ffplay [options] [‘input_file’] |
FFplay is a very simple and portable media player using the FFmpeg libraries and the SDL library. It is mostly used as a testbed for the various FFmpeg APIs.
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 specifier, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several streams, 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. An optional parameter may be specified to print help about a specific item.
Possible values of arg are:
Print detailed information about the decoder named decoder_name. Use the ‘-decoders’ option to get a list of all decoders.
Print detailed information about the encoder named encoder_name. Use the ‘-encoders’ option to get a list of all encoders.
Print detailed information about the demuxer named demuxer_name. Use the ‘-formats’ option to get a list of all demuxers and muxers.
Print detailed information about the muxer named muxer_name. Use the ‘-formats’ option to get a list of all muxers and demuxers.
Show version.
Show available formats.
The fields preceding the format names have the following meanings:
Decoding available
Encoding available
Show all codecs known to libavcodec.
Note that the term ’codec’ is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.
Show available decoders.
Show all available encoders.
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Show channel names and standard channel layouts.
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 should not 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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
5/3 wavelet, only used in snow
9/7 wavelet, only used in snow
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
8-bit unsigned integer planar
16-bit signed integer planar
32-bit signed integer planar
32-bit float planar
64-bit double planar
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
reduce the latency introduced by optional buffering
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
use wallclock as timestamps
avoid negative timestamps
Force displayed width.
Force displayed height.
Set frame size (WxH or abbreviation), needed for videos which do not contain a header with the frame size like raw YUV. This option has been deprecated in favor of private options, try -video_size.
Disable audio.
Disable video.
Seek to a given position in seconds.
play <duration> seconds of audio/video
Seek by bytes.
Disable graphical display.
Force format.
Set window title (default is the input filename).
Loops movie playback <number> times. 0 means forever.
Set the show mode to use. Available values for mode are:
show video
show audio waves
show audio frequency band using RDFT ((Inverse) Real Discrete Fourier Transform)
Default value is "video", if video is not present or cannot be played "rdft" is automatically selected.
You can interactively cycle through the available show modes by pressing the key <w>.
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).
Read input_file.
Set pixel format. This option has been deprecated in favor of private options, try -pixel_format.
Show the stream duration, the codec parameters, the current position in the stream and the audio/video synchronisation drift.
Work around bugs.
Non-spec-compliant optimizations.
Generate pts.
Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful if you are streaming with the RTSP protocol.
Set the master clock to audio (type=audio), video
(type=video) or external (type=ext). Default is audio. The
master clock is used to control audio-video synchronization. Most media
players use audio as master clock, but in some cases (streaming or high
quality broadcast) it is necessary to change that. This option is mainly
used for debugging purposes.
Set the thread count.
Select the desired audio stream number, counting from 0. The number refers to the list of all the input audio streams. If it is greater than the number of audio streams minus one, then the last one is selected, if it is negative the audio playback is disabled.
Select the desired video stream number, counting from 0. The number refers to the list of all the input video streams. If it is greater than the number of video streams minus one, then the last one is selected, if it is negative the video playback is disabled.
Select the desired subtitle stream number, counting from 0. The number refers to the list of all the input subtitle streams. If it is greater than the number of subtitle streams minus one, then the last one is selected, if it is negative the subtitle rendering is disabled.
Exit when video is done playing.
Exit if any key is pressed.
Exit if any mouse button is pressed.
Force a specific decoder implementation
Quit.
Toggle full screen.
Pause.
Cycle audio channel.
Cycle video channel.
Cycle subtitle channel.
Show audio waves.
Seek backward/forward 10 seconds.
Seek backward/forward 1 minute.
Seek backward/forward 10 minutes.
Seek to percentage in file corresponding to fraction of width.
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:
Compute hyperbolic sine of x.
Compute hyperbolic cosine of x.
Compute hyperbolic tangent of x.
Compute sine of x.
Compute cosine of x.
Compute tangent of x.
Compute arctangent of x.
Compute arcsine of x.
Compute arccosine of x.
Compute exponential of x (with base e, the Euler’s number).
Compute natural logarithm of x.
Compute absolute value of x.
Compute expression 1/(1 + exp(4*x)).
Compute Gauss function of x, corresponding to
exp(-x*x/2) / sqrt(2*PI).
Return 1.0 if x is +/-INFINITY, 0.0 otherwise.
Return 1.0 if x is NAN, 0.0 otherwise.
Compute the remainder of division of x by y.
Return the maximum between x and y.
Return the maximum between x and y.
Return 1 if x and y are equivalent, 0 otherwise.
Return 1 if x is greater than or equal to y, 0 otherwise.
Return 1 if x is greater than y, 0 otherwise.
Return 1 if x is lesser than or equal to y, 0 otherwise.
Return 1 if x is lesser than y, 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.
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 syntax and meaning of the pattern is specified by the option pattern_type.
The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files.
The size, the pixel format, and the format of each image must be the same for all the files in the sequence.
This demuxer accepts the following options:
Set the framerate for the video stream. It defaults to 25.
If set to 1, loop over the input. Default value is 0.
Select the pattern type used to interpret the provided filename.
pattern_type accepts one of the following values.
Select a sequence pattern type, used to specify a sequence of files indexed by sequential numbers.
A sequence 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 sequence pattern contains "%d" or "%0Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between start_number and start_number+start_number_range-1, and all the following numbers must be sequential.
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.
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 |
Select a glob wildcard pattern type.
The pattern is interpreted like a glob() pattern. This is only
selectable if libavformat was compiled with globbing support.
Select a mixed glob wildcard/sequence pattern.
