Debian 7.3.0

GNU/Linux	Debian 7.3.0 (Wheezy)
	ffmpeg(1)

FFMPEG

NAME
SYNOPSIS
DESCRIPTION
DETAILED DESCRIPTION
STREAM SELECTION
OPTIONS
TIPS
EXAMPLES
SYNTAX
EXPRESSION EVALUATION
DECODERS
VIDEO DECODERS
AUDIO DECODERS
ENCODERS
AUDIO ENCODERS
VIDEO ENCODERS
DEMUXERS
MUXERS
INPUT DEVICES
OUTPUT DEVICES
PROTOCOLS
BITSTREAM FILTERS
FILTERING INTRODUCTION
GRAPH
FILTERGRAPH DESCRIPTION
AUDIO FILTERS
AUDIO SOURCES
AUDIO SINKS
VIDEO FILTERS
VIDEO SOURCES
VIDEO SINKS
MULTIMEDIA FILTERS
MULTIMEDIA SOURCES
METADATA

NAME

ffmpeg − ffmpeg video converter

SYNOPSIS

The generic syntax is:

        ffmpeg [global options] [[infile options][B<−i> <infile>]]... {[outfile options] <outfile>}...

DESCRIPTION

ffmpeg is a very fast video and audio converter that can also grab from a live audio/video source. It can also convert between arbitrary sample rates and resize video on the fly with a high quality polyphase filter.

ffmpeg reads from an arbitrary number of input "files" (which can be regular files, pipes, network streams, grabbing devices, etc.), specified by the "−i" option, and writes to an arbitrary number of output "files", which are specified by a plain output filename. Anything found on the command line which cannot be interpreted as an option is considered to be an output filename.

Each input or output file can in principle contain any number of streams of different types (video/audio/subtitle/attachment/data). Allowed number and/or types of streams can be limited by the container format. Selecting, which streams from which inputs go into output, is done either automatically or with the "−map" option (see the Stream selection chapter).

To refer to input files in options, you must use their indices (0−based). E.g. the first input file is 0, the second is 1 etc. Similarly, streams within a file are referred to by their indices. E.g. "2:3" refers to the fourth stream in the third input file. See also the Stream specifiers chapter.

As a general rule, options are applied to the next specified file. Therefore, order is important, and you can have the same option on the command line multiple times. Each occurrence is then applied to the next input or output file. Exceptions from this rule are the global options (e.g. verbosity level), which should be specified first.

Do not mix input and output files -- first specify all input files, then all output files. Also do not mix options which belong to different files. All options apply ONLY to the next input or output file and are reset between files.

•

To set the video bitrate of the output file to 64kbit/s:

        ffmpeg −i input.avi −b:v 64k output.avi

•

To force the frame rate of the output file to 24 fps:

        ffmpeg −i input.avi −r 24 output.avi

•

To force the frame rate of the input file (valid for raw formats only) to 1 fps and the frame rate of the output file to 24 fps:

        ffmpeg −r 1 −i input.m2v −r 24 output.avi

The format option may be needed for raw input files.

DETAILED DESCRIPTION

The transcoding process in ffmpeg for each output can be described by the following diagram:

         _______              ______________               _________              ______________            ________
        |       |            |              |             |         |            |              |          |        |
        | input |  demuxer   | encoded data |   decoder   | decoded |  encoder   | encoded data |  muxer   | output |
        | file  | −−−−−−−−−> | packets      |  −−−−−−−−−> | frames  | −−−−−−−−−> | packets      | −−−−−−−> | file   |
        |_______|            |______________|             |_________|            |______________|          |________|

ffmpeg calls the libavformat library (containing demuxers) to read input files and get packets containing encoded data from them. When there are multiple input files, ffmpeg tries to keep them synchronized by tracking lowest timestamp on any active input stream.

Encoded packets are then passed to the decoder (unless streamcopy is selected for the stream, see further for a description). The decoder produces uncompressed frames (raw video/PCM audio/...) which can be processed further by filtering (see next section). After filtering the frames are passed to the encoder, which encodes them and outputs encoded packets again. Finally those are passed to the muxer, which writes the encoded packets to the output file.

Filtering
Before encoding, ffmpeg can process raw audio and video frames using filters from the libavfilter library. Several chained filters form a filter graph. ffmpeg distinguishes between two types of filtergraphs − simple and complex.

Simple filtergraphs

Simple filtergraphs are those that have exactly one input and output, both of the same type. In the above diagram they can be represented by simply inserting an additional step between decoding and encoding:

         _________                        __________              ______________
        |         |                      |          |            |              |
        | decoded |  simple filtergraph  | filtered |  encoder   | encoded data |
        | frames  | −−−−−−−−−−−−−−−−−−−> | frames   | −−−−−−−−−> | packets      |
        |_________|                      |__________|            |______________|

Simple filtergraphs are configured with the per-stream −filter option (with −vf and −af aliases for video and audio respectively). A simple filtergraph for video can look for example like this:

         _______        _____________        _______        _____        ________
        |       |      |             |      |       |      |     |      |        |
        | input | −−−> | deinterlace | −−−> | scale | −−−> | fps | −−−> | output |
        |_______|      |_____________|      |_______|      |_____|      |________|

Note that some filters change frame properties but not frame contents. E.g. the "fps" filter in the example above changes number of frames, but does not touch the frame contents. Another example is the "setpts" filter, which only sets timestamps and otherwise passes the frames unchanged.

Complex filtergraphs

Complex filtergraphs are those which cannot be described as simply a linear processing chain applied to one stream. This is the case e.g. when the graph has more than one input and/or output, or when output stream type is different from input. They can be represented with the following diagram:

         _________
        |         |
        | input 0 |\                    __________
        |_________| \                  |          |
                     \   _________    /| output 0 |
                      \ |         |  / |__________|
         _________     \| complex | /
        |         |     |         |/
        | input 1 |−−−−>| filter  |\
        |_________|     |         | \   __________
                       /| graph   |  \ |          |
                      / |         |   \| output 1 |
         _________   /  |_________|    |__________|
        |         | /
        | input 2 |/
        |_________|

Complex filtergraphs are configured with the −filter_complex option. Note that this option is global, since a complex filtergraph by its nature cannot be unambiguously associated with a single stream or file.

A trivial example of a complex filtergraph is the "overlay" filter, which has two video inputs and one video output, containing one video overlaid on top of the other. Its audio counterpart is the "amix" filter.

Stream copy
Stream copy is a mode selected by supplying the "copy" parameter to the −codec option. It makes ffmpeg omit the decoding and encoding step for the specified stream, so it does only demuxing and muxing. It is useful for changing the container format or modifying container-level metadata. The diagram above will in this case simplify to this:

         _______              ______________            ________
        |       |            |              |          |        |
        | input |  demuxer   | encoded data |  muxer   | output |
        | file  | −−−−−−−−−> | packets      | −−−−−−−> | file   |
        |_______|            |______________|          |________|

Since there is no decoding or encoding, it is very fast and there is no quality loss. However it might not work in some cases because of many factors. Applying filters is obviously also impossible, since filters work on uncompressed data.

STREAM SELECTION

By default ffmpeg includes only one stream of each type (video, audio, subtitle) present in the input files and adds them to each output file. It picks the "best" of each based upon the following criteria; for video it is the stream with the highest resolution, for audio the stream with the most channels, for subtitle it’s the first subtitle stream. In the case where several streams of the same type rate equally, the lowest numbered stream is chosen.

You can disable some of those defaults by using "−vn/−an/−sn" options. For full manual control, use the "−map" option, which disables the defaults just described.

OPTIONS

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".

Stream specifiers
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:
stream_index

Matches the stream with this index. E.g. "−threads:1 4" would set the thread count for the second stream to 4.

stream_type[:stream_index]

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.

p:program_id[:stream_index]

If stream_index is given, then matches stream number stream_index in program with id program_id. Otherwise matches all streams in this program.

#stream_id

Matches the stream by format-specific ID .

Generic options
These options are shared amongst the av* tools.

−L

Show license.

−h, −?, −help, −−help [arg]

Show help. An optional parameter may be specified to print help about a specific item.

Possible values of arg are:
decoder=decoder_name

Print detailed information about the decoder named decoder_name. Use the −decoders option to get a list of all decoders.

encoder=encoder_name

Print detailed information about the encoder named encoder_name. Use the −encoders option to get a list of all encoders.

demuxer=demuxer_name

Print detailed information about the demuxer named demuxer_name. Use the −formats option to get a list of all demuxers and muxers.

muxer=muxer_name

Print detailed information about the muxer named muxer_name. Use the −formats option to get a list of all muxers and demuxers.

−version

Show version.

−formats

Show available formats.

The fields preceding the format names have the following meanings:

	D		Decoding available
	E		Encoding available

−codecs

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.

−decoders

Show available decoders.

−encoders

Show all available encoders.

−bsfs

Show available bitstream filters.

−protocols

Show available protocols.

−filters

Show available libavfilter filters.

−pix_fmts

Show available pixel formats.

−sample_fmts

Show available sample formats.

−layouts

Show channel names and standard channel layouts.

−loglevel loglevel | −v loglevel

Set the logging level used by the library. loglevel is a number or a string containing one of the following values:
quiet
panic
fatal
error
warning
info
verbose
debug

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.

−report

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.

−cpuflags flags (global)

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 ...

AVOptions
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:
generic

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.

private

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.

Codec AVOptions
−b[:stream_specifier] integer (output,audio,video)

set bitrate (in bits/s)

−ab[:stream_specifier] integer (output,audio)

set bitrate (in bits/s)

−bt[:stream_specifier] integer (output,video)

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.

−flags[:stream_specifier] flags (input/output,audio,video,subtitles)

Possible values:

mv4

use four motion vector by macroblock (mpeg4)

qpel

use 1/4 pel motion compensation

loop

use loop filter

qscale

use fixed qscale

	gmc		use gmc
	mv0		always try a mb with mv=<0,0>

input_preserved
pass1

use internal 2pass ratecontrol in first pass mode

pass2

use internal 2pass ratecontrol in second pass mode

gray

only decode/encode grayscale

emu_edge

don’t draw edges

psnr

error[?] variables will be set during encoding

truncated

naq

normalize adaptive quantization

ildct

use interlaced dct

low_delay

force low delay

global_header

place global headers in extradata instead of every keyframe

bitexact

use only bitexact stuff (except (i)dct)

	aic		h263 advanced intra coding / mpeg4 ac prediction
	cbp		Deprecated, use mpegvideo private options instead

qprd

Deprecated, use mpegvideo private options instead

ilme

interlaced motion estimation

cgop

closed gop

−sub_id[:stream_specifier] integer ()
−me_method[:stream_specifier] integer (output,video)

set motion estimation method

Possible values:
zero

zero motion estimation (fastest)

full

full motion estimation (slowest)

epzs

EPZS motion estimation (default)

esa

esa motion estimation (alias for full)

tesa

tesa motion estimation

	dia		dia motion estimation (alias for epzs)
	log		log motion estimation

phods

phods motion estimation

	x1		X1 motion estimation
	hex		hex motion estimation
	umh		umh motion estimation

iter

iter motion estimation

−extradata_size[:stream_specifier] integer ()
−time_base[:stream_specifier] rational number ()
−g[:stream_specifier] integer (output,video)

set the group of picture size

−ar[:stream_specifier] integer (input/output,audio)

set audio sampling rate (in Hz)

−ac[:stream_specifier] integer (input/output,audio)

set number of audio channels

−cutoff[:stream_specifier] integer (output,audio)

set cutoff bandwidth

−frame_size[:stream_specifier] integer (output,audio)
−frame_number[:stream_specifier] integer ()
−delay[:stream_specifier] integer ()
−qcomp[:stream_specifier] float (output,video)

video quantizer scale compression ( VBR ). Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0−1.0

−qblur[:stream_specifier] float (output,video)

video quantizer scale blur ( VBR )

−qmin[:stream_specifier] integer (output,video)

min video quantizer scale ( VBR )

−qmax[:stream_specifier] integer (output,video)

max video quantizer scale ( VBR )

−qdiff[:stream_specifier] integer (output,video)

max difference between the quantizer scale ( VBR )

