flag. Useful for interlaced video displays with no field synchronisation.
Numeric values are deprecated but are accepted for backward compatibility reasons.
Default mode is merge.
Transpose rows with columns in the input video and optionally flip it.
It accepts a parameter representing an integer, which can assume the values:
Rotate by 90 degrees counterclockwise and vertically flip (default), that is:
L.R L.l . . -> . . l.r R.r |
Rotate by 90 degrees clockwise, that is:
L.R l.L . . -> . . l.r r.R |
Rotate by 90 degrees counterclockwise, that is:
L.R R.r . . -> . . l.r L.l |
Rotate by 90 degrees clockwise and vertically flip, that is:
L.R r.R . . -> . . l.r l.L |
Sharpen or blur the input video.
It accepts the following parameters: luma_msize_x:luma_msize_y:luma_amount:chroma_msize_x:chroma_msize_y:chroma_amount
Negative values for the amount will blur the input video, while positive values will sharpen. All parameters are optional and default to the equivalent of the string '5:5:1.0:5:5:0.0'.
Set the luma matrix horizontal size. It can be an integer between 3 and 13, default value is 5.
Set the luma matrix vertical size. It can be an integer between 3 and 13, default value is 5.
Set the luma effect strength. It can be a float number between -2.0 and 5.0, default value is 1.0.
Set the chroma matrix horizontal size. It can be an integer between 3 and 13, default value is 5.
Set the chroma matrix vertical size. It can be an integer between 3 and 13, default value is 5.
Set the chroma effect strength. It can be a float number between -2.0 and 5.0, default value is 0.0.
# Strong luma sharpen effect parameters
unsharp=7:7:2.5
# Strong blur of both luma and chroma parameters
unsharp=7:7:-2:7:7:-2
# Use the default values with |
Flip the input video vertically.
ffmpeg -i in.avi -vf "vflip" out.avi |
Deinterlace the input video ("yadif" means "yet another deinterlacing filter").
It accepts the optional parameters: mode:parity:auto.
mode specifies the interlacing mode to adopt, accepts one of the following values:
output 1 frame for each frame
output 1 frame for each field
like 0 but skips spatial interlacing check
like 1 but skips spatial interlacing check
Default value is 0.
parity specifies the picture field parity assumed for the input interlaced video, accepts one of the following values:
assume top field first
assume bottom field first
enable automatic detection
Default value is -1. If interlacing is unknown or decoder does not export this information, top field first will be assumed.
auto specifies if deinterlacer should trust the interlaced flag and only deinterlace frames marked as interlaced
deinterlace all frames
only deinterlace frames marked as interlaced
Default value is 0.
Below is a description of the currently available video sources.
Buffer video frames, and make them available to the filter chain.
This source is mainly intended for a programmatic use, in particular through the interface defined in `libavfilter/vsrc_buffer.h'.
It accepts the following parameters: width:height:pix_fmt_string:timebase_num:timebase_den:sample_aspect_ratio_num:sample_aspect_ratio.den:scale_params
All the parameters but scale_params need to be explicitly defined.
Follows the list of the accepted parameters.
Specify the width and height of the buffered video frames.
A string representing the pixel format of the buffered video frames. It may be a number corresponding to a pixel format, or a pixel format name.
Specify numerator and denomitor of the timebase assumed by the timestamps of the buffered frames.
Specify numerator and denominator of the sample aspect ratio assumed by the video frames.
Specify the optional parameters to be used for the scale filter which is automatically inserted when an input change is detected in the input size or format.
For example:
buffer=320:240:yuv410p:1:24:1:1 |
will instruct the source to accept video frames with size 320x240 and with format "yuv410p", assuming 1/24 as the timestamps timebase and square pixels (1:1 sample aspect ratio). Since the pixel format with name "yuv410p" corresponds to the number 6 (check the enum PixelFormat definition in `libavutil/pixfmt.h'), this example corresponds to:
buffer=320:240:6:1:24:1:1 |
Create a pattern generated by an elementary cellular automaton.
The initial state of the cellular automaton can be defined through the `filename', and `pattern' options. If such options are not specified an initial state is created randomly.
At each new frame a new row in the video is filled with the result of the cellular automaton next generation. The behavior when the whole frame is filled is defined by the `scroll' option.
This source accepts a list of options in the form of key=value pairs separated by ":". A description of the accepted options follows.
Read the initial cellular automaton state, i.e. the starting row, from the specified file. In the file, each non-whitespace character is considered an alive cell, a newline will terminate the row, and further characters in the file will be ignored.
Read the initial cellular automaton state, i.e. the starting row, from the specified string.
Each non-whitespace character in the string is considered an alive cell, a newline will terminate the row, and further characters in the string will be ignored.
Set the video rate, that is the number of frames generated per second. Default is 25.
Set the random fill ratio for the initial cellular automaton row. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI.