If your version of libavformat was compiled with globbing support, and
the provided pattern contains at least one glob meta character among
%*?[]{} that is preceded by an unescaped "%", the pattern is
interpreted like a glob() pattern, otherwise it is interpreted
like a sequence pattern.
All glob special characters %*?[]{} must be prefixed
with "%". To escape a literal "%" you shall use "%%".
For example the pattern foo-%*.jpeg will match all the
filenames prefixed by "foo-" and terminating with ".jpeg", and
foo-%?%?%?.jpeg will match all the filenames prefixed with
"foo-", followed by a sequence of three characters, and terminating
with ".jpeg".
This pattern type is deprecated in favor of glob and sequence.
Default value is glob_sequence.
Set the pixel format of the images to read. If not specified the pixel format is guessed from the first image file in the sequence.
Set the index of the file matched by the image file pattern to start to read from. Default value is 0.
Set the index interval range to check when looking for the first image file in the sequence, starting from start_number. Default value is 5.
Set the video size of the images to read. If not specified the video size is guessed from the first image file in the sequence.
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 |
ffmpeg -start_number 100 -i 'img-%03d.jpeg' -r 10 out.mkv |
ffmpeg -pattern_type glob -i "*.png" -r 10 out.mkv |
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.
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.
ICO file muxer.
Microsoft’s icon file format (ICO) has some strict limitations that should be noted:
BMP Bit Depth FFmpeg Pixel Format 1bit pal8 4bit pal8 8bit pal8 16bit rgb555le 24bit bgr24 32bit bgra |
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.
stream_segment is a variant of the muxer used to write to
streaming output formats, i.e. which do not require global headers,
and is recommended for outputting e.g. to MPEG transport stream segments.
ssegment is a shorter alias for stream_segment.
Every segment starts with a video keyframe, if a video stream is present. Note that if you want accurate splitting for a video file, you need to make the input key frames correspond to the exact splitting times expected by the segmenter, or the segment muxer will start the new segment with the key frame found next after the specified start time.
The segment muxer works best with a single constant frame rate video.
Optionally it can generate a list of the created segments, by setting the option segment_list. The list type is specified by the segment_list_type option.
The segment muxer supports the following options:
Override the inner container format, by default it is guessed by the filename extension.
Generate also a listfile named name. If not specified no listfile is generated.
Set flags affecting the segment list generation.
It currently supports the following flags:
Allow caching (only affects M3U8 list files).
Allow live-friendly file generation.
This currently only affects M3U8 lists. In particular, write a fake EXT-X-TARGETDURATION duration field at the top of the file, based on the specified segment_time.
Default value is cache.
Overwrite the listfile once it reaches size entries. If 0 the listfile is never overwritten. Default value is 0.
Specify the format for the segment list file.
The following values are recognized:
Generate a flat list for the created segments, one segment per line.
Generate a list for the created segments, one segment per line, each line matching the format (comma-separated values):
segment_filename,segment_start_time,segment_end_time |
segment_filename is the name of the output file generated by the muxer according to the provided pattern. CSV escaping (according to RFC4180) is applied if required.
segment_start_time and segment_end_time specify the segment start and end time expressed in seconds.
A list file with the suffix ".csv" or ".ext" will
auto-select this format.
ext is deprecated in favor or csv.
Generate an extended M3U8 file, version 4, compliant with http://tools.ietf.org/id/draft-pantos-http-live-streaming-08.txt.
A list file with the suffix ".m3u8" will auto-select this format.
If not specified the type is guessed from the list file name suffix.
Set segment duration to time. Default value is "2".
Specify the accuracy time when selecting the start time for a segment. Default value is "0".
When delta is specified a key-frame will start a new segment if its PTS satisfies the relation:
PTS >= start_time - time_delta |
This option is useful when splitting video content, which is always split at GOP boundaries, in case a key frame is found just before the specified split time.
In particular may be used in combination with the ‘ffmpeg’ option force_key_frames. The key frame times specified by force_key_frames may not be set accurately because of rounding issues, with the consequence that a key frame time may result set just before the specified time. For constant frame rate videos a value of 1/2*frame_rate should address the worst case mismatch between the specified time and the time set by force_key_frames.
Specify a list of split points. times contains a list of comma separated duration specifications, in increasing order.
Wrap around segment index once it reaches limit.
Some examples follow.
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.list out%03d.nut |
ffmpeg -i in.mkv -codec copy -map 0 -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 out%03d.nut |
ffmpeg force_key_frames
option to force key frames in the input at the specified location, together
with the segment option segment_time_delta to account for
possible roundings operated when setting key frame times.
ffmpeg -i in.mkv -force_key_frames 1,2,3,5,8,13,21 -vcodec mpeg4 -acodec pcm_s16le -map 0 \ -f segment -segment_list out.csv -segment_times 1,2,3,5,8,13,21 -segment_time_delta 0.05 out%03d.nut |
In order to force key frames on the input file, transcoding is required.
libx264
and libfaac encoders:
ffmpeg -i in.mkv -map 0 -codec:v libx264 -codec:a libfaac -f ssegment -segment_list out.list out%03d.ts |
ffmpeg -re -i in.mkv -codec copy -map 0 -f segment -segment_list playlist.m3u8 \ -segment_list_flags +live -segment_time 10 out%03d.mkv |
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).
Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.
Set audio device buffer size in milliseconds (which can directly impact latency, depending on the device). Defaults to using the audio device’s default buffer size (typically some multiple of 500ms). Setting this value too low can degrade performance. See also http://msdn.microsoft.com/en-us/library/windows/desktop/dd377582(v=vs.85).aspx
$ 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).