−bf[:stream_specifier] integer (output,video)

use ’frames’ B frames

−b_qfactor[:stream_specifier] float (output,video)

qp factor between p and b frames

−rc_strategy[:stream_specifier] integer (output,video)

ratecontrol method

−b_strategy[:stream_specifier] integer (output,video)

strategy to choose between I/P/B−frames

−ps[:stream_specifier] integer (output,video)

rtp payload size in bytes

−mv_bits[:stream_specifier] integer ()
−header_bits[:stream_specifier] integer ()
−i_tex_bits[:stream_specifier] integer ()
−p_tex_bits[:stream_specifier] integer ()
−i_count[:stream_specifier] integer ()
−p_count[:stream_specifier] integer ()
−skip_count[:stream_specifier] integer ()
−misc_bits[:stream_specifier] integer ()
−frame_bits[:stream_specifier] integer ()
−codec_tag[:stream_specifier] integer ()
−bug[:stream_specifier] flags (input,video)

workaround not auto detected encoder bugs

Possible values:
autodetect
old_msmpeg4

some old lavc generated msmpeg4v3 files (no autodetection)

xvid_ilace

Xvid interlacing bug (autodetected if fourcc==XVIX)

ump4

(autodetected if fourcc==UMP4)

no_padding

padding bug (autodetected)

amv

ac_vlc

illegal vlc bug (autodetected per fourcc)

qpel_chroma
std_qpel

old standard qpel (autodetected per fourcc/version)

qpel_chroma2
direct_blocksize

direct-qpel-blocksize bug (autodetected per fourcc/version)

edge

edge padding bug (autodetected per fourcc/version)

hpel_chroma
dc_clip

workaround various bugs in microsofts broken decoders

trunc

trancated frames

−lelim[:stream_specifier] integer (output,video)

single coefficient elimination threshold for luminance (negative values also consider dc coefficient)

−celim[:stream_specifier] integer (output,video)

single coefficient elimination threshold for chrominance (negative values also consider dc coefficient)

−strict[:stream_specifier] integer (input/output,audio,video)

how strictly to follow the standards

Possible values:
very

strictly conform to a older more strict version of the spec or reference software

strict

strictly conform to all the things in the spec no matter what consequences

normal
unofficial

allow unofficial extensions

experimental

allow non standardized experimental things

−b_qoffset[:stream_specifier] float (output,video)

qp offset between P and B frames

−err_detect[:stream_specifier] flags (input,audio,video)

set error detection flags

Possible values:
crccheck

verify embedded CRCs

bitstream

detect bitstream specification deviations

buffer

detect improper bitstream length

explode

abort decoding on minor error detection

careful

consider things that violate the spec and have not been seen in the wild as errors

compliant

consider all spec non compliancies as errors

aggressive

consider things that a sane encoder should not do as an error

−has_b_frames[:stream_specifier] integer ()
−block_align[:stream_specifier] integer ()
−mpeg_quant[:stream_specifier] integer (output,video)

use MPEG quantizers instead of H.263

−stats_out[:stream_specifier] string ()
−qsquish[:stream_specifier] float (output,video)

how to keep quantizer between qmin and qmax (0 = clip, 1 = use differentiable function)

−rc_qmod_amp[:stream_specifier] float (output,video)

experimental quantizer modulation

−rc_qmod_freq[:stream_specifier] integer (output,video)

experimental quantizer modulation

−rc_override_count[:stream_specifier] integer ()
−rc_eq[:stream_specifier] string (output,video)

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.

−maxrate[:stream_specifier] integer (output,audio,video)

Set max bitrate tolerance (in bits/s). Requires bufsize to be set.

−minrate[:stream_specifier] integer (output,audio,video)

Set min bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use elsewise.

−bufsize[:stream_specifier] integer (output,audio,video)

set ratecontrol buffer size (in bits)

−rc_buf_aggressivity[:stream_specifier] float (output,video)

currently useless

−i_qfactor[:stream_specifier] float (output,video)

qp factor between P and I frames

−i_qoffset[:stream_specifier] float (output,video)

qp offset between P and I frames

−rc_init_cplx[:stream_specifier] float (output,video)

initial complexity for 1−pass encoding

−dct[:stream_specifier] integer (output,video)

DCT algorithm

Possible values:
auto

autoselect a good one (default)

fastint

fast integer

	int		accurate integer
	mmx

altivec
faan

floating point AAN DCT

−lumi_mask[:stream_specifier] float (output,video)

compresses bright areas stronger than medium ones

−tcplx_mask[:stream_specifier] float (output,video)

temporal complexity masking

−scplx_mask[:stream_specifier] float (output,video)

spatial complexity masking

−p_mask[:stream_specifier] float (output,video)

inter masking

−dark_mask[:stream_specifier] float (output,video)

compresses dark areas stronger than medium ones

−idct[:stream_specifier] integer (input/output,video)

select IDCT implementation

Possible values:
auto

int

simple
simplemmx
libmpeg2mmx

	mmi
	arm

altivec

sh4

simplearm
simplearmv5te
simplearmv6
simpleneon
simplealpha
h264

	vp3
	ipp

xvidmmx
faani

floating point AAN IDCT

−slice_count[:stream_specifier] integer ()
−ec[:stream_specifier] flags (input,video)

set error concealment strategy

Possible values:
guess_mvs

iterative motion vector ( MV ) search (slow)

deblock

use strong deblock filter for damaged MBs

−bits_per_coded_sample[:stream_specifier] integer ()
−pred[:stream_specifier] integer (output,video)

prediction method

Possible values:
left
plane
median

−aspect[:stream_specifier] rational number (output,video)

sample aspect ratio

−debug[:stream_specifier] flags (input/output,audio,video,subtitles)

print specific debug info

Possible values:
pict

picture info

rate control

bitstream
mb_type

macroblock ( MB ) type

	qp		per-block quantization parameter ( QP )
	mv		motion vector

dct_coeff
skip
startcode

	pts
	er		error recognition

mmco

memory management control operations (H.264)

bugs
vis_qp

visualize quantization parameter ( QP ), lower QP are tinted greener

vis_mb_type

visualize block types

buffers

picture buffer allocations

thread_ops

threading operations

−vismv[:stream_specifier] integer (input,video)

visualize motion vectors (MVs)

Possible values:

	pf		forward predicted MVs of P−frames
	bf		forward predicted MVs of B−frames
	bb		backward predicted MVs of B−frames

−cmp[:stream_specifier] integer (output,video)

full pel me compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−subcmp[:stream_specifier] integer (output,video)

sub pel me compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−mbcmp[:stream_specifier] integer (output,video)

macroblock compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−ildctcmp[:stream_specifier] integer (output,video)

interlaced dct compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−dia_size[:stream_specifier] integer (output,video)

diamond type & size for motion estimation

−last_pred[:stream_specifier] integer (output,video)

amount of motion predictors from the previous frame

−preme[:stream_specifier] integer (output,video)

pre motion estimation

−precmp[:stream_specifier] integer (output,video)

pre motion estimation compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−pre_dia_size[:stream_specifier] integer (output,video)

diamond type & size for motion estimation pre-pass

−subq[:stream_specifier] integer (output,video)

sub pel motion estimation quality

−dtg_active_format[:stream_specifier] integer ()
−me_range[:stream_specifier] integer (output,video)

limit motion vectors range (1023 for DivX player)

−ibias[:stream_specifier] integer (output,video)

intra quant bias

−pbias[:stream_specifier] integer (output,video)

inter quant bias

−color_table_id[:stream_specifier] integer ()
−global_quality[:stream_specifier] integer (output,audio,video)
−coder[:stream_specifier] integer (output,video)

Possible values:

	vlc		variable length coder / huffman coder
	ac		arithmetic coder
	raw		raw (no encoding)
	rle		run-length coder

deflate

deflate-based coder

−context[:stream_specifier] integer (output,video)

context model

−slice_flags[:stream_specifier] integer ()
−xvmc_acceleration[:stream_specifier] integer ()
−mbd[:stream_specifier] integer (output,video)

macroblock decision algorithm (high quality mode)

Possible values:
simple

use mbcmp (default)

bits

use fewest bits

use best rate distortion

−stream_codec_tag[:stream_specifier] integer ()
−sc_threshold[:stream_specifier] integer (output,video)

scene change threshold

−lmin[:stream_specifier] integer (output,video)

min lagrange factor ( VBR )

−lmax[:stream_specifier] integer (output,video)

max lagrange factor ( VBR )

−nr[:stream_specifier] integer (output,video)

noise reduction

−rc_init_occupancy[:stream_specifier] integer (output,video)

number of bits which should be loaded into the rc buffer before decoding starts

−inter_threshold[:stream_specifier] integer (output,video)
−flags2[:stream_specifier] flags (input/output,audio,video)

Possible values:
fast

allow non spec compliant speedup tricks

sgop

Deprecated, use mpegvideo private options instead

noout

skip bitstream encoding

local_header

place global headers at every keyframe instead of in extradata

chunks

Frame data might be split into multiple chunks

showall

Show all frames before the first keyframe

skiprd

Deprecated, use mpegvideo private options instead

−error[:stream_specifier] integer (output,video)
−qns[:stream_specifier] integer (output,video)

deprecated, use mpegvideo private options instead

−threads[:stream_specifier] integer (input/output,video)

Possible values:
auto

detect a good number of threads

−me_threshold[:stream_specifier] integer (output,video)

motion estimaton threshold

−mb_threshold[:stream_specifier] integer (output,video)

macroblock threshold

−dc[:stream_specifier] integer (output,video)

intra_dc_precision

−nssew[:stream_specifier] integer (output,video)

nsse weight

−skip_top[:stream_specifier] integer (input,video)

number of macroblock rows at the top which are skipped

−skip_bottom[:stream_specifier] integer (input,video)

number of macroblock rows at the bottom which are skipped

−profile[:stream_specifier] integer (output,audio,video)

Possible values:
unknown
aac_main
aac_low
aac_ssr
aac_ltp
aac_he
aac_he_v2
aac_ld
aac_eld

dts

dts_es
dts_96_24
dts_hd_hra
dts_hd_ma

−level[:stream_specifier] integer (output,audio,video)

Possible values:
unknown

−lowres[:stream_specifier] integer (input,audio,video)

decode at 1= 1/2, 2=1/4, 3=1/8 resolutions

−skip_threshold[:stream_specifier] integer (output,video)

frame skip threshold

−skip_factor[:stream_specifier] integer (output,video)

frame skip factor

−skip_exp[:stream_specifier] integer (output,video)

frame skip exponent

−skipcmp[:stream_specifier] integer (output,video)

frame skip compare function

Possible values:

	sad		sum of absolute differences, fast (default)
	sse		sum of squared errors

satd

sum of absolute Hadamard transformed differences

dct

sum of absolute DCT transformed differences

psnr

sum of squared quantization errors (avoid, low quality)

	bit		number of bits needed for the block
	rd		rate distortion optimal, slow

zero

vsad

sum of absolute vertical differences

vsse

sum of squared vertical differences

nsse

noise preserving sum of squared differences

	w53		5/3 wavelet, only used in snow
	w97		9/7 wavelet, only used in snow

dctmax
chroma

−border_mask[:stream_specifier] float (output,video)

increases the quantizer for macroblocks close to borders

−mblmin[:stream_specifier] integer (output,video)

min macroblock lagrange factor ( VBR )

−mblmax[:stream_specifier] integer (output,video)

max macroblock lagrange factor ( VBR )

−mepc[:stream_specifier] integer (output,video)

motion estimation bitrate penalty compensation (1.0 = 256)

−skip_loop_filter[:stream_specifier] integer (input,video)

Possible values:
none
default
noref
bidir
nokey

all

−skip_idct[:stream_specifier] integer (input,video)

Possible values:
none
default
noref
bidir
nokey

all

−skip_frame[:stream_specifier] integer (input,video)

Possible values:
none
default
noref
bidir
nokey

all

−bidir_refine[:stream_specifier] integer (output,video)

refine the two motion vectors used in bidirectional macroblocks

−brd_scale[:stream_specifier] integer (output,video)

downscales frames for dynamic B−frame decision

−keyint_min[:stream_specifier] integer (output,video)

minimum interval between IDR-frames

−refs[:stream_specifier] integer (output,video)

reference frames to consider for motion compensation

−chromaoffset[:stream_specifier] integer (output,video)

chroma qp offset from luma

−trellis[:stream_specifier] integer (output,audio,video)

rate-distortion optimal quantization

−sc_factor[:stream_specifier] integer (output,video)

multiplied by qscale for each frame and added to scene_change_score

−mv0_threshold[:stream_specifier] integer (output,video)
−b_sensitivity[:stream_specifier] integer (output,video)

adjusts sensitivity of b_frame_strategy 1

−compression_level[:stream_specifier] integer (output,audio,video)
−min_prediction_order[:stream_specifier] integer (output,audio)
−max_prediction_order[:stream_specifier] integer (output,audio)
−timecode_frame_start[:stream_specifier] integer (output,video)

GOP timecode frame start number, in non drop frame format

−request_channels[:stream_specifier] integer (input,audio)

set desired number of audio channels

−bits_per_raw_sample[:stream_specifier] integer ()
−channel_layout[:stream_specifier] integer (input/output,audio)

Possible values:

−request_channel_layout[:stream_specifier] integer (input,audio)

Possible values:

−rc_max_vbv_use[:stream_specifier] float (output,video)
−rc_min_vbv_use[:stream_specifier] float (output,video)
−ticks_per_frame[:stream_specifier] integer (input/output,audio,video)
−color_primaries[:stream_specifier] integer (input/output,video)
−color_trc[:stream_specifier] integer (input/output,video)
−colorspace[:stream_specifier] integer (input/output,video)
−color_range[:stream_specifier] integer (input/output,video)
−chroma_sample_location[:stream_specifier] integer (input/output,video)
−log_level_offset[:stream_specifier] integer ()

set the log level offset

−slices[:stream_specifier] integer (output,video)

number of slices, used in parallelized encoding

−thread_type[:stream_specifier] flags (input/output,video)

select multithreading type

Possible values:
slice
frame

−audio_service_type[:stream_specifier] integer (output,audio)

audio service type

Possible values:

	ma		Main Audio Service
	ef		Effects
	vi		Visually Impaired
	hi		Hearing Impaired
	di		Dialogue
	co		Commentary
	em		Emergency
	vo		Voice Over
	ka		Karaoke

−request_sample_fmt[:stream_specifier] integer (input,audio)

sample format audio decoders should prefer

Possible values:

	u8		8−bit unsigned integer
	s16		16−bit signed integer
	s32		32−bit signed integer
	flt		32−bit float
	dbl		64−bit double
	u8p		8−bit unsigned integer planar

s16p

16−bit signed integer planar

s32p

32−bit signed integer planar

fltp

32−bit float planar

dblp

64−bit double planar

−pkt_timebase[:stream_specifier] rational number ()

Format AVOptions
−avioflags flags (input/output)

Possible values:
direct

reduce buffering

−probesize integer (input)

set probing size

−packetsize integer (output)

set packet size

−fflags flags (input/output)

Possible values:
ignidx

ignore index

genpts

generate pts

nofillin

do not fill in missing values that can be exactly calculated

noparse

disable AVParsers, this needs nofillin too

igndts

ignore dts

discardcorrupt

discard corrupted frames

sortdts

try to interleave outputted packets by dts

keepside

dont merge side data

latm

enable RTP MP4A−LATM payload

nobuffer

reduce the latency introduced by optional buffering

−analyzeduration integer (input)

how many microseconds are analyzed to estimate duration

−cryptokey hexadecimal string (input)

decryption key

−indexmem integer (input)

max memory used for timestamp index (per stream)

−rtbufsize integer (input)

max memory used for buffering real-time frames

−fdebug flags (input/output)

print specific debug info

Possible values:

−max_delay integer (input/output)

maximum muxing or demuxing delay in microseconds

−fpsprobesize integer (input)

number of frames used to probe fps

−audio_preload integer (output)

microseconds by which audio packets should be interleaved earlier

−chunk_duration integer (output)

microseconds for each chunk

−chunk_size integer (output)

size in bytes for each chunk

−f_err_detect flags (input)

set error detection flags (deprecated; use err_detect, save via avconv)

Possible values:
crccheck

verify embedded CRCs

bitstream

detect bitstream specification deviations

buffer

detect improper bitstream length

explode

abort decoding on minor error detection

careful

consider things that violate the spec and have not been seen in the wild as errors

compliant

consider all spec non compliancies as errors

aggressive

consider things that a sane encoder shouldnt do as an error

−err_detect flags (input)

set error detection flags

Possible values:
crccheck

verify embedded CRCs

bitstream

detect bitstream specification deviations

buffer

detect improper bitstream length

explode

abort decoding on minor error detection

careful

consider things that violate the spec and have not been seen in the wild as errors

compliant

consider all spec non compliancies as errors

aggressive

consider things that a sane encoder shouldnt do as an error

−use_wallclock_as_timestamps integer (input)

use wallclock as timestamps

−avoid_negative_ts integer (output)

avoid negative timestamps

Main options
−f fmt (input/output)

Force input or output file format. The format is normally auto detected for input files and guessed from file extension for output files, so this option is not needed in most cases.

−i filename (input)

input file name

−y (global)

Overwrite output files without asking.

−n (global)

Do not overwrite output files but exit if file exists.

−c[:stream_specifier] codec (input/output,per−stream)
−codec[:stream_specifier] codec (input/output,per−stream)

Select an encoder (when used before an output file) or a decoder (when used before an input file) for one or more streams. codec is the name of a decoder/encoder or a special value "copy" (output only) to indicate that the stream is not to be re-encoded.

For example

        ffmpeg −i INPUT −map 0 −c:v libx264 −c:a copy OUTPUT

encodes all video streams with libx264 and copies all audio streams.

For each stream, the last matching "c" option is applied, so

        ffmpeg −i INPUT −map 0 −c copy −c:v:1 libx264 −c:a:137 libvorbis OUTPUT

will copy all the streams except the second video, which will be encoded with libx264, and the 138th audio, which will be encoded with libvorbis.

−t duration (output)

Stop writing the output after its duration reaches duration. duration may be a number in seconds, or in "hh:mm:ss[.xxx]" form.

−fs limit_size (output)

Set the file size limit, expressed in bytes.

−ss position (input/output)

When used as an input option (before "−i"), seeks in this input file to position. When used as an output option (before an output filename), decodes but discards input until the timestamps reach position. This is slower, but more accurate.

position may be either in seconds or in "hh:mm:ss[.xxx]" form.

−itsoffset offset (input)

Set the input time offset in seconds. "[−]hh:mm:ss[.xxx]" syntax is also supported. The offset is added to the timestamps of the input files. Specifying a positive offset means that the corresponding streams are delayed by offset seconds.

−timestamp time (output)

Set the recording timestamp in the container. The syntax for time is:

        now|([(YYYY−MM−DD|YYYYMMDD)[T|t| ]]((HH:MM:SS[.m...])|(HHMMSS[.m...]))[Z|z])

If the value is "now" it takes the current time. Time is local time unless ’Z’ or ’z’ is appended, in which case it is interpreted as UTC . If the year-month-day part is not specified it takes the current year-month-day.

−metadata[:metadata_specifier] key=value (output,per−metadata)

Set a metadata key/value pair.

An optional metadata_specifier may be given to set metadata on streams or chapters. See "−map_metadata" documentation for details.

This option overrides metadata set with "−map_metadata". It is also possible to delete metadata by using an empty value.

For example, for setting the title in the output file:

        ffmpeg −i in.avi −metadata title="my title" out.flv

To set the language of the first audio stream:

        ffmpeg −i INPUT −metadata:s:a:1 language=eng OUTPUT

−target type (output)

Specify target file type ("vcd", "svcd", "dvd", "dv", "dv50"). type may be prefixed with "pal−", "ntsc−" or "film−" to use the corresponding standard. All the format options (bitrate, codecs, buffer sizes) are then set automatically. You can just type:

        ffmpeg −i myfile.avi −target vcd /tmp/vcd.mpg

Nevertheless you can specify additional options as long as you know they do not conflict with the standard, as in:

        ffmpeg −i myfile.avi −target vcd −bf 2 /tmp/vcd.mpg

−dframes number (output)

Set the number of data frames to record. This is an alias for "−frames:d".

−frames[:stream_specifier] framecount (output,per−stream)

Stop writing to the stream after framecount frames.

−q[:stream_specifier] q (output,per−stream)
−qscale[:stream_specifier] q (output,per−stream)

Use fixed quality scale ( VBR ). The meaning of q is codec-dependent.

−filter[:stream_specifier] filter_graph (output,per−stream)

filter_graph is a description of the filter graph to apply to the stream. Use "−filters" to show all the available filters (including also sources and sinks).

See also the −filter_complex option if you want to create filter graphs with multiple inputs and/or outputs.

−pre[:stream_specifier] preset_name (output,per−stream)

Specify the preset for matching stream(s).

−stats (global)

Print encoding progress/statistics. On by default.

−progress url (global)

Send program-friendly progress information to url.

Progress information is written approximately every second and at the end of the encoding process. It is made of "key=value" lines. key consists of only alphanumeric characters. The last key of a sequence of progress information is always "progress".

−stdin

Enable interaction on standard input. On by default unless standard input is used as an input. To explicitly disable interaction you need to specify "−nostdin".

Disabling interaction on standard input is useful, for example, if ffmpeg is in the background process group. Roughly the same result can be achieved with "ffmpeg ... < /dev/null" but it requires a shell.

−debug_ts (global)

Print timestamp information. It is off by default. This option is mostly useful for testing and debugging purposes, and the output format may change from one version to another, so it should not be employed by portable scripts.

TIPS

•

For streaming at very low bitrate application, use a low frame rate and a small GOP size. This is especially true for RealVideo where the Linux player does not seem to be very fast, so it can miss frames. An example is:

        ffmpeg −g 3 −r 3 −t 10 −b:v 50k −s qcif −f rv10 /tmp/b.rm

	•		The parameter ’q’ which is displayed while encoding is the current quantizer. The value 1 indicates that a very good quality could be achieved. The value 31 indicates the worst quality. If q=31 appears too often, it means that the encoder cannot compress enough to meet your bitrate. You must either increase the bitrate, decrease the frame rate or decrease the frame size.
	•		If your computer is not fast enough, you can speed up the compression at the expense of the compression ratio. You can use ’−me zero’ to speed up motion estimation, and ’−g 0’ to disable motion estimation completely (you have only I−frames, which means it is about as good as JPEG compression).
	•		To have very low audio bitrates, reduce the sampling frequency (down to 22050 Hz for MPEG audio, 22050 or 11025 for AC−3 ).
	•		To have a constant quality (but a variable bitrate), use the option ’−qscale n’ when ’n’ is between 1 (excellent quality) and 31 (worst quality).

EXAMPLES

Preset files
A preset file contains a sequence of option=value pairs, one for each line, specifying a sequence of options which can be specified also on the command line. Lines starting with the hash (’#’) character are ignored and are used to provide comments. Empty lines are also ignored. Check the presets directory in the FFmpeg source tree for examples.

Preset files are specified with the "pre" option, this option takes a preset name as input. FFmpeg searches for a file named preset_name.avpreset in the directories $AVCONV_DATADIR (if set), and $HOME/.ffmpeg, and in the data directory defined at configuration time (usually $PREFIX/share/ffmpeg) in that order. For example, if the argument is "libx264−max", it will search for the file libx264−max.avpreset.

Video and Audio grabbing
If you specify the input format and device then ffmpeg can grab video and audio directly.

        ffmpeg −f oss −i /dev/dsp −f video4linux2 −i /dev/video0 /tmp/out.mpg

Or with an ALSA audio source (mono input, card id 1) instead of OSS:

        ffmpeg −f alsa −ac 1 −i hw:1 −f video4linux2 −i /dev/video0 /tmp/out.mpg

Note that you must activate the right video source and channel before launching ffmpeg with any TV viewer such as <http://linux.bytesex.org/xawtv/> by Gerd Knorr. You also have to set the audio recording levels correctly with a standard mixer.

X11 grabbing
Grab the X11 display with ffmpeg via

        ffmpeg −f x11grab −s cif −r 25 −i :0.0 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable.

        ffmpeg −f x11grab −s cif −r 25 −i :0.0+10,20 /tmp/out.mpg

0.0 is display.screen number of your X11 server, same as the DISPLAY environment variable. 10 is the x−offset and 20 the y−offset for the grabbing.

Video and Audio file format conversion
Any supported file format and protocol can serve as input to ffmpeg:

Examples:

•

You can use YUV files as input:

        ffmpeg −i /tmp/test%d.Y /tmp/out.mpg

It will use the files:

        /tmp/test0.Y, /tmp/test0.U, /tmp/test0.V,
        /tmp/test1.Y, /tmp/test1.U, /tmp/test1.V, etc...

The Y files use twice the resolution of the U and V files. They are raw files, without header. They can be generated by all decent video decoders. You must specify the size of the image with the −s option if ffmpeg cannot guess it.

•

You can input from a raw YUV420P file:

        ffmpeg −i /tmp/test.yuv /tmp/out.avi

test.yuv is a file containing raw YUV planar data. Each frame is composed of the Y plane followed by the U and V planes at half vertical and horizontal resolution.

•

You can output to a raw YUV420P file:

        ffmpeg −i mydivx.avi hugefile.yuv

•

You can set several input files and output files:

        ffmpeg −i /tmp/a.wav −s 640x480 −i /tmp/a.yuv /tmp/a.mpg

Converts the audio file a.wav and the raw YUV video file a.yuv to MPEG file a.mpg.

•

You can also do audio and video conversions at the same time:

        ffmpeg −i /tmp/a.wav −ar 22050 /tmp/a.mp2

Converts a.wav to MPEG audio at 22050 Hz sample rate.

•

You can encode to several formats at the same time and define a mapping from input stream to output streams:

        ffmpeg −i /tmp/a.wav −map 0:a −b:a 64k /tmp/a.mp2 −map 0:a −b:a 128k /tmp/b.mp2

Converts a.wav to a.mp2 at 64 kbits and to b.mp2 at 128 kbits. ’−map file:index’ specifies which input stream is used for each output stream, in the order of the definition of output streams.