This option is ignored when a file or a pattern is specified.
Set the seed for filling randomly the initial row, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.
Set the size of the output video.
If `filename' or `pattern' is specified, the size is set by default to the width of the specified initial state row, and the height is set to width * PHI.
If `size' is set, it must contain the width of the specified pattern string, and the specified pattern will be centered in the larger row.
If a filename or a pattern string is not specified, the size value defaults to "320x518" (used for a randomly generated initial state).
If set to 1, scroll the output upward when all the rows in the output have been already filled. If set to 0, the new generated row will be written over the top row just after the bottom row is filled. Defaults to 1.
If set to 1, completely fill the output with generated rows before outputting the first frame. This is the default behavior, for disabling set the value to 0.
If set to 1, stitch the left and right row edges together. This is the default behavior, for disabling set the value to 0.
cellauto=f=pattern:s=200x400 |
cellauto=ratio=2/3:s=200x200 |
cellauto=p=@:s=100x400:full=0:rule=18 |
cellauto=p='@@ @ @@':s=100x400:full=0:rule=18 |
Provide an uniformly colored input.
It accepts the following parameters: color:frame_size:frame_rate
Follows the description of the accepted parameters.
Specify the color of the source. It can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence, possibly followed by an alpha specifier. The default value is "black".
Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation. The default value is "320x240".
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
For example the following graph description will generate a red source with an opacity of 0.2, with size "qcif" and a frame rate of 10 frames per second, which will be overlayed over the source connected to the pad with identifier "in".
"color=red@0.2:qcif:10 [color]; [in][color] overlay [out]" |
Read a video stream from a movie container.
It accepts the syntax: movie_name[:options] where movie_name is the name of the resource to read (not necessarily a file but also a device or a stream accessed through some protocol), and options is an optional sequence of key=value pairs, separated by ":".
The description of the accepted options follows.
Specifies the format assumed for the movie to read, and can be either the name of a container or an input device. If not specified the format is guessed from movie_name or by probing.
Specifies the seek point in seconds, the frames will be output
starting from this seek point, the parameter is evaluated with
av_strtod so the numerical value may be suffixed by an IS
postfix. Default value is "0".
Specifies the index of the video stream to read. If the value is -1, the best suited video stream will be automatically selected. Default value is "-1".
Specifies how many times to read the video stream in sequence. If the value is less than 1, the stream will be read again and again. Default value is "1".
Note that when the movie is looped the source timestamps are not changed, so it will generate non monotonically increasing timestamps.
This filter allows to overlay a second video on top of main input of a filtergraph as shown in this graph:
input -----------> deltapts0 --> overlay --> output
^
|
movie --> scale--> deltapts1 -------+
|
Some examples follow:
# skip 3.2 seconds from the start of the avi file in.avi, and overlay it # on top of the input labelled as "in". movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] # read from a video4linux2 device, and overlay it on top of the input # labelled as "in" movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] |
Generate various test patterns, as generated by the MPlayer test filter.
The size of the generated video is fixed, and is 256x256. This source is useful in particular for testing encoding features.
This source accepts an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows.
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
Set the video duration of the sourced video. The accepted syntax is:
[-]HH:MM:SS[.m...] [-]S+[.m...] |
See also the function av_parse_time().
If not specified, or the expressed duration is negative, the video is supposed to be generated forever.
Set the number or the name of the test to perform. Supported tests are:
Default value is "all", which will cycle through the list of all tests.
For example the following:
testsrc=t=dc_luma |
will generate a "dc_luma" test pattern.
Provide a frei0r source.
To enable compilation of this filter you need to install the frei0r
header and configure FFmpeg with --enable-frei0r.
The source supports the syntax:
size:rate:src_name[{=|:}param1:param2:...:paramN]
|
size is the size of the video to generate, may be a string of the form widthxheight or a frame size abbreviation. rate is the rate of the video to generate, may be a string of the form num/den or a frame rate abbreviation. src_name is the name to the frei0r source to load. For more information regarding frei0r and how to set the parameters read the section frei0r in the description of the video filters.
Some examples follow:
# generate a frei0r partik0l source with size 200x200 and frame rate 10 # which is overlayed on the overlay filter main input frei0r_src=200x200:10:partik0l=1234 [overlay]; [in][overlay] overlay |
Generate a life pattern.
This source is based on a generalization of John Conway's life game.
The sourced input represents a life grid, each pixel represents a cell which can be in one of two possible states, alive or dead. Every cell interacts with its eight neighbours, which are the cells that are horizontally, vertically, or diagonally adjacent.
At each interaction the grid evolves according to the adopted rule, which specifies the number of neighbor alive cells which will make a cell stay alive or born. The `rule' option allows to specify the rule to adopt.
This source accepts a list of options in the form of key=value pairs separated by ":". A description of the accepted options follows.