FireWire DV/HDV input device using libiec61883.
To enable this input device, you need libiec61883, libraw1394 and
libavc1394 installed on your system. Use the configure option
--enable-libiec61883 to compile with the device enabled.
The iec61883 capture device supports capturing from a video device connected via IEEE1394 (FireWire), using libiec61883 and the new Linux FireWire stack (juju). This is the default DV/HDV input method in Linux Kernel 2.6.37 and later, since the old FireWire stack was removed.
Specify the FireWire port to be used as input file, or "auto" to choose the first port connected.
Override autodetection of DV/HDV. This should only be used if auto detection does not work, or if usage of a different device type should be prohibited. Treating a DV device as HDV (or vice versa) will not work and result in undefined behavior. The values ‘auto’, ‘dv’ and ‘hdv’ are supported.
Set maxiumum size of buffer for incoming data, in frames. For DV, this is an exact value. For HDV, it is not frame exact, since HDV does not have a fixed frame size.
ffplay -f iec61883 -i auto |
ffmpeg -f iec61883 -i auto -hdvbuffer 100000 out.mpg |
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 |
Specify whether to draw the mouse pointer. A value of 0 specify
not to draw the pointer. Default value is 1.
Make the grabbed area follow the mouse. The argument can be
centered or a number of pixels 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 |
To follow 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 |
Set the grabbing frame rate. Default value is ntsc,
corresponding to a framerate of 30000/1001.
Show grabbed region on screen.
If show_region is specified with 1, then the grabbing
region will be indicated on screen. With this option, it is 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 |
Set the video frame size. Default value is vga.
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.
CACA output device.
This output devices allows to show a video stream in CACA window. Only one CACA 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 to configure FFmpeg with
--enable-libcaca.
libcaca is a graphics library that outputs text instead of pixels.
For more information about libcaca, check: http://caca.zoy.org/wiki/libcaca
Set the CACA window title, if not specified default to the filename specified for the output device.
Set the CACA 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.
Set display driver.
Set dithering algorithm. Dithering is necessary
because the picture being rendered has usually far more colours than
the available palette.
The accepted values are listed with -list_dither algorithms.
Set antialias method. Antialiasing smoothens the rendered
image and avoids the commonly seen staircase effect.
The accepted values are listed with -list_dither antialiases.
Set which characters are going to be used when rendering text.
The accepted values are listed with -list_dither charsets.
Set color to be used when rendering text.
The accepted values are listed with -list_dither colors.
If set to ‘true’, print a list of available drivers and exit.
List available dither options related to the argument.
The argument must be one of algorithms, antialiases,
charsets, colors.
ffmpeg output is an
CACA window, forcing its size to 80x25:
ffmpeg -i INPUT -vcodec rawvideo -pix_fmt rgb24 -window_size 80x25 -f caca - |
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_drivers true - |
ffmpeg -i INPUT -pix_fmt rgb24 -f caca -list_dither colors - |
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, downscaled according to the aspect ratio.
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][/instance][/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.). You can override
the value parsed from the URI through the rtmp_app option, too.
It is the path or name of the resource to play with reference to the
application specified in app, may be prefixed by "mp4:". You
can override the value parsed from the URI through the rtmp_playpath
option, too.
Act as a server, listening for an incoming connection.
Maximum time to wait for the incoming connection. Implies listen.
Additionally, the following parameters can be set via command line options
(or in code via AVOptions):
Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.
Set the client buffer time in milliseconds. The default is 3000.
Extra arbitrary AMF connection parameters, parsed from a string,
e.g. like B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0.
Each value is prefixed by a single character denoting the type,
B for Boolean, N for number, S for string, O for object, or Z for null,
followed by a colon. For Booleans the data must be either 0 or 1 for
FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
1 to end or begin an object, respectively. Data items in subobjects may
be named, by prefixing the type with ’N’ and specifying the name before
the value (i.e. NB:myFlag:1). This option may be used multiple
times to construct arbitrary AMF sequences.
Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.
Number of packets flushed in the same request (RTMPT only). The default is 10.
Specify that the media is a live stream. No resuming or seeking in
live streams is possible. The default value is any, which means the
subscriber first tries to play the live stream specified in the
playpath. If a live stream of that name is not found, it plays the
recorded stream. The other possible values are live and
recorded.
URL of the web page in which the media was embedded. By default no value will be sent.
Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.
SHA256 hash of the decompressed SWF file (32 bytes).
Size of the decompressed SWF file, required for SWFVerification.
URL of the SWF player for the media. By default no value will be sent.
URL to player swf file, compute hash/size automatically.
URL of the target stream. Defaults to proto://host[:port]/app.
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 |
Encrypted Real-Time Messaging Protocol.
The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys.
Real-Time Messaging Protocol over a secure SSL connection.
The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection.
Real-Time Messaging Protocol tunneled through HTTP.
The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls.
Encrypted Real-Time Messaging Protocol tunneled through HTTP.
The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls.
Real-Time Messaging Protocol tunneled through HTTPS.
The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls.
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.
Act as a server, listening for an incoming connection.
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 |
To receive a stream in realtime:
ffmpeg -rtsp_flags listen -i rtsp://ownaddress/live.sdp output |
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
In read mode: if no data arrived in more than this time interval, raise error. In write mode: if socket cannot be written in more than this time interval, raise error. This also sets timeout on TCP connection establishing.
ffmpeg -i input -f format tcp://hostname:port?listen ffplay tcp://hostname:port |
Transport Layer Security/Secure Sockets Layer
The required syntax for a TLS/SSL url is:
tls://hostname:port[?options] |
Act as a server, listening for an incoming connection.