•

You can transcode decrypted VOBs:

        ffmpeg −i snatch_1.vob −f avi −c:v mpeg4 −b:v 800k −g 300 −bf 2 −c:a libmp3lame −b:a 128k snatch.avi

This is a typical DVD ripping example; the input is a VOB file, the output an AVI file with MPEG−4 video and MP3 audio. Note that in this command we use B−frames so the MPEG−4 stream is DivX5 compatible, and GOP size is 300 which means one intra frame every 10 seconds for 29.97fps input video. Furthermore, the audio stream is MP3−encoded so you need to enable LAME support by passing "−−enable−libmp3lame" to configure. The mapping is particularly useful for DVD transcoding to get the desired audio language.

NOTE: To see the supported input formats, use "ffmpeg −formats".

•

You can extract images from a video, or create a video from many images:

For extracting images from a video:

        ffmpeg −i foo.avi −r 1 −s WxH −f image2 foo−%03d.jpeg

This will extract one video frame per second from the video and will output them in files named foo−001.jpeg, foo−002.jpeg, etc. Images will be rescaled to fit the new WxH values.

If you want to extract just a limited number of frames, you can use the above command in combination with the −vframes or −t option, or in combination with −ss to start extracting from a certain point in time.

For creating a video from many images:

        ffmpeg −f image2 −i foo−%03d.jpeg −r 12 −s WxH foo.avi

The syntax "foo−%03d.jpeg" specifies to use a decimal number composed of three digits padded with zeroes to express the sequence number. It is the same syntax supported by the C printf function, but only formats accepting a normal integer are suitable.

When importing an image sequence, −i also supports expanding shell-like wildcard patterns (globbing) internally, by selecting the image2−specific "−pattern_type glob" option.

For example, for creating a video from filenames matching the glob pattern "foo−*.jpeg":

        ffmpeg −f image2 −pattern_type glob −i 'foo−*.jpeg' −r 12 −s WxH foo.avi

•

You can put many streams of the same type in the output:

        ffmpeg −i test1.avi −i test2.avi −map 0.3 −map 0.2 −map 0.1 −map 0.0 −c copy test12.nut

The resulting output file test12.avi will contain first four streams from the input file in reverse order.

•

To force CBR video output:

        ffmpeg −i myfile.avi −b 4000k −minrate 4000k −maxrate 4000k −bufsize 1835k out.m2v

•

The four options lmin, lmax, mblmin and mblmax use ’lambda’ units, but you may use the QP2LAMBDA constant to easily convert from ’q’ units:

        ffmpeg −i src.ext −lmax 21*QP2LAMBDA dst.ext

SYNTAX

When evaluating specific formats, FFmpeg uses internal library parsing functions, shared by the tools. This section documents the syntax of some of these formats.

Date
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.

Time duration
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.

Video size
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:
sqcif

128x96

qcif

176x144

cif

352x288

4cif

704x576

16cif

1408x1152

qqvga

160x120

qvga

320x240

vga

640x480

svga

800x600

xga

1024x768

uxga

1600x1200

qxga

2048x1536

sxga

1280x1024

qsxga

2560x2048

hsxga

5120x4096

wvga

852x480

wxga

1366x768

wsxga

1600x1024

wuxga

1920x1200

woxga

2560x1600

wqsxga

3200x2048

wquxga

3840x2400

whsxga

6400x4096

whuxga

7680x4800

	cga		320x200
	ega		640x350

hd480

852x480

hd720

1280x720

hd1080

1920x1080

Video rate
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:
ntsc

30000/1001

pal

25/1

qntsc

30000/1

qpal

25/1

sntsc

30000/1

spal

25/1

film

24/1

ntsc-film

24000/1

Ratio
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.

Color
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.

EXPRESSION EVALUATION

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:
sinh(x)

Compute hyperbolic sine of x.

cosh(x)

Compute hyperbolic cosine of x.

tanh(x)

Compute hyperbolic tangent of x.

sin(x)

Compute sine of x.

cos(x)

Compute cosine of x.

tan(x)

Compute tangent of x.

atan(x)

Compute arctangent of x.

asin(x)

Compute arcsine of x.

acos(x)

Compute arccosine of x.

exp(x)

Compute exponential of x (with base "e", the Euler’s number).

log(x)

Compute natural logarithm of x.

abs(x)

Compute absolute value of x.

squish(x)

Compute expression "1/(1 + exp(4*x))".

gauss(x)

Compute Gauss function of x, corresponding to "exp(−x*x/2) / sqrt(2*PI)".

isinf(x)

Return 1.0 if x is +/−INFINITY, 0.0 otherwise.

isnan(x)

Return 1.0 if x is NAN , 0.0 otherwise.

mod(x, y)

Compute the remainder of division of x by y.

max(x, y)

Return the maximum between x and y.

min(x, y)

Return the maximum between x and y.

eq(x, y)

Return 1 if x and y are equivalent, 0 otherwise.

gte(x, y)

Return 1 if x is greater than or equal to y, 0 otherwise.

gt(x, y)

Return 1 if x is greater than y, 0 otherwise.

lte(x, y)

Return 1 if x is lesser than or equal to y, 0 otherwise.

lt(x, y)

Return 1 if x is lesser than y, 0 otherwise.

st(var, expr)

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.

ld(var)

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.

while(cond, expr)

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.

ceil(expr)

Round the value of expression expr upwards to the nearest integer. For example, "ceil(1.5)" is "2.0".

floor(expr)

Round the value of expression expr downwards to the nearest integer. For example, "floor(−1.5)" is "−2.0".

trunc(expr)

Round the value of expression expr towards zero to the nearest integer. For example, "trunc(−1.5)" is "−1.0".

sqrt(expr)

Compute the square root of expr. This is equivalent to "(expr)^.5".

not(expr)

Return 1.0 if expr is zero, 0.0 otherwise.

pow(x, y)

Compute the power of x elevated y, it is equivalent to "(x)^(y)".

random(x)

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.

hypot(x, y)

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.

gcd(x, y)

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.

if(x, y)

Evaluate x, and if the result is non-zero return the result of the evaluation of y, return 0 otherwise.

ifnot(x, y)

Evaluate x, and if the result is zero return the result of the evaluation of y, return 0 otherwise.

taylor(expr, x) taylor(expr, x, id)

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.

root(expr, max)

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:

	PI		area of the unit disc, approximately 3.14
	E		exp(1) (Euler’s number), approximately 2.718
	PHI		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.

	y		−24 / −80
	z		−21 / −70
	a		−18 / −60
	f		−15 / −50
	p		−12 / −40
	n		−9 / −30
	u		−6 / −20
	m		−3 / −10
	c		−2
	d		−1
	h		2
	k		3 / 10
	K		3 / 10
	M		6 / 20
	G		9 / 30
	T		12 / 40
	P		15 / 40
	E		18 / 50
	Z		21 / 60
	Y		24 / 70

DECODERS

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.

VIDEO DECODERS

A description of some of the currently available video decoders follows.

rawvideo
Raw video decoder.

This decoder decodes rawvideo streams.

Options
top top_field_first

Specify the assumed field type of the input video.

	−1		the video is assumed to be progressive (default)
	0		bottom-field-first is assumed
	1		top-field-first is assumed

AUDIO DECODERS

ffwavesynth
Internal wave synthetizer.

This decoder generates wave patterns according to predefined sequences. Its use is purely internal and the format of the data it accepts is not publicly documented.

libcelt
libcelt decoder wrapper

libcelt allows libavcodec to decode the Xiph CELT ultra-low delay audio codec. Requires the presence of the libcelt headers and library during configuration. You need to explicitly configure the build with "−−enable−libcelt".

libgsm
libgsm decoder wrapper

libgsm allows libavcodec to decode the GSM full rate audio codec. Requires the presence of the libgsm headers and library during configuration. You need to explicitly configure the build with "−−enable−libgsm".

This decoder supports both the ordinary GSM and the Microsoft variant.

libilbc
libilbc decoder wrapper

libilbc allows libavcodec to decode the Internet Low Bitrate Codec (iLBC) audio codec. Requires the presence of the libilbc headers and library during configuration. You need to explicitly configure the build with "−−enable−libilbc".

Options

The following option is supported by the libilbc wrapper.
enhance

Enable the enhancement of the decoded audio when set to 1. The default value is 0 (disabled).

libopencore-amrnb
libopencore-amrnb decoder wrapper

libopencore-amrnb allows libavcodec to decode the Adaptive Multi-Rate Narrowband audio codec. Using it requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "−−enable−libopencore−amrnb".

An FFmpeg native decoder for AMR-NB exists, so users can decode AMR-NB without this library.

libopencore-amrwb
libopencore-amrwb decoder wrapper.

libopencore-amrwb allows libavcodec to decode the Adaptive Multi-Rate Wideband audio codec. Using it requires the presence of the libopencore-amrwb headers and library during configuration. You need to explicitly configure the build with "−−enable−libopencore−amrwb".

An FFmpeg native decoder for AMR-WB exists, so users can decode AMR-WB without this library.

ENCODERS

Encoders are configured elements in FFmpeg which allow the encoding of multimedia streams.

When you configure your FFmpeg build, all the supported native encoders are enabled by default. Encoders requiring an external library must be enabled manually via the corresponding "−−enable−lib" option. You can list all available encoders using the configure option "−−list−encoders".

You can disable all the encoders with the configure option "−−disable−encoders" and selectively enable / disable single encoders with the options "−−enable−encoder=ENCODER" / "−−disable−encoder=ENCODER".

The option "−codecs" of the ff* tools will display the list of enabled encoders.

AUDIO ENCODERS

A description of some of the currently available audio encoders follows.

ac3 and ac3_fixed
AC−3 audio encoders.

These encoders implement part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet).

The ac3 encoder uses floating-point math, while the ac3_fixed encoder only uses fixed-point integer math. This does not mean that one is always faster, just that one or the other may be better suited to a particular system. The floating-point encoder will generally produce better quality audio for a given bitrate. The ac3_fixed encoder is not the default codec for any of the output formats, so it must be specified explicitly using the option "−acodec ac3_fixed" in order to use it.

AC−3 Metadata

The AC−3 metadata options are used to set parameters that describe the audio, but in most cases do not affect the audio encoding itself. Some of the options do directly affect or influence the decoding and playback of the resulting bitstream, while others are just for informational purposes. A few of the options will add bits to the output stream that could otherwise be used for audio data, and will thus affect the quality of the output. Those will be indicated accordingly with a note in the option list below.

These parameters are described in detail in several publicly-available documents.
*<<http://www.atsc.org/cms/standards/a_52−2010.pdf>>
*<<http://www.atsc.org/cms/standards/a_54a_with_corr_1.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz−_Shared_Assets/English_PDFs/Professional/18_Metadata.Guide.pdf>>
*<<http://www.dolby.com/uploadedFiles/zz−_Shared_Assets/English_PDFs/Professional/46_DDEncodingGuidelines.pdf>>

Metadata Control Options
−per_frame_metadata boolean

Allow Per-Frame Metadata. Specifies if the encoder should check for changing metadata for each frame.

	0		The metadata values set at initialization will be used for every frame in the stream. (default)
	1		Metadata values can be changed before encoding each frame.

Downmix Levels
−center_mixlev level

Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo. This field will only be written to the bitstream if a center channel is present. The value is specified as a scale factor. There are 3 valid values:
0.707

Apply −3dB gain

0.595

Apply −4.5dB gain (default)

0.500

Apply −6dB gain

−surround_mixlev level

Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo. This field will only be written to the bitstream if one or more surround channels are present. The value is specified as a scale factor. There are 3 valid values:
0.707

Apply −3dB gain

0.500

Apply −6dB gain (default)

0.000

Silence Surround Channel(s)

Audio Production Information

Audio Production Information is optional information describing the mixing environment. Either none or both of the fields are written to the bitstream.
−mixing_level number

Mixing Level. Specifies peak sound pressure level ( SPL ) in the production environment when the mix was mastered. Valid values are 80 to 111, or −1 for unknown or not indicated. The default value is −1, but that value cannot be used if the Audio Production Information is written to the bitstream. Therefore, if the "room_type" option is not the default value, the "mixing_level" option must not be −1.

−room_type type

Room Type. Describes the equalization used during the final mixing session at the studio or on the dubbing stage. A large room is a dubbing stage with the industry standard X−curve equalization; a small room has flat equalization. This field will not be written to the bitstream if both the "mixing_level" option and the "room_type" option have the default values.

notindicated

Not Indicated (default)

large

Large Room

small

Small Room

Other Metadata Options
−copyright boolean

	0
	off		No Copyright Exists (default)
	1
	on		Copyright Exists

−dialnorm value

Dialogue Normalization. Indicates how far the average dialogue level of the program is below digital 100% full scale (0 dBFS). This parameter determines a level shift during audio reproduction that sets the average volume of the dialogue to a preset level. The goal is to match volume level between program sources. A value of −31dB will result in no volume level change, relative to the source volume, during audio reproduction. Valid values are whole numbers in the range −31 to −1, with −31 being the default.