Set the file from which to read the initial grid state. In the file, each non-whitespace character is considered an alive cell, and newline is used to delimit the end of each row.
If this option is not specified, the initial grid is generated randomly.
Set the video rate, that is the number of frames generated per second. Default is 25.
Set the random fill ratio for the initial random grid. It is a floating point number value ranging from 0 to 1, defaults to 1/PHI. It is ignored when a file is specified.
Set the seed for filling the initial random grid, must be an integer included between 0 and UINT32_MAX. If not specified, or if explicitly set to -1, the filter will try to use a good random seed on a best effort basis.
Set the life rule.
A rule can be specified with a code of the kind "SNS/BNB", where NS and NB are sequences of numbers in the range 0-8, NS specifies the number of alive neighbor cells which make a live cell stay alive, and NB the number of alive neighbor cells which make a dead cell to become alive (i.e. to "born"). "s" and "b" can be used in place of "S" and "B", respectively.
Alternatively a rule can be specified by an 18-bits integer. The 9
high order bits are used to encode the next cell state if it is alive
for each number of neighbor alive cells, the low order bits specify
the rule for "borning" new cells. Higher order bits encode for an
higher number of neighbor cells.
For example the number 6153 = (12<<9)+9 specifies a stay alive
rule of 12 and a born rule of 9, which corresponds to "S23/B03".
Default value is "S23/B3", which is the original Conway's game of life rule, and will keep a cell alive if it has 2 or 3 neighbor alive cells, and will born a new cell if there are three alive cells around a dead cell.
Set the size of the output video.
If `filename' is specified, the size is set by default to the same size of the input file. If `size' is set, it must contain the size specified in the input file, and the initial grid defined in that file is centered in the larger resulting area.
If a filename is not specified, the size value defaults to "320x240" (used for a randomly generated initial grid).
If set to 1, stitch the left and right grid edges together, and the top and bottom edges also. Defaults to 1.
Set cell mold speed. If set, a dead cell will go from `death_color' to `mold_color' with a step of `mold'. `mold' can have a value from 0 to 255.
Set the color of living (or new born) cells.
Set the color of dead cells. If `mold' is set, this is the first color used to represent a dead cell.
Set mold color, for definitely dead and moldy cells.
life=f=pattern:s=300x300 |
life=ratio=2/3:s=200x200 |
life=rule=S14/B34 |
ffplay:
ffplay -f lavfi life=s=300x200:mold=10:r=60:ratio=0.1:death_color=#C83232:life_color=#00ff00,scale=1200:800:flags=16 |
The nullsrc source returns unprocessed video frames. It is
mainly useful to be employed in analysis / debugging tools, or as the
source for filters which ignore the input data.
The rgbtestsrc source generates an RGB test pattern useful for
detecting RGB vs BGR issues. You should see a red, green and blue
stripe from top to bottom.
The testsrc source generates a test video pattern, showing a
color pattern, a scrolling gradient and a timestamp. This is mainly
intended for testing purposes.
These sources accept an optional sequence of key=value pairs, separated by ":". The description of the accepted options follows.
Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation. The default value is "320x240".
Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
Set the sample aspect ratio of the sourced video.
Set the video duration of the sourced video. The accepted syntax is:
[-]HH[:MM[:SS[.m...]]] [-]S+[.m...] |
See also the function av_parse_time().
If not specified, or the expressed duration is negative, the video is supposed to be generated forever.
Set the number of decimals to show in the timestamp, only used in the
testsrc source.
The displayed timestamp value will correspond to the original timestamp value multiplied by the power of 10 of the specified value. Default value is 0.
For example the following:
testsrc=duration=5.3:size=qcif:rate=10 |
will generate a video with a duration of 5.3 seconds, with size 176x144 and a frame rate of 10 frames per second.
If the input content is to be ignored, nullsrc can be used. The
following command generates noise in the luminance plane by employing
the mp=geq filter:
nullsrc=s=256x256, mp=geq=random(1)*255:128:128 |
Below is a description of the currently available video sinks.
Buffer video frames, and make them available to the end of the filter graph.
This sink is mainly intended for a programmatic use, in particular through the interface defined in `libavfilter/buffersink.h'.
It does not require a string parameter in input, but you need to
specify a pointer to a list of supported pixel formats terminated by
-1 in the opaque parameter provided to avfilter_init_filter
when initializing this sink.
Null video sink, do absolutely nothing with the input video. It is mainly useful as a template and to be employed in analysis / debugging tools.
FFmpeg is able to dump metadata from media files into a simple UTF-8-encoded INI-like text file and then load it back using the metadata muxer/demuxer.
The file format is as follows:
A ffmetadata file might look like this:
;FFMETADATA1 title=bike\\shed ;this is a comment artist=FFmpeg troll team [CHAPTER] TIMEBASE=1/1000 START=0 #chapter ends at 0:01:00 END=60000 title=chapter \#1 [STREAM] title=multi\ line |
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