Certificate authority file. The file must be in OpenSSL PEM format.
Certificate file. The file must be in OpenSSL PEM format.
Private key file.
Verify the peer’s certificate.
Example command lines:
To create a TLS/SSL server that serves an input stream.
ffmpeg -i input -f format tls://hostname:port?listen&cert=server.crt&key=server.key |
To play back a stream from the TLS/SSL server using ffplay:
ffplay tls://hostname:port |
User Datagram Protocol.
The required syntax for a UDP url is:
udp://hostname:port[?options] |
options contains a list of &-separated options of the form key=val.
In case threading is enabled on the system, a circular buffer is used to store the incoming data, which allows to reduce loss of data due to UDP socket buffer overruns. The fifo_size and overrun_nonfatal options are related to this buffer.
The list of supported options follows.
Set the UDP socket buffer size in bytes. This is used both for the receiving and the sending buffer size.
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.
Only receive packets sent to the multicast group from one of the specified sender IP addresses.
Ignore packets sent to the multicast group from the specified sender IP addresses.
Set the UDP receiving circular buffer size, expressed as a number of packets with size of 188 bytes. If not specified defaults to 7*4096.
Survive in case of UDP receiving circular buffer overrun. Default value is 0.
In read mode: if no data arrived in more than this time interval, raise error.
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 |
Filtering in FFmpeg is enabled through the libavfilter library.
Libavfilter is the filtering API of FFmpeg. It is the substitute of the now deprecated ’vhooks’ and started as a Google Summer of Code project.
Audio filtering integration into the main FFmpeg repository is a work in progress, so audio API and ABI should not be considered stable yet.
In libavfilter, it is possible for filters to have multiple inputs and multiple outputs. To illustrate the sorts of things that are possible, we can use a complex filter graph. For example, the following one:
input --> split --> fifo -----------------------> overlay --> output
| ^
| |
+------> fifo --> crop --> vflip --------+
|
splits the stream in two streams, sends one stream through the crop filter and the vflip filter before merging it back with the other stream by overlaying it on top. You can use the following command to achieve this:
ffmpeg -i input -vf "[in] split [T1], fifo, [T2] overlay=0:H/2 [out]; [T1] fifo, crop=iw:ih/2:0:ih/2, vflip [T2]" output |
The result will be that in output the top half of the video is mirrored onto the bottom half.
Filters are loaded using the -vf or -af option passed to
ffmpeg or to ffplay. Filters in the same linear
chain are separated by commas. In our example, split, fifo,
overlay are in one linear chain, and fifo, crop, vflip are in
another. The points where the linear chains join are labeled by names
enclosed in square brackets. In our example, that is [T1] and
[T2]. The special labels [in] and [out] are the points
where video is input and output.
Some filters take in input a list of parameters: they are specified after the filter name and an equal sign, and are separated from each other by a colon.
There exist so-called source filters that do not have an audio/video input, and sink filters that will not have audio/video output.
The ‘graph2dot’ program included in the FFmpeg ‘tools’ directory can be used to parse a filter graph description and issue a corresponding textual representation in the dot language.
Invoke the command:
graph2dot -h |
to see how to use ‘graph2dot’.
You can then pass the dot description to the ‘dot’ program (from the graphviz suite of programs) and obtain a graphical representation of the filter graph.
For example the sequence of commands:
echo GRAPH_DESCRIPTION | \ tools/graph2dot -o graph.tmp && \ dot -Tpng graph.tmp -o graph.png && \ display graph.png |
can be used to create and display an image representing the graph described by the GRAPH_DESCRIPTION string. Note that this string must be a complete self-contained graph, with its inputs and outputs explicitly defined. For example if your command line is of the form:
ffmpeg -i infile -vf scale=640:360 outfile |
your GRAPH_DESCRIPTION string will need to be of the form:
nullsrc,scale=640:360,nullsink |
you may also need to set the nullsrc parameters and add a format filter in order to simulate a specific input file.
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 or more audio streams into a single multi-channel stream.
The filter accepts the following named options:
Set the number of inputs. Default is 2.
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.
All 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 |
Example: multiple merges:
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][a2][a3][a4][a5] amerge=inputs=6" -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 |
Set the number of samples per each output audio frame.
The last output packet may contain a different number of samples, as the filter will flush all the remaining samples when the input audio signal its end.
The filter accepts parameters as a list of key=value pairs, separated by ":".
Set the number of frames per each output audio frame. The number is intended as the number of samples per each channel. Default value is 1024.
If set to 1, the filter will pad the last audio frame with zeroes, so that the last frame will contain the same number of samples as the previous ones. Default value is 1.
For example, to set the number of per-frame samples to 1234 and disable padding for the last frame, use:
asetnsamples=n=1234:p=0 |
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 |
Adjust audio tempo.
The filter accepts exactly one parameter, the audio tempo. If not specified then the filter will assume nominal 1.0 tempo. Tempo must be in the [0.5, 2.0] range.
For example, to slow down audio to 80% tempo:
atempo=0.8 |
For example, to speed up audio to 125% tempo:
atempo=1.25 |
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 |
Detect the volume of the input video.
The filter has no parameters. The input is not modified. Statistics about the volume will be printed in the log when the input stream end is reached.
In particular it will show the mean volume (root mean square), maximum volume (on a per-sample basis), and the beginning of an histogram of the registered volume values (from the maximum value to a cumulated 1/1000 of the samples).
All volumes are in decibels relative to the maximum PCM value.