−dsur_mode mode

Dolby Surround Mode. Specifies whether the stereo signal uses Dolby Surround (Pro Logic). This field will only be written to the bitstream if the audio stream is stereo. Using this option does NOT mean the encoder will actually apply Dolby Surround processing.

notindicated

Not Indicated (default)

	1
	off		Not Dolby Surround Encoded
	2
	on		Dolby Surround Encoded

−original boolean

Original Bit Stream Indicator. Specifies whether this audio is from the original source and not a copy.

	0
	off		Not Original Source
	1
	on		Original Source (default)

Extended Bitstream Information

The extended bitstream options are part of the Alternate Bit Stream Syntax as specified in Annex D of the A/52:2010 standard. It is grouped into 2 parts. If any one parameter in a group is specified, all values in that group will be written to the bitstream. Default values are used for those that are written but have not been specified. If the mixing levels are written, the decoder will use these values instead of the ones specified in the "center_mixlev" and "surround_mixlev" options if it supports the Alternate Bit Stream Syntax.

Extended Bitstream Information − Part 1
−dmix_mode mode

Preferred Stereo Downmix Mode. Allows the user to select either Lt/Rt (Dolby Surround) or Lo/Ro (normal stereo) as the preferred stereo downmix mode.

notindicated

Not Indicated (default)

ltrt

Lt/Rt Downmix Preferred

loro

Lo/Ro Downmix Preferred

−ltrt_cmixlev level

Lt/Rt Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lt/Rt mode.
1.414

Apply +3dB gain

1.189

Apply +1.5dB gain

1.000

Apply 0dB gain

0.841

Apply −1.5dB gain

0.707

Apply −3.0dB gain

0.595

Apply −4.5dB gain (default)

0.500

Apply −6.0dB gain

0.000

Silence Center Channel

−ltrt_surmixlev level

Lt/Rt Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lt/Rt mode.
0.841

Apply −1.5dB gain

0.707

Apply −3.0dB gain

0.595

Apply −4.5dB gain

0.500

Apply −6.0dB gain (default)

0.000

Silence Surround Channel(s)

−loro_cmixlev level

Lo/Ro Center Mix Level. The amount of gain the decoder should apply to the center channel when downmixing to stereo in Lo/Ro mode.
1.414

Apply +3dB gain

1.189

Apply +1.5dB gain

1.000

Apply 0dB gain

0.841

Apply −1.5dB gain

0.707

Apply −3.0dB gain

0.595

Apply −4.5dB gain (default)

0.500

Apply −6.0dB gain

0.000

Silence Center Channel

−loro_surmixlev level

Lo/Ro Surround Mix Level. The amount of gain the decoder should apply to the surround channel(s) when downmixing to stereo in Lo/Ro mode.
0.841

Apply −1.5dB gain

0.707

Apply −3.0dB gain

0.595

Apply −4.5dB gain

0.500

Apply −6.0dB gain (default)

0.000

Silence Surround Channel(s)

Extended Bitstream Information − Part 2
−dsurex_mode mode

Dolby Surround EX Mode. Indicates whether the stream uses Dolby Surround EX (7.1 matrixed to 5.1). Using this option does NOT mean the encoder will actually apply Dolby Surround EX processing.

notindicated

Not Indicated (default)

	1
	on		Dolby Surround EX Off
	2
	off		Dolby Surround EX On

−dheadphone_mode mode

Dolby Headphone Mode. Indicates whether the stream uses Dolby Headphone encoding (multi-channel matrixed to 2.0 for use with headphones). Using this option does NOT mean the encoder will actually apply Dolby Headphone processing.

notindicated

Not Indicated (default)

	1
	on		Dolby Headphone Off
	2
	off		Dolby Headphone On

−ad_conv_type type

A/D Converter Type. Indicates whether the audio has passed through HDCD A/D conversion.

standard

Standard A/D Converter (default)

hdcd

HDCD A/D Converter

Other AC−3 Encoding Options
−stereo_rematrixing boolean

Stereo Rematrixing. Enables/Disables use of rematrixing for stereo input. This is an optional AC−3 feature that increases quality by selectively encoding the left/right channels as mid/side. This option is enabled by default, and it is highly recommended that it be left as enabled except for testing purposes.

Floating-Point-Only AC−3 Encoding Options

These options are only valid for the floating-point encoder and do not exist for the fixed-point encoder due to the corresponding features not being implemented in fixed-point.
−channel_coupling boolean

Enables/Disables use of channel coupling, which is an optional AC−3 feature that increases quality by combining high frequency information from multiple channels into a single channel. The per-channel high frequency information is sent with less accuracy in both the frequency and time domains. This allows more bits to be used for lower frequencies while preserving enough information to reconstruct the high frequencies. This option is enabled by default for the floating-point encoder and should generally be left as enabled except for testing purposes or to increase encoding speed.

−1

auto

Selected by Encoder (default)

	0
	off		Disable Channel Coupling
	1
	on		Enable Channel Coupling

−cpl_start_band number

Coupling Start Band. Sets the channel coupling start band, from 1 to 15. If a value higher than the bandwidth is used, it will be reduced to 1 less than the coupling end band. If auto is used, the start band will be determined by the encoder based on the bit rate, sample rate, and channel layout. This option has no effect if channel coupling is disabled.

−1

auto

Selected by Encoder (default)

libmp3lame
LAME (Lame Ain’t an MP3 Encoder) MP3 encoder wrapper

Requires the presence of the libmp3lame headers and library during configuration. You need to explicitly configure the build with "−−enable−libmp3lame".

Option Mapping

The following options are supported by the libmp3lame wrapper, the LAME-equivalent options follow the FFmpeg ones.
FFmpeg : LAME
b : b

FFmpeg "b" option is expressed in bits/s, lame "bitrate" in kilobits/s.

q : V

Quality setting for VBR .

compression_level : q

Algorithm quality. Valid options are integers from 0−9.

reservoir : N.A.

Enable use of bit reservoir. LAME has this enabled by default.

joint_stereo : −m j

Enables the the encoder to use (on a frame by frame basis) either L/R stereo or mid/side stereo.

libopencore-amrnb
OpenCORE Adaptive Multi-Rate Narrowband encoder.

Requires the presence of the libopencore-amrnb headers and library during configuration. You need to explicitly configure the build with "−−enable−libopencore−amrnb −−enable−version3".

This is a mono-only encoder. Officially it only supports 8000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

Set bitrate in bits per second. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

4750
5150
5900
6700
7400
7950
10200
12200

dtx

Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).

libtwolame
TwoLAME MP2 encoder wrapper

Requires the presence of the libtwolame headers and library during configuration. You need to explicitly configure the build with "−−enable−libtwolame".

Options Mapping

The following options are supported by the libtwolame wrapper. The TwoLAME-equivalent options follow the FFmpeg ones and are in parentheses.

	b		(b) Set bitrate in bits/s. Note that FFmpeg `"b"` option is expressed in bits/s, twolame `"b"` in kilobits/s. The default value is 128k.
	q		(V) Set quality for experimental VBR support. Maximum value range is from −50 to 50, useful range is from −10 to 10.

mode

(mode) Set MPEG mode. Possible values:
auto

Choose mode automatically based on the input. This is the default.

stereo

Stereo

joint_stereo

Joint stereo

dual_channel

Dual channel

mono

Mono

psymodel

(psyc-mode) Set psychoacoustic model to use in encoding. The argument must be an integer between −1 and 4, inclusive. The higher the value, the better the quality. The default value is 3.

energy_levels

(energy) Enable energy levels extensions when set to 1. The default value is 0 (disabled).

error_protection

(protect) Enable CRC error protection when set to 1. The default value is 0 (disabled).

copyright

(copyright) Set MPEG audio copyright flag when set to 1. The default value is 0 (disabled).

original

(original) Set MPEG audio original flag when set to 1. The default value is 0 (disabled).

libvo-aacenc
VisualOn AAC encoder

Requires the presence of the libvo-aacenc headers and library during configuration. You need to explicitly configure the build with "−−enable−libvo−aacenc −−enable−version3".

Options

The VisualOn AAC encoder only support encoding AAC-LC and up to 2 channels. It is also CBR-only. It is considered to be worse than the native experimental FFmpeg AAC encoder.

Bitrate.

libvo-amrwbenc
VisualOn Adaptive Multi-Rate Wideband encoder

Requires the presence of the libvo-amrwbenc headers and library during configuration. You need to explicitly configure the build with "−−enable−libvo−amrwbenc −−enable−version3".

This is a mono-only encoder. Officially it only supports 16000Hz sample rate, but you can override it by setting strict to unofficial or lower.

Options

Set bitrate in bits/s. Only the following bitrates are supported, otherwise libavcodec will round to the nearest valid bitrate.

6600
8850
12650
14250
15850
18250
19850
23050
23850

dtx

Allow discontinuous transmission (generate comfort noise) when set to 1. The default value is 0 (disabled).

VIDEO ENCODERS

A description of some of the currently available video encoders follows.

libvpx
VP8 format supported through libvpx.

Requires the presence of the libvpx headers and library during configuration. You need to explicitly configure the build with "−−enable−libvpx".

Options

Mapping from FFmpeg to libvpx options with conversion notes in parentheses.
threads

g_threads

profile

g_profile

	vb		rc_target_bitrate
	g		kf_max_dist

keyint_min

kf_min_dist

qmin

rc_min_quantizer

qmax

rc_max_quantizer

bufsize, vb

rc_buf_sz "(bufsize * 1000 / vb)"

rc_buf_optimal_sz "(bufsize * 1000 / vb * 5 / 6)"

rc_init_occupancy, vb

rc_buf_initial_sz "(rc_init_occupancy * 1000 / vb)"

rc_buffer_aggressivity

rc_undershoot_pct

skip_threshold

rc_dropframe_thresh

qcomp

rc_2pass_vbr_bias_pct

maxrate, vb

rc_2pass_vbr_maxsection_pct "(maxrate * 100 / vb)"

minrate, vb

rc_2pass_vbr_minsection_pct "(minrate * 100 / vb)"

minrate, maxrate, vb

"VPX_CBR" "(minrate == maxrate == vb)"

crf

"VPX_CQ", "VP8E_SET_CQ_LEVEL"

quality

best

"VPX_DL_BEST_QUALITY"

good

"VPX_DL_GOOD_QUALITY"

realtime

"VPX_DL_REALTIME"

speed

"VP8E_SET_CPUUSED"

"VP8E_SET_NOISE_SENSITIVITY"

mb_threshold

"VP8E_SET_STATIC_THRESHOLD"

slices

"VP8E_SET_TOKEN_PARTITIONS"

max-intra-rate

"VP8E_SET_MAX_INTRA_BITRATE_PCT"

force_key_frames

"VPX_EFLAG_FORCE_KF"

Alternate reference frame related

vp8flags altref

"VP8E_SET_ENABLEAUTOALTREF"

arnr_max_frames

"VP8E_SET_ARNR_MAXFRAMES"

arnr_type

"VP8E_SET_ARNR_TYPE"

arnr_strength

"VP8E_SET_ARNR_STRENGTH"

rc_lookahead

g_lag_in_frames

vp8flags error_resilient

g_error_resilient

For more information about libvpx see: <http://www.webmproject.org/>

libx264
H.264 / AVC / MPEG−4 AVC / MPEG−4 part 10 format supported through libx264.

Requires the presence of the libx264 headers and library during configuration. You need to explicitly configure the build with "−−enable−libx264".

Options
preset preset_name

Set the encoding preset.

tune tune_name

Tune the encoding params.

fastfirstpass bool

Use fast settings when encoding first pass, default value is 1.

profile profile_name

Set profile restrictions.

level level

Specify level (as defined by Annex A). Deprecated in favor of x264opts.

passlogfile filename

Specify filename for 2 pass stats. Deprecated in favor of x264opts (see stats libx264 option).

wpredp wpred_type

Specify Weighted prediction for P−frames. Deprecated in favor of x264opts (see weightp libx264 option).

x264opts options

Allow to set any x264 option, see x264 −−fullhelp for a list.

options is a list of key=value couples separated by ":".

For example to specify libx264 encoding options with ffmpeg:

        ffmpeg −i foo.mpg −vcodec libx264 −x264opts keyint=123:min−keyint=20 −an out.mkv

For more information about libx264 and the supported options see: <http://www.videolan.org/developers/x264.html>

libxvid
Xvid MPEG−4 Part 2 encoder wrapper.

This encoder requires the presence of the libxvidcore headers and library during configuration. You need to explicitly configure the build with "−−enable−libxvid −−enable−gpl".

The native "mpeg4" encoder supports the MPEG−4 Part 2 format, so users can encode to this format without this library.