Here is an excerpt of the output:
[Parsed_volumedetect_0 0xa23120] mean_volume: -27 dB [Parsed_volumedetect_0 0xa23120] max_volume: -4 dB [Parsed_volumedetect_0 0xa23120] histogram_4db: 6 [Parsed_volumedetect_0 0xa23120] histogram_5db: 62 [Parsed_volumedetect_0 0xa23120] histogram_6db: 286 [Parsed_volumedetect_0 0xa23120] histogram_7db: 1042 [Parsed_volumedetect_0 0xa23120] histogram_8db: 2551 [Parsed_volumedetect_0 0xa23120] histogram_9db: 4609 [Parsed_volumedetect_0 0xa23120] histogram_10db: 8409 |
It means that:
In other words, raising the volume by +4 dB does not cause any clipping, raising it by +5 dB causes clipping for 6 samples, etc.
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. Disabled by default. When disabled, time gaps are covered with silence.
Minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. Default value is 0.1. If you get non-perfect sync with this filter, try setting this parameter to 0.
Maximum compensation in samples per second. Relevant only with compensate=1. Default value 500.
Assume the first pts should be this value. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame’s expected pts, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with silence if an audio stream starts after the video stream.
Split each channel in input audio stream into a separate output stream.
This filter accepts the following named parameters:
Channel layout of the input stream. Default is "stereo".
For example, assuming a stereo input MP3 file
ffmpeg -i in.mp3 -filter_complex channelsplit out.mkv |
will create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel.
To split a 5.1 WAV file into per-channel files
ffmpeg -i in.wav -filter_complex 'channelsplit=channel_layout=5.1[FL][FR][FC][LFE][SL][SR]' -map '[FL]' front_left.wav -map '[FR]' front_right.wav -map '[FC]' front_center.wav -map '[LFE]' lfe.wav -map '[SL]' side_left.wav -map '[SR]' side_right.wav |
Remap input channels to new locations.
This filter accepts the following named parameters:
Channel layout of the output stream.
Map channels from input to output. The argument is a comma-separated list of
mappings, each in the in_channel-out_channel or
in_channel form. in_channel can be either the name of the input
channel (e.g. FL for front left) or its index in the input channel layout.
out_channel is the name of the output channel or its index in the output
channel layout. If out_channel is not given then it is implicitly an
index, starting with zero and increasing by one for each mapping.
If no mapping is present, the filter will implicitly map input channels to output channels preserving index.
For example, assuming a 5.1+downmix input MOV file
ffmpeg -i in.mov -filter 'channelmap=map=DL-FL\,DR-FR' out.wav |
will create an output WAV file tagged as stereo from the downmix channels of the input.
To fix a 5.1 WAV improperly encoded in AAC’s native channel order
ffmpeg -i in.wav -filter 'channelmap=1\,2\,0\,5\,3\,4:channel_layout=5.1' out.wav |
Join multiple input streams into one multi-channel stream.
The filter accepts the following named parameters:
Number of input streams. Defaults to 2.
Desired output channel layout. Defaults to stereo.
Map channels from inputs to output. The argument is a comma-separated list of
mappings, each in the input_idx.in_channel-out_channel
form. input_idx is the 0-based index of the input stream. in_channel
can be either the name of the input channel (e.g. FL for front left) or its
index in the specified input stream. out_channel is the name of the output
channel.
The filter will attempt to guess the mappings when those are not specified explicitly. It does so by first trying to find an unused matching input channel and if that fails it picks the first unused input channel.
E.g. to join 3 inputs (with properly set channel layouts)
ffmpeg -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT |
To build a 5.1 output from 6 single-channel streams:
ffmpeg -i fl -i fr -i fc -i sl -i sr -i lfe -filter_complex 'join=inputs=6:channel_layout=5.1:map=0.0-FL\,1.0-FR\,2.0-FC\,3.0-SL\,4.0-SR\,5.0-LFE' out |
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)" |
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.
Synthesize a voice utterance using the libflite library.
To enable compilation of this filter you need to configure FFmpeg with
--enable-libflite.
Note that the flite library is not thread-safe.
The source accepts parameters as a list of key=value pairs, separated by ":".
The description of the accepted parameters follows.
If set to 1, list the names of the available voices and exit immediately. Default value is 0.
Set the maximum number of samples per frame. Default value is 512.
Set the filename containing the text to speak.
Set the text to speak.
Set the voice to use for the speech synthesis. Default value is
kal. See also the list_voices option.
flite=textfile=speech.txt |
slt voice:
flite=text='So fare thee well, poor devil of a Sub-Sub, whose commentator I am':voice=slt |
flite and
the lavfi device:
ffplay -f lavfi flite='No more be grieved for which that thou hast done.' |
For more information about libflite, check: http://www.speech.cs.cmu.edu/flite/
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.
Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.
Add or replace the alpha component of the primary input with the grayscale value of a second input. This is intended for use with alphaextract to allow the transmission or storage of frame sequences that have alpha in a format that doesn’t support an alpha channel.
For example, to reconstruct full frames from a normal YUV-encoded video and a separate video created with alphaextract, you might use:
movie=in_alpha.mkv [alpha]; [in][alpha] alphamerge [out] |
Since this filter is designed for reconstruction, it operates on frame sequences without considering timestamps, and terminates when either input reaches end of stream. This will cause problems if your encoding pipeline drops frames. If you’re trying to apply an image as an overlay to a video stream, consider the overlay filter instead.
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.
This filter drops frames that do not differ greatly from the previous frame in order to reduce framerate. The main use of this filter is for very-low-bitrate encoding (e.g. streaming over dialup modem), but it could in theory be used for fixing movies that were inverse-telecined incorrectly.
It accepts the following parameters: max:hi:lo:frac.
Set the maximum number of consecutive frames which can be dropped (if positive), or the minimum interval between dropped frames (if negative). If the value is 0, the frame is dropped unregarding the number of previous sequentially dropped frames.
Default value is 0.
Set the dropping threshold values.
Values for hi and lo are for 8x8 pixel blocks and represent actual pixel value differences, so a threshold of 64 corresponds to 1 unit of difference for each pixel, or the same spread out differently over the block.
A frame is a candidate for dropping if no 8x8 blocks differ by more than a threshold of hi, and if no more than frac blocks (1 meaning the whole image) differ by more than a threshold of lo.
Default value for hi is 64*12, default value for lo is 64*5, and default value for frac is 0.33.
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.
Detect and draw edges. The filter uses the Canny Edge Detection algorithm.
This filter accepts the following optional named parameters:
Set low and high threshold values used by the Canny thresholding algorithm.
The high threshold selects the "strong" edge pixels, which are then connected through 8-connectivity with the "weak" edge pixels selected by the low threshold.
low and high threshold values must be choosen in the range [0,1], and low should be lesser or equal to high.
Default value for low is 20/255, and default value for high
is 50/255.
Example:
edgedetect=low=0.1:high=0.4 |
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.
Select one frame every N.
This filter accepts in input a string representing a positive
integer. Default argument is 1.
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:
frei0r=distort0r:0.5:0.01 |
frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 |
frei0r=perspective:0.2/0.2:0.8/0.2 |
For more information see: http://frei0r.dyne.org
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
Modify the hue and/or the saturation of the input.
This filter accepts the following optional named options:
Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0.
Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0.
Specify the saturation in the [-10,10] range. It accepts a float number and defaults to 1.0.
The h, H and s parameters are expressions containing the following constants:
frame count of the input frame starting from 0
presentation timestamp of the input frame expressed in time base units
frame rate of the input video, NAN if the input frame rate is unknown
timestamp expressed in seconds, NAN if the input timestamp is unknown
time base of the input video
The options can also be set using the syntax: hue:saturation
In this case hue is expressed in degrees.
Some examples follow:
hue=h=90:s=1 |
hue=H=PI/2:s=1 |
hue=90:1 |
hue=PI/2:1 |
hue="H=2*PI*t: s=sin(2*PI*t)+1" |
hue="s=min(t/3\,1)" |
The general fade-in expression can be written as:
hue="s=min(0\, max((t-START)/DURATION\, 1))" |
hue="s=max(0\, min(1\, (8-t)/3))" |
The general fade-out expression can be written as:
hue="s=max(0\, min(1\, (START+DURATION-t)/DURATION))" |
This filter supports the following command:
Modify the hue and/or the saturation of the input video. The command accepts the same named options and syntax than when calling the filter from the command-line.
If a parameter is omitted, it is kept at its current value.
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:
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:
The lutyuv filter requires YUV pixel formats in input, and accepts the options:
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:
mp=eq2=1.0:2:0.5 |
mp=noise=20t |
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] # play an original video and a filtered version (here with the deshake filter) # side by side ffplay input.avi -vf 'split[a][b]; [a]pad=iw*2:ih[src]; [b]deshake[filt]; [src][filt]overlay=w' # the previous example is the same as: ffplay input.avi -vf 'split[b], pad=iw*2[src], [b]deshake, [src]overlay=w' |
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".
pad=640:480:0:40:violet |
pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" |
pad="max(iw\,ih):ow:(ow-iw)/2:(oh-ih)/2" |
pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" |
(ih * X / ih) * sar = output_dar X = output_dar / sar |
Thus the previous example needs to be modified to:
pad="ih*16/9/sar:ih:(ow-iw)/2:(oh-ih)/2" |
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)
value between 0 and 1 to indicate a new scene; a low value reflects a low probability for the current frame to introduce a new scene, while a higher value means the current frame is more likely to be one (see the example below)
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)' |
Complete example to create a mosaic of the first scenes:
ffmpeg -i video.avi -vf select='gt(scene\,0.4)',scale=160:120,tile -frames:v 1 preview.png |
Comparing scene against a value between 0.3 and 0.5 is generally a sane choice.
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.
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.
Blur the input video without impacting the outlines.
The filter accepts the following parameters: luma_radius:luma_strength:luma_threshold[:chroma_radius:chroma_strength:chroma_threshold]
Parameters prefixed by luma indicate that they work on the luminance of the pixels whereas parameters prefixed by chroma refer to the chrominance of the pixels.
If the chroma parameters are not set, the luma parameters are used for either the luminance and the chrominance of the pixels.
luma_radius or chroma_radius must be a float number in the range [0.1,5.0] that specifies the variance of the gaussian filter used to blur the image (slower if larger).
luma_strength or chroma_strength must be a float number in the range [-1.0,1.0] that configures the blurring. A value included in [0.0,1.0] will blur the image whereas a value included in [-1.0,0.0] will sharpen the image.
luma_threshold or chroma_threshold must be an integer in the range [-30,30] that is used as a coefficient to determine whether a pixel should be blurred or not. A value of 0 will filter all the image, a value included in [0,30] will filter flat areas and a value included in [-30,0] will filter edges.
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.
This filter accepts the following named parameters:
Specify the transposition direction. Can assume the following 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 |
For values between 4-7, the transposition is only done if the input
video geometry is portrait and not landscape. These values are
deprecated, the passthrough option should be used instead.
Do not apply the transposition if the input geometry matches the one specified by the specified value. It accepts the following values:
Always apply transposition.