Options

The following options are supported by the libxvid wrapper. Some of the following options are listed but are not documented, and correspond to shared codec options. The other shared options which are not listed have no effect for the libxvid encoder.

	b
	g

qmin
qmax
mpeg_quant
threads

b_qfactor
b_qoffset
flags

Set specific encoding flags. Possible values:

	mv4		Use four motion vector by macroblock.
	aic		Enable high quality AC prediction.

gray

Only encode grayscale.

gmc

Enable the use of global motion compensation ( GMC ).

qpel

Enable quarter-pixel motion compensation.

cgop

Enable closed GOP .

global_header

Place global headers in extradata instead of every keyframe.

trellis
me_method

Set motion estimation method. Possible values in decreasing order of speed and increasing order of quality:
zero

Use no motion estimation (default).

phods

	x1
	log		Enable advanced diamond zonal search for 16x16 blocks and half-pixel refinement for 16x16 blocks. x1 and log are aliases for phods.

epzs

Enable all of the things described above, plus advanced diamond zonal search for 8x8 blocks, half-pixel refinement for 8x8 blocks, and motion estimation on chroma planes.

full

Enable all of the things described above, plus extended 16x16 and 8x8 blocks search.

mbd

Set macroblock decision algorithm. Possible values in the increasing order of quality:

simple

Use macroblock comparing function algorithm (default).

bits

Enable rate distortion-based half pixel and quarter pixel refinement for 16x16 blocks.

Enable all of the things described above, plus rate distortion-based half pixel and quarter pixel refinement for 8x8 blocks, and rate distortion-based search using square pattern.

DEMUXERS

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.

image2
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:
framerate

Set the framerate for the video stream. It defaults to 25.

loop

If set to 1, loop over the input. Default value is 0.

pattern_type

Select the pattern type used to interpret the provided filename.

pattern_type accepts one of the following values.
sequence

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

glob

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.

glob_sequence (deprecated, will be removed)

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.

pixel_format

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.

start_number

Set the index of the file matched by the image file pattern to start to read from. Default value is 0.

start_number_range

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.

video_size

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.

Examples

•

Use ffmpeg for creating a video from the images in the file sequence img−001.jpeg, img−002.jpeg, ..., assuming an input frame rate of 10 frames per second:

        ffmpeg −i 'img−%03d.jpeg' −r 10 out.mkv

•

As above, but start by reading from a file with index 100 in the sequence:

        ffmpeg −start_number 100 −i 'img−%03d.jpeg' −r 10 out.mkv

•

Read images matching the "*.png" glob pattern , that is all the files terminating with the ".png" suffix:

        ffmpeg −pattern_type glob −i "*.png" −r 10 out.mkv

applehttp
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".

sbg
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

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.

aiff
Audio Interchange File Format muxer.

It accepts the following options:
write_id3v2

Enable ID3v2 tags writing when set to 1. Default is 0 (disabled).

id3v2_version

Select ID3v2 version to write. Currently only version 3 and 4 (aka. ID3v2.3 and ID3v2.4) are supported. The default is version 4.

crc
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=0x CRC , 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 −

INPUT DEVICES

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
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>

bktr
BSD video input device.

dshow
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.

Options

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.
video_size

Set the video size in the captured video.

framerate

Set the framerate in the captured video.

sample_rate

Set the sample rate (in Hz) of the captured audio.

sample_size

Set the sample size (in bits) of the captured audio.

channels

Set the number of channels in the captured audio.

list_devices

If set to true, print a list of devices and exit.

list_options

If set to true, print a list of selected device’s options and exit.

video_device_number

Set video device number for devices with same name (starts at 0, defaults to 0).

audio_device_number

Set audio device number for devices with same name (starts at 0, defaults to 0).

pixel_format

Select pixel format to be used by DirectShow. This may only be set when the video codec is not set or set to rawvideo.

audio_buffer_size

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>

Examples

•

Print the list of DirectShow supported devices and exit:

        $ ffmpeg −list_devices true −f dshow −i dummy

•

Open video device Camera:

        $ ffmpeg −f dshow −i video="Camera"

•

Open second video device with name Camera:

        $ ffmpeg −f dshow −video_device_number 1 −i video="Camera"

•

Open video device Camera and audio device Microphone:

        $ ffmpeg −f dshow −i video="Camera":audio="Microphone"

•

Print the list of supported options in selected device and exit:

        $ ffmpeg −list_options true −f dshow −i video="Camera"

dv1394
Linux DV 1394 input device.

fbdev
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).

iec61883
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.

Options
dvtype

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.

dvbuffer

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.

Examples

•

Grab and show the input of a FireWire DV/HDV device.

        ffplay −f iec61883 −i auto

•

Grab and record the input of a FireWire DV/HDV device, using a packet buffer of 100000 packets if the source is HDV .

        ffmpeg −f iec61883 −i auto −hdvbuffer 100000 out.mpg

jack
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/>

lavfi
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.

Options
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.

Examples

•

Create a color video stream and play it back with ffplay:

        ffplay −f lavfi −graph "color=pink [out0]" dummy

•

As the previous example, but use filename for specifying the graph description, and omit the "out0" label:

        ffplay −f lavfi color=pink

•

Create three different video test filtered sources and play them:

        ffplay −f lavfi −graph "testsrc [out0]; testsrc,hflip [out1]; testsrc,negate [out2]" test3

•

Read an audio stream from a file using the amovie source and play it back with ffplay:

        ffplay −f lavfi "amovie=test.wav"

•

Read an audio stream and a video stream and play it back with ffplay:

        ffplay −f lavfi "movie=test.avi[out0];amovie=test.wav[out1]"

libdc1394
IIDC1394 input device, based on libdc1394 and libraw1394.

openal
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:
Creative

The official Windows implementation, providing hardware acceleration with supported devices and software fallback. See <http://openal.org/>.

OpenAL Soft

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>.

Apple

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.

Options
channels

Set the number of channels in the captured audio. Only the values 1 (monaural) and 2 (stereo) are currently supported. Defaults to 2.

sample_size

Set the sample size (in bits) of the captured audio. Only the values 8 and 16 are currently supported. Defaults to 16.

sample_rate

Set the sample rate (in Hz) of the captured audio. Defaults to 44.1k.

list_devices

If set to true, print a list of devices and exit. Defaults to false.

Examples

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.

oss
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>

pulse
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

server AVOption

The syntax is:

        −server <server name>

Connects to a specific server.

name AVOption

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

stream_name AVOption

The syntax is:

        −stream_name <stream name>

Specify the stream name pulse will use when showing active streams, by default it is "record"

sample_rate AVOption

The syntax is:

        −sample_rate <samplerate>

Specify the samplerate in Hz, by default 48kHz is used.

channels AVOption

The syntax is:

        −channels <N>

Specify the channels in use, by default 2 (stereo) is set.

frame_size AVOption

The syntax is:

        −frame_size <bytes>

Specify the number of byte per frame, by default it is set to 1024.

fragment_size AVOption

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
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
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".

vfwcap
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.

x11grab
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

Options
draw_mouse

Specify whether to draw the mouse pointer. A value of 0 specify not to draw the pointer. Default value is 1.

follow_mouse

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

framerate

Set the grabbing frame rate. Default value is "ntsc", corresponding to a framerate of "30000/1001".

show_region

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

video_size

Set the video frame size. Default value is "vga".

OUTPUT DEVICES

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
ALSA (Advanced Linux Sound Architecture) output device.

caca
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>

Options
window_title

Set the CACA window title, if not specified default to the filename specified for the output device.

window_size

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.

driver

Set display driver.

algorithm

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".

antialias

Set antialias method. Antialiasing smoothens the rendered image and avoids the commonly seen staircase effect. The accepted values are listed with "−list_dither antialiases".

charset

Set which characters are going to be used when rendering text. The accepted values are listed with "−list_dither charsets".

color

Set color to be used when rendering text. The accepted values are listed with "−list_dither colors".

list_drivers

If set to true, print a list of available drivers and exit.

list_dither

List available dither options related to the argument. The argument must be one of "algorithms", "antialiases", "charsets", "colors".

Examples

•

The following command shows the ffmpeg output is an CACA window, forcing its size to 80x25:

        ffmpeg −i INPUT −vcodec rawvideo −pix_fmt rgb24 −window_size 80x25 −f caca −

•

Show the list of available drivers and exit:

        ffmpeg −i INPUT −pix_fmt rgb24 −f caca −list_drivers true −

•

Show the list of available dither colors and exit:

        ffmpeg −i INPUT −pix_fmt rgb24 −f caca −list_dither colors −

oss
OSS (Open Sound System) output device.

sdl
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/>

Options
window_title

Set the SDL window title, if not specified default to the filename specified for the output device.

icon_title

Set the name of the iconified SDL window, if not specified it is set to the same value of window_title.

window_size

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.

Examples

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
sndio audio output device.

PROTOCOLS

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.

bluray
Read BluRay playlist.

The accepted options are:
angle

BluRay angle

chapter

Start chapter (1...N)

playlist

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

concat
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
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
Gopher protocol.

hls
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
HTTP (Hyper Text Transfer Protocol).

mmst
MMS (Microsoft Media Server) protocol over TCP .

mmsh
MMS (Microsoft Media Server) protocol over HTTP .

The required syntax is:

        mmsh://<server>[:<port>][/<app>][/<playpath>]

md5
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.

pipe
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.

rtmp
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:
server

The address of the RTMP server.

port

The number of the TCP port to use (by default is 1935).

app

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.

playpath

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.

listen

Act as a server, listening for an incoming connection.

timeout

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 "AVOption"s):
rtmp_app

Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI .

rtmp_buffer

Set the client buffer time in milliseconds. The default is 3000.

rtmp_conn

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.

rtmp_flashver

Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2.

rtmp_flush_interval

Number of packets flushed in the same request ( RTMPT only). The default is 10.

rtmp_live

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".

rtmp_pageurl

URL of the web page in which the media was embedded. By default no value will be sent.

rtmp_playpath

Stream identifier to play or to publish. This option overrides the parameter specified in the URI .

rtmp_subscribe

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.

rtmp_swfhash

SHA256 hash of the decompressed SWF file (32 bytes).

rtmp_swfsize

Size of the decompressed SWF file, required for SWFVerification.

rtmp_swfurl

URL of the SWF player for the media. By default no value will be sent.

rtmp_swfverify

URL to player swf file, compute hash/size automatically.

rtmp_tcurl

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

rtmpe
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.

rtmps
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.

rtmpt
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.

rtmpte
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.

rtmpts
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.

rtmp, rtmpe, rtmps, rtmpt, rtmpte
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"

rtp
Real-Time Protocol.

rtsp
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 <http://github.com/revmischa/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 "AVOption"s or in "avformat_open_input"), are supported:

Flags for "rtsp_transport":

	udp		Use UDP as lower transport protocol.
	tcp		Use TCP (interleaving within the RTSP control channel) as lower transport protocol.

udp_multicast

Use UDP multicast as lower transport protocol.

http

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":
filter_src

Accept packets only from negotiated peer address and port.

listen

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>

sap
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.

Muxer

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:
announce_addr=address

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.

announce_port=port

Specify the port to send the announcements on, defaults to 9875 if not specified.

ttl=ttl

Specify the time to live value for the announcements and RTP packets, defaults to 255.

same_port=0|1

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]

Demuxer

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]

tcp
Trasmission Control Protocol.

The required syntax for a TCP url is:

        tcp://<hostname>:<port>[?<options>]

listen

Listen for an incoming connection

timeout=microseconds

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>

tls
Transport Layer Security/Secure Sockets Layer

The required syntax for a TLS/SSL url is:

        tls://<hostname>:<port>[?<options>]

listen

Act as a server, listening for an incoming connection.

cafile=filename

Certificate authority file. The file must be in OpenSSL PEM format.

cert=filename

Certificate file. The file must be in OpenSSL PEM format.

key=filename

Private key file.

verify=0|1

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>

udp
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.
buffer_size=size

Set the UDP socket buffer size in bytes. This is used both for the receiving and the sending buffer size.

localport=port

Override the local UDP port to bind with.

localaddr=addr

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.

pkt_size=size

Set the size in bytes of UDP packets.

reuse=1|0

Explicitly allow or disallow reusing UDP sockets.

ttl=ttl

Set the time to live value (for multicast only).

connect=1|0

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.

sources=address[,address]

Only receive packets sent to the multicast group from one of the specified sender IP addresses.

block=address[,address]

Ignore packets sent to the multicast group from the specified sender IP addresses.

fifo_size=units

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.

overrun_nonfatal=1|0

Survive in case of UDP receiving circular buffer overrun. Default value is 0.

timeout=microseconds

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>

BITSTREAM FILTERS

When you configure your FFmpeg build, all the supported bitstream filters are enabled by default. You can list all available ones using the configure option "−−list−bsfs".

You can disable all the bitstream filters using the configure option "−−disable−bsfs", and selectively enable any bitstream filter using the option "−−enable−bsf=BSF", or you can disable a particular bitstream filter using the option "−−disable−bsf=BSF".