Preserve portrait geometry (when height >= width).
Preserve landscape geometry (when width >= height).
Default value is none.
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 a list of options in the form of key=value pairs separated by ":". A description of the accepted options follows.
Specify the size (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 the timebase assumed by the timestamps of the buffered frames.
Specify the frame rate expected for the video stream.
Specify 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=size=320x240:pix_fmt=yuv410p:time_base=1/24:pixel_aspect=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 AVPixelFormat definition in ‘libavutil/pixfmt.h’), this example corresponds to:
buffer=size=320x240:pixfmt=6:time_base=1/24:pixel_aspect=1/1 |
Alternatively, the options can be specified as a flat string, but this syntax is deprecated:
width:height:pix_fmt:time_base.num:time_base.den:pixel_aspect.num:pixel_aspect.den[:sws_param]
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 |
Generate a Mandelbrot set fractal, and progressively zoom towards the point specified with start_x and start_y.
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 terminal pts value. Default value is 400.
Set the terminal scale value. Must be a floating point value. Default value is 0.3.
Set the inner coloring mode, that is the algorithm used to draw the Mandelbrot fractal internal region.
It shall assume one of the following values:
Set black mode.
Show time until convergence.
Set color based on point closest to the origin of the iterations.
Set period mode.
Default value is mincol.
Set the bailout value. Default value is 10.0.
Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.
Set outer coloring mode. It shall assume one of following values:
Set iteration cound mode.
set normalized iteration count mode.
Default value is normalized_iteration_count.
Set frame rate, expressed as number of frames per second. Default value is "25".
Set frame size. Default value is "640x480".
Set the initial scale value. Default value is 3.0.
Set the initial x position. Must be a floating point value between -100 and 100. Default value is -0.743643887037158704752191506114774.
Set the initial y position. Must be a floating point value between -100 and 100. Default value is -0.131825904205311970493132056385139.
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.
For example, to 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 color source provides an uniformly colored input.
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 smptebars source generates a color bars pattern, based on
the SMPTE Engineering Guideline EG 1-1990.
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 color of the source, only used in the color
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".
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.
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.
color=c=red@0.2:s=qcif:r=10 |
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.
Below is a description of the currently available multimedia filters.
Send commands to filters in the filtergraph.
These filters read commands to be sent to other filters in the filtergraph.
asendcmd must be inserted between two audio filters,
sendcmd must be inserted between two video filters, but apart
from that they act the same way.
The specification of commands can be provided in the filter arguments with the commands option, or in a file specified by the filename option.
These filters accept the following options:
Set the commands to be read and sent to the other filters.
Set the filename of the commands to be read and sent to the other filters.
A commands description consists of a sequence of interval specifications, comprising a list of commands to be executed when a particular event related to that interval occurs. The occurring event is typically the current frame time entering or leaving a given time interval.
An interval is specified by the following syntax:
START[-END] COMMANDS; |
The time interval is specified by the START and END times. END is optional and defaults to the maximum time.
The current frame time is considered within the specified interval if it is included in the interval [START, END), that is when the time is greater or equal to START and is lesser than END.
COMMANDS consists of a sequence of one or more command specifications, separated by ",", relating to that interval. The syntax of a command specification is given by:
[FLAGS] TARGET COMMAND ARG |
FLAGS is optional and specifies the type of events relating to the time interval which enable sending the specified command, and must be a non-null sequence of identifier flags separated by "+" or "|" and enclosed between "[" and "]".
The following flags are recognized:
The command is sent when the current frame timestamp enters the specified interval. In other words, the command is sent when the previous frame timestamp was not in the given interval, and the current is.
The command is sent when the current frame timestamp leaves the specified interval. In other words, the command is sent when the previous frame timestamp was in the given interval, and the current is not.
If FLAGS is not specified, a default value of [enter] is
assumed.
TARGET specifies the target of the command, usually the name of the filter class or a specific filter instance name.
COMMAND specifies the name of the command for the target filter.
ARG is optional and specifies the optional list of argument for the given COMMAND.
Between one interval specification and another, whitespaces, or
sequences of characters starting with # until the end of line,
are ignored and can be used to annotate comments.
A simplified BNF description of the commands specification syntax follows:
COMMAND_FLAG ::= "enter" | "leave" COMMAND_FLAGS ::= COMMAND_FLAG [(+|"|")COMMAND_FLAG] COMMAND ::= ["[" COMMAND_FLAGS "]"] TARGET COMMAND [ARG] COMMANDS ::= COMMAND [,COMMANDS] INTERVAL ::= START[-END] COMMANDS INTERVALS ::= INTERVAL[;INTERVALS] |
asendcmd=c='4.0 atempo tempo 1.5',atempo |
# show text in the interval 5-10
5.0-10.0 [enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=hello world',
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=';
# desaturate the image in the interval 15-20
15.0-20.0 [enter] hue reinit s=0,
[enter] drawtext reinit 'fontfile=FreeSerif.ttf:text=nocolor',
[leave] hue reinit s=1,
[leave] drawtext reinit 'fontfile=FreeSerif.ttf:text=color';
# apply an exponential saturation fade-out effect, starting from time 25
25 [enter] hue s=exp(t-25)
|
A filtergraph allowing to read and process the above command list stored in a file ‘test.cmd’, can be specified with:
sendcmd=f=test.cmd,drawtext=fontfile=FreeSerif.ttf:text='',hue |
Change the PTS (presentation timestamp) of the input frames.
asetpts works on audio frames, setpts on video frames.
Accept in input an expression evaluated through the eval API, which can contain the following constants:
frame rate, only defined for constant frame-rate video
the presentation timestamp in input
the count of the input frame, starting from 0.
the number of consumed samples, not including the current frame (only audio)
the number of samples in the current frame (only audio)
audio sample rate
the PTS of the first frame
the time in seconds of the first frame
tell if the current frame is interlaced
the time in seconds of the current frame
the time base
original position in the file of the frame, or undefined if undefined for the current frame
previous input PTS
previous input time in seconds
previous output PTS
previous output time in seconds
setpts=PTS-STARTPTS |
setpts=0.5*PTS |
setpts=2.0*PTS |
setpts=N/(25*TB) |
setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))' |
setpts=PTS+10/TB |
EBU R128 scanner filter. This filter takes an audio stream as input and outputs
it unchanged. By default, it logs a message at a frequency of 10Hz with the
Momentary loudness (identified by M), Short-term loudness (S),
Integrated loudness (I) and Loudness Range (LRA).
The filter also has a video output (see the video option) with a real time graph to observe the loudness evolution. The graphic contains the logged message mentioned above, so it is not printed anymore when this option is set, unless the verbose logging is set. The main graphing area contains the short-term loudness (3 seconds of analysis), and the gauge on the right is for the momentary loudness (400 milliseconds).
More information about the Loudness Recommendation EBU R128 on http://tech.ebu.ch/loudness.
The filter accepts the following named parameters:
Activate the video output. The audio stream is passed unchanged whether this
option is set or no. The video stream will be the first output stream if
activated. Default is 0.
Set the video size. This option is for video only. Default and minimum
resolution is 640x480.
Set the EBU scale meter. Default is 9. Common values are 9 and
18, respectively for EBU scale meter +9 and EBU scale meter +18. Any
other integer value between this range is allowed.
Example of real-time graph using ffplay, with a EBU scale meter +18:
ffplay -f lavfi -i "amovie=input.mp3,ebur128=video=1:meter=18 [out0][out1]" |
Run an analysis with ffmpeg:
ffmpeg -nostats -i input.mp3 -filter_complex ebur128 -f null - |
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), "intb" (the input timebase) and "sr" (the sample rate, audio only).
The default value for the input is "intb".
settb=1/25 |
settb=0.1 |
settb=1+0.001 |
settb=2*intb |
settb=AVTB |
Concatenate audio and video streams, joining them together one after the other.
The filter works on segments of synchronized video and audio streams. All segments must have the same number of streams of each type, and that will also be the number of streams at output.
The filter accepts the following named parameters:
Set the number of segments. Default is 2.
Set the number of output video streams, that is also the number of video streams in each segment. Default is 1.
Set the number of output audio streams, that is also the number of video streams in each segment. Default is 0.
The filter has v+a outputs: first v video outputs, then a audio outputs.
There are n×(v+a) inputs: first the inputs for the first segment, in the same order as the outputs, then the inputs for the second segment, etc.
Related streams do not always have exactly the same duration, for various reasons including codec frame size or sloppy authoring. For that reason, related synchronized streams (e.g. a video and its audio track) should be concatenated at once. The concat filter will use the duration of the longest stream in each segment (except the last one), and if necessary pad shorter audio streams with silence.
For this filter to work correctly, all segments must start at timestamp 0.
All corresponding streams must have the same parameters in all segments; the filtering system will automatically select a common pixel format for video streams, and a common sample format, sample rate and channel layout for audio streams, but other settings, such as resolution, must be converted explicitly by the user.
Different frame rates are acceptable but will result in variable frame rate at output; be sure to configure the output file to handle it.
Examples:
ffmpeg -i opening.mkv -i episode.mkv -i ending.mkv -filter_complex \ '[0:0] [0:1] [0:2] [1:0] [1:1] [1:2] [2:0] [2:1] [2:2] concat=n=3:v=1:a=2 [v] [a1] [a2]' \ -map '[v]' -map '[a1]' -map '[a2]' output.mkv |
movie=part1.mp4, scale=512:288 [v1] ; amovie=part1.mp4 [a1] ; movie=part2.mp4, scale=512:288 [v2] ; amovie=part2.mp4 [a2] ; [v1] [v2] concat [outv] ; [a1] [a2] concat=v=0:a=1 [outa] |
Note that a desync will happen at the stitch if the audio and video streams do not have exactly the same duration in the first file.
Convert input audio to a video output, representing the audio frequency spectrum.
The filter accepts the following named parameters:
Specify the video size for the output. Default value is 640x480.
The usage is very similar to the showwaves filter; see the examples in that section.
Convert input audio to a video output, representing the samples waves.
The filter accepts the following named parameters:
Set the number of samples which are printed on the same column. A larger value will decrease the frame rate. Must be a positive integer. This option can be set only if the value for rate is not explicitly specified.
Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".
Specify the video size for the output. Default value is "600x240".
Some examples follow.
amovie=a.mp3,asplit[out0],showwaves[out1] |
aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1] |
Below is a description of the currently available multimedia sources.
This is the same as src_movie source, except it selects an audio stream by default.
Read audio and/or video stream(s) 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 streams to read. Several streams can be specified, separated by "+". The source will then have as many outputs, in the same order. The syntax is explained in the Stream specifiers chapter. Two special names, "dv" and "da" specify respectively the default (best suited) video and audio stream. Default is "dv", or "da" if the filter is called as "amovie".
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". Deprecated. If the filter is called "amovie", it will select audio instead of video.
Specifies how many times to read the 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.
movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] |
movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] |
movie=dvd.vob:s=v:0+#0x81 [video] [audio] |