The option "−bsfs" of the ff* tools will display the list of all the supported bitstream filters included in your build.

Below is a description of the currently available bitstream filters.

aac_adtstoasc
chomp
dump_extradata
h264_mp4toannexb
Convert an H.264 bitstream from length prefixed mode to start code prefixed mode (as defined in the Annex B of the ITU-T H.264 specification).

This is required by some streaming formats, typically the MPEG−2 transport stream format ("mpegts").

For example to remux an MP4 file containing an H.264 stream to mpegts format with ffmpeg, you can use the command:

        ffmpeg −i INPUT.mp4 −codec copy −bsf:v h264_mp4toannexb OUTPUT.ts

imx_dump_header
mjpeg2jpeg
Convert MJPEG/AVI1 packets to full JPEG/JFIF packets.

MJPEG is a video codec wherein each video frame is essentially a JPEG image. The individual frames can be extracted without loss, e.g. by

        ffmpeg −i ../some_mjpeg.avi −c:v copy frames_%d.jpg

Unfortunately, these chunks are incomplete JPEG images, because they lack the DHT segment required for decoding. Quoting from <http://www.digitalpreservation.gov/formats/fdd/fdd000063.shtml>:

Avery Lee, writing in the rec.video.desktop newsgroup in 2001, commented that " MJPEG , or at least the MJPEG in AVIs having the MJPG fourcc, is restricted JPEG with a fixed -- and *omitted* -- Huffman table. The JPEG must be YCbCr colorspace, it must be 4:2:2, and it must use basic Huffman encoding, not arithmetic or progressive. . . . You can indeed extract the MJPEG frames and decode them with a regular JPEG decoder, but you have to prepend the DHT segment to them, or else the decoder won’t have any idea how to decompress the data. The exact table necessary is given in the OpenDML spec."

This bitstream filter patches the header of frames extracted from an MJPEG stream (carrying the AVI1 header ID and lacking a DHT segment) to produce fully qualified JPEG images.

        ffmpeg −i mjpeg−movie.avi −c:v copy −bsf:v mjpeg2jpeg frame_%d.jpg
        exiftran −i −9 frame*.jpg
        ffmpeg −i frame_%d.jpg −c:v copy rotated.avi

mjpega_dump_header
movsub
mp3_header_compress
mp3_header_decompress
noise
remove_extradata

FILTERING INTRODUCTION

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 −−−−−−−−−−−−−−−−−−−−−> overlay −−> output
                    |                             ^
                    |                             |
                    +−−−−−> 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], [T2] overlay=0:H/2 [out]; [T1] 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, overlay are in one linear chain, and 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.

GRAPH

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.

FILTERGRAPH DESCRIPTION

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".

Filtergraph syntax
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>]

AUDIO FILTERS

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.

aconvert
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.

•

Convert input to float, planar, stereo:

        aconvert=fltp:stereo

•

Convert input to unsigned 8−bit, automatically select out channel layout:

        aconvert=u8:auto

aformat
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:
sample_fmts

A comma-separated list of requested sample formats.

sample_rates

A comma-separated list of requested sample rates.

channel_layouts

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

amerge
Merge two or more audio streams into a single multi-channel stream.

The filter accepts the following named options:
inputs

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

amix
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:
inputs

Number of inputs. If unspecified, it defaults to 2.

duration

How to determine the end-of-stream.
longest

Duration of longest input. (default)

shortest

Duration of shortest input.

first

Duration of first input.

dropout_transition

Transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.

anull
Pass the audio source unchanged to the output.

aresample
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

asetnsamples
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 ":".
nb_out_samples, n

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.

pad, p

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

ashowinfo
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:

	n		sequential number of the input frame, starting from 0
	pts		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.

pts_time

presentation TimeStamp of the input frame, expressed as a number of seconds

	pos		position of the frame in the input stream, −1 if this information in unavailable and/or meaningless (for example in case of synthetic audio)
	fmt		sample format name

chlayout

channel layout description

nb_samples

number of samples (per each channel) contained in the filtered frame

rate

sample rate for the audio frame

checksum

Adler−32 checksum (printed in hexadecimal) of all the planes of the input frame

plane_checksum

Adler−32 checksum (printed in hexadecimal) for each input frame plane, expressed in the form "[c0 c1 c2 c3 c4 c5 c6 c7]"

asplit
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.

astreamsync
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:
b1 b2

number of buffers forwarded so far on each stream

s1 s2

number of samples forwarded so far on each stream

t1 t2

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

atempo
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

earwax
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.

pan
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

outdef

output channel specification, of the form: "out_name=[gain*]in_name[+[gain*]in_name...]"

out_name

output channel to define, either a channel name ( FL , FR , etc.) or a channel number (c0, c1, etc.)

gain

multiplicative coefficient for the channel, 1 leaving the volume unchanged

in_name

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.

Mixing examples

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.

Remapping examples

The channel remapping will be effective if, and only if:
*<gain coefficients are zeroes or ones,>
*<only one input per channel output,>

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"

silencedetect
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.
duration, d

Set silence duration until notification (default is 2 seconds).

noise, n

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 −

volume
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.

Examples

•

Half the input audio volume:

        volume=0.5

The above example is equivalent to:

        volume=1/2

•

Decrease input audio power by 12 decibels:

        volume=−12dB

volumedetect
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:

	•		The mean square energy is approximately −27 dB, or 10^−2.7.
	•		The largest sample is at −4 dB, or more precisely between −4 dB and −5 dB.
	•		There are 6 samples at −4 dB, 62 at −5 dB, 286 at −6 dB, etc.

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.

asyncts
Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed.

The filter accepts the following named parameters:
compensate

Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence.

min_delta

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.

max_comp

Maximum compensation in samples per second. Relevant only with compensate=1. Default value 500.

first_pts

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.

channelsplit
Split each channel in input audio stream into a separate output stream.

This filter accepts the following named parameters:
channel_layout

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

channelmap
Remap input channels to new locations.

This filter accepts the following named parameters:
channel_layout

Channel layout of the output stream.

map

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
Join multiple input streams into one multi-channel stream.

The filter accepts the following named parameters:
inputs

Number of input streams. Defaults to 2.

channel_layout

Desired output channel layout. Defaults to stereo.

map

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

resample
Convert the audio sample format, sample rate and channel layout. This filter is not meant to be used directly.

AUDIO SOURCES

Below is a description of the currently available audio sources.

abuffer
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
sample_rate

The sample rate of the incoming audio buffers.

sample_fmt

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

channel_layout

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

aevalsrc
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.
channel_layout, c

Set the channel layout. The number of channels in the specified layout must be equal to the number of specified expressions.

duration, d

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.

nb_samples, n

Set the number of samples per channel per each output frame, default to 1024.

sample_rate, s

Specify the sample rate, default to 44100.

Each expression in exprs can contain the following constants:

	n		number of the evaluated sample, starting from 0
	t		time of the evaluated sample expressed in seconds, starting from 0
	s		sample rate

Examples

•

Generate silence:

        aevalsrc=0

•

Generate a sin signal with frequency of 440 Hz, set sample rate to 8000 Hz:

        aevalsrc="sin(440*2*PI*t)::s=8000"

•

Generate a two channels signal, specify the channel layout (Front Center + Back Center) explicitly:

        aevalsrc="sin(420*2*PI*t):cos(430*2*PI*t)::c=FC|BC"

•

Generate white noise:

        aevalsrc="−2+random(0)"

•

Generate an amplitude modulated signal:

        aevalsrc="sin(10*2*PI*t)*sin(880*2*PI*t)"

•

Generate 2.5 Hz binaural beats on a 360 Hz carrier:

        aevalsrc="0.1*sin(2*PI*(360−2.5/2)*t) : 0.1*sin(2*PI*(360+2.5/2)*t)"

anullsrc
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.
sample_rate, s

Specify the sample rate, and defaults to 44100.

channel_layout, cl

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.

nb_samples, n

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

abuffer
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:
time_base

Timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.

sample_rate

Audio sample rate.

sample_fmt

Name of the sample format, as returned by "av_get_sample_fmt_name()".

channel_layout

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.

flite
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.
list_voices

If set to 1, list the names of the available voices and exit immediately. Default value is 0.

nb_samples, n

Set the maximum number of samples per frame. Default value is 512.

textfile

Set the filename containing the text to speak.

text

Set the text to speak.

voice, v

Set the voice to use for the speech synthesis. Default value is "kal". See also the list_voices option.

Examples

•

Read from file speech.txt, and synthetize the text using the standard flite voice:

        flite=textfile=speech.txt

•

Read the specified text selecting the "slt" voice:

        flite=text='So fare thee well, poor devil of a Sub−Sub, whose commentator I am':voice=slt

•

Make ffplay speech the specified text, using "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/>

AUDIO SINKS

Below is a description of the currently available audio sinks.

abuffersink
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.

anullsink
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.

abuffersink
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.

VIDEO FILTERS

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.

alphaextract
Extract the alpha component from the input as a grayscale video. This is especially useful with the alphamerge filter.

alphamerge
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.

ass
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.
original_size

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

bbox
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.

blackdetect
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.
black_min_duration, d

Set the minimum detected black duration expressed in seconds. It must be a non-negative floating point number.

Default value is 2.0.

picture_black_ratio_th, pic_th

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.

pixel_black_th, pix_th

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

blackframe
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.

boxblur
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:
w, h

the input width and height in pixels

cw, ch

the input chroma image width and height in pixels

hsub, vsub

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:

•

Apply a boxblur filter with luma, chroma, and alpha radius set to 2:

        boxblur=2:1

•

Set luma radius to 2, alpha and chroma radius to 0

        boxblur=2:1:0:0:0:0

•

Set luma and chroma radius to a fraction of the video dimension

        boxblur=min(h\,w)/10:1:min(cw\,ch)/10:1

colormatrix
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
Copy the input source unchanged to the output. Mainly useful for testing purposes.

crop
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:
x, y

the computed values for x and y. They are evaluated for each new frame.

in_w, in_h

the input width and height

iw, ih

same as in_w and in_h

out_w, out_h

the output (cropped) width and height

ow, oh

same as out_w and out_h

	a		same as iw / ih
	sar		input sample aspect ratio
	dar		input display aspect ratio, it is the same as (iw / ih) * sar

hsub, vsub

horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.

	n		the number of input frame, starting from 0
	pos		the position in the file of the input frame, NAN if unknown
	t		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)"

cropdetect
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>]]]

limit

Threshold, which can be optionally specified from nothing (0) to everything (255), defaults to 24.

round

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.

reset

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.

decimate
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.

max

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.

hi, lo, frac

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.

delogo
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.
x, y

Specify the top left corner coordinates of the logo. They must be specified.

w, h

Specify the width and height of the logo to clear. They must be specified.

band, t

Specify the thickness of the fuzzy edge of the rectangle (added to w and h). The default value is 4.

show

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.

•

Set a rectangle covering the area with top left corner coordinates 0,0 and size 100x77, setting a band of size 10:

        delogo=0:0:100:77:10

•

As the previous example, but use named options:

        delogo=x=0:y=0:w=100:h=77:band=10

deshake
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.
x, y, w, h

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.

rx, ry

Specify the maximum extent of movement in x and y directions in the range 0−64 pixels. Default 16.

edge

Specify how to generate pixels to fill blanks at the edge of the frame. An integer from 0 to 3 as follows:

	0		Fill zeroes at blank locations
	1		Original image at blank locations
	2		Extruded edge value at blank locations
	3		Mirrored edge at blank locations

The default setting is mirror edge at blank locations.

blocksize

Specify the blocksize to use for motion search. Range 4−128 pixels, default 8.

contrast

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.

search

Specify the search strategy 0 = exhaustive search, 1 = less exhaustive search. Default − exhaustive search.

filename

If set then a detailed log of the motion search is written to the specified file.

drawbox
Draw a colored box on the input image.

It accepts the syntax:

        drawbox=<x>:<y>:<width>:<height>:<color>

x, y

Specify the top left corner coordinates of the box. Default to 0.

width, height

Specify the width and height of the box, if 0 they are interpreted as the input width and height. Default to 0.

color

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"

drawtext
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.

box

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.

boxcolor

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".

draw

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.

fix_bounds

If true, check and fix text coords to avoid clipping.

fontcolor

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".

fontfile

The font file to be used for drawing text. Path must be included. This parameter is mandatory.

fontsize

The font size to be used for drawing text. The default value of fontsize is 16.

ft_load_flags

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
no_scale
no_hinting
render
no_bitmap
vertical_layout
force_autohint
crop_bitmap
pedantic
ignore_global_advance_width
no_recurse
ignore_transform
monochrome
linear_design
no_autohint
end table

Default value is "render".

For more information consult the documentation for the FT_LOAD_* libfreetype flags.

shadowcolor

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".

shadowx, shadowy

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".

tabsize

The size in number of spaces to use for rendering the tab. Default value is 4.

timecode

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.

timecode_rate, rate, r

Set the timecode frame rate (timecode only).

text

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.

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.

x, y

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:

dar

input display aspect ratio, it is the same as (w / h) * sar

hsub, vsub

horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.

line_h, lh

the height of each text line

main_h, h, H

the input height

main_w, w, W

the input width

max_glyph_a, ascent

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.

max_glyph_d, descent

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.

max_glyph_h

maximum glyph height, that is the maximum height for all the glyphs contained in the rendered text, it is equivalent to ascent − descent.

max_glyph_w

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

rand(min, max)

return a random number included between min and max

	sar		input sample aspect ratio
	t		timestamp expressed in seconds, NAN if the input timestamp is unknown

text_h, th

the height of the rendered text

text_w, tw

the width of the rendered text

x, y

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.

•

Draw "Test Text" with font FreeSerif, using the default values for the optional parameters.

        drawtext="fontfile=/usr/share/fonts/truetype/freefont/FreeSerif.ttf: text='Test Text'"

•

Draw ’Test Text’ with font FreeSerif of size 24 at position x=100 and y=50 (counting from the top-left corner of the screen), text is yellow with a red box around it. Both the text and the box have an opacity of 20%.

        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.

•

Show the text at the center of the video frame:

        drawtext="fontsize=30:fontfile=FreeSerif.ttf:text='hello world':x=(w−text_w)/2:y=(h−text_h−line_h)/2"

•

Show a text line sliding from right to left in the last row of the video frame. The file LONG_LINE is assumed to contain a single line with no newlines.

        drawtext="fontsize=15:fontfile=FreeSerif.ttf:text=LONG_LINE:y=h−line_h:x=−50*t"

•

Show the content of file CREDITS off the bottom of the frame and scroll up.

        drawtext="fontsize=20:fontfile=FreeSerif.ttf:textfile=CREDITS:y=h−20*t"

•

Draw a single green letter "g", at the center of the input video. The glyph baseline is placed at half screen height.

        drawtext="fontsize=60:fontfile=FreeSerif.ttf:fontcolor=green:text=g:x=(w−max_glyph_w)/2:y=h/2−ascent"

•

Show text for 1 second every 3 seconds:

        drawtext="fontfile=FreeSerif.ttf:fontcolor=white:x=100:y=x/dar:draw=lt(mod(t\\,3)\\,1):text='blink'"

•

Use fontconfig to set the font. Note that the colons need to be escaped.

        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>.

edgedetect
Detect and draw edges. The filter uses the Canny Edge Detection algorithm.

This filter accepts the following optional named parameters:
low, high

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

fade
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.
type, t

See type.

start_frame, s

See start_frame.

nb_frames, n

See nb_frames.

alpha

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

fieldorder
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:
0 or bff

output bottom field first

1 or tff

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

fifo
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.

format
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

fps
Convert the video to specified constant framerate by duplicating or dropping frames as necessary.

This filter accepts the following named parameters:

fps

Desired output framerate.

framestep
Select one frame every N.

This filter accepts in input a string representing a positive integer. Default argument is 1.

frei0r
Apply a frei0r effect to the input video.

To enable compilation of this filter you need to install the frei0r header and configure FFmpeg with "−−enable−frei0r".

The filter supports the syntax:

        <filter_name>[{:|=}<param1>:<param2>:...:<paramN>]

filter_name is the name to the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched in each one of the directories specified by the colon separated list in FREIOR_PATH , otherwise in the standard frei0r paths, which are in this order: HOME/ .frei0r−1/lib/, /usr/local/lib/frei0r−1/, /usr/lib/frei0r−1/.

param1, param2, ... , paramN specify the parameters for the frei0r effect.

A frei0r effect parameter can be a boolean (whose values are specified with "y" and "n"), a double, a color (specified by the syntax R/G/B, R, G, and B being float numbers from 0.0 to 1.0) or by an "av_parse_color()" color description), a position (specified by the syntax X/Y, X and Y being float numbers) and a string.

The number and kind of parameters depend on the loaded effect. If an effect parameter is not specified the default value is set.

Some examples follow:

•

Apply the distort0r effect, set the first two double parameters:

        frei0r=distort0r:0.5:0.01

•

Apply the colordistance effect, takes a color as first parameter:

        frei0r=colordistance:0.2/0.3/0.4
        frei0r=colordistance:violet
        frei0r=colordistance:0x112233

•

Apply the perspective effect, specify the top left and top right image positions:

        frei0r=perspective:0.2/0.2:0.8/0.2

For more information see: <http://frei0r.dyne.org>

gradfun
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

hflip
Flip the input video horizontally.

For example to horizontally flip the input video with ffmpeg:

        ffmpeg −i in.avi −vf "hflip" out.avi

hqdn3d
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
luma_spatial

a non-negative float number which specifies spatial luma strength, defaults to 4.0

chroma_spatial

a non-negative float number which specifies spatial chroma strength, defaults to 3.0*luma_spatial/4.0

luma_tmp

a float number which specifies luma temporal strength, defaults to 6.0*luma_spatial/4.0

chroma_tmp

a float number which specifies chroma temporal strength, defaults to luma_tmp*chroma_spatial/luma_spatial

hue
Modify the hue and/or the saturation of the input.

This filter accepts the following optional named options:

	h		Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0.
	H		Specify the hue angle as a number of degrees. It accepts a float number or an expression, and defaults to 0.0.
	s		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:

	n		frame count of the input frame starting from 0
	pts		presentation timestamp of the input frame expressed in time base units
	r		frame rate of the input video, NAN if the input frame rate is unknown
	t		timestamp expressed in seconds, NAN if the input timestamp is unknown
	tb		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:

•

Set the hue to 90 degrees and the saturation to 1.0:

        hue=h=90:s=1

•

Same command but expressing the hue in radians:

        hue=H=PI/2:s=1

•

Same command without named options, hue must be expressed in degrees:

        hue=90:1

•

Note that "h:s" syntax does not support expressions for the values of h and s, so the following example will issue an error:

        hue=PI/2:1

•

Rotate hue and make the saturation swing between 0 and 2 over a period of 1 second:

        hue="H=2*PI*t: s=sin(2*PI*t)+1"

•

Apply a 3 seconds saturation fade-in effect starting at 0:

        hue="s=min(t/3\,1)"

The general fade-in expression can be written as:

        hue="s=min(0\, max((t−START)/DURATION\, 1))"

•

Apply a 3 seconds saturation fade-out effect starting at 5 seconds:

        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))"

Commands

This filter supports the following command:
reinit

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.

idet
Interlaceing detect filter. This filter tries to detect if the input is interlaced or progressive. Top or bottom field first.

lut, lutrgb, lutyuv
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:

	c0		first pixel component
	c1		second pixel component
	c2		third pixel component
	c3		fourth pixel component, corresponds to the alpha component

The exact component associated to each option depends on the format in input.

The lutrgb filter requires RGB pixel formats in input, and accepts the options:

	r		red component
	g		green component
	b		blue component
	a		alpha component

The lutyuv filter requires YUV pixel formats in input, and accepts the options:

	y		Y/luminance component
	u		U/Cb component
	v		V/Cr component
	a		alpha component

The expressions can contain the following constants and functions:
w, h

the input width and height

val

input value for the pixel component

clipval

the input value clipped in the minval−maxval range

maxval

maximum value for the pixel component

minval

minimum value for the pixel component

negval

the negated value for the pixel component value clipped in the minval−maxval range , it corresponds to the expression "maxval−clipval+minval"

clip(val)

the computed value in val clipped in the minval−maxval range

gammaval(gamma)

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)

mp
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:
denoise3d
detc
dint
divtc
down3dright
dsize

	eq2
	eq

field

fil

fixpts
fspp

geq

harddup
hqdn3d

ilpack
ivtc
kerndeint
mcdeint
noise

palette
perspective
phase

pp7

pullup

rectangle

sab

softpulldown
softskip

spp

telecine
tile
tinterlace
unsharp
uspp
yuvcsp
yvu9

The parameter syntax and behavior for the listed filters are the same of the corresponding MPlayer filters. For detailed instructions check the " VIDEO FILTERS " section in the MPlayer manual.

Some examples follow:

•

Adjust gamma, brightness, contrast:

        mp=eq2=1.0:2:0.5

•

Add temporal noise to input video:

        mp=noise=20t

See also mplayer(1), <http://www.mplayerhq.hu/>.

negate
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.

noformat
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

null
Pass the video source unchanged to the output.

ocv
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

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

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

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
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_w, main_h

main input width and height

W, H

same as main_w and main_h

overlay_w, overlay_h

overlay input width and height

w, h

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.

rgb

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.

pad
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:
in_w, in_h

the input video width and height

iw, ih

same as in_w and in_h

out_w, out_h

the output width and height, that is the size of the padded area as specified by the width and height expressions

ow, oh

same as out_w and out_h

x, y

x and y offsets as specified by the x and y expressions, or NAN if not yet specified

	a		same as iw / ih
	sar		input sample aspect ratio
	dar		input display aspect ratio, it is the same as (iw / ih) * sar

hsub, vsub

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.
width, height

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.

x, y

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.

color

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".

Examples

•

Add paddings with color "violet" to the input video. Output video size is 640x480, the top-left corner of the input video is placed at column 0, row 40:

        pad=640:480:0:40:violet

•

Pad the input to get an output with dimensions increased by 3/2, and put the input video at the center of the padded area:

        pad="3/2*iw:3/2*ih:(ow−iw)/2:(oh−ih)/2"

•

Pad the input to get a squared output with size equal to the maximum value between the input width and height, and put the input video at the center of the padded area:

        pad="max(iw\,ih):ow:(ow−iw)/2:(oh−ih)/2"

•

Pad the input to get a final w/h ratio of 16:9:

        pad="ih*16/9:ih:(ow−iw)/2:(oh−ih)/2"

•

In case of anamorphic video, in order to set the output display aspect correctly, it is necessary to use sar in the expression, according to the relation:

        (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"

•

Double output size and put the input video in the bottom-right corner of the output padded area:

        pad="2*iw:2*ih:ow−iw:oh−ih"

pixdesctest
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.

removelogo
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
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:
in_w, in_h

the input width and height

iw, ih

same as in_w and in_h

out_w, out_h

the output (cropped) width and height

ow, oh

same as out_w and out_h

	a		same as iw / ih
	sar		input sample aspect ratio
	dar		input display aspect ratio, it is the same as (iw / ih) * sar

hsub, vsub

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:

	1		force interlaced aware scaling
	−1		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
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

selected_n

the sequential number of the selected frame, starting from 0

prev_selected_n

the sequential number of the last selected frame, NAN if undefined

	TB		timebase of the input timestamps
	pts		the PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units, NAN if undefined
	t		the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined

prev_pts

the PTS of the previously filtered video frame, NAN if undefined

prev_selected_pts

the PTS of the last previously filtered video frame, NAN if undefined

prev_selected_t

the PTS of the last previously selected video frame, NAN if undefined

start_pts

the PTS of the first video frame in the video, NAN if undefined

start_t

the time of the first video frame in the video, NAN if undefined

pict_type

the type of the filtered frame, can assume one of the following values:

	I
	P
	B
	S
	SI
	SP
	BI

interlace_type

the frame interlace type, can assume one of the following values:
PROGRESSIVE

the frame is progressive (not interlaced)

TOPFIRST

the frame is top-field-first

BOTTOMFIRST

the frame is bottom-field-first

	key		1 if the filtered frame is a key-frame, 0 otherwise
	pos		the position in the file of the filtered frame, −1 if the information is not available (e.g. for synthetic video)

scene

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.

setdar, setsar
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

setfield
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:
auto

Keep the same field property.

	bff		Mark the frame as bottom-field-first.
	tff		Mark the frame as top-field-first.

prog

Mark the frame as progressive.

showinfo
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:

	n		sequential number of the input frame, starting from 0
	pts		Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.

pts_time

Presentation TimeStamp of the input frame, expressed as a number of seconds

	pos		position of the frame in the input stream, −1 if this information in unavailable and/or meaningless (for example in case of synthetic video)
	fmt		pixel format name
	sar		sample aspect ratio of the input frame, expressed in the form num/den
	s		size of the input frame, expressed in the form widthxheight
	i		interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first)

iskey

1 if the frame is a key frame, 0 otherwise

type

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.

checksum

Adler−32 checksum (printed in hexadecimal) of all the planes of the input frame

plane_checksum

Adler−32 checksum (printed in hexadecimal) of each plane of the input frame, expressed in the form "[c0 c1 c2 c3]"

slicify
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.

smartblur
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
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.

super2xsai
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.

swapuv
Swap U & V plane.

thumbnail
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
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 8x8 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.

tinterlace
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:
merge, 0

Move odd frames into the upper field, even into the lower field, generating a double height frame at half framerate.

drop_odd, 1

Only output even frames, odd frames are dropped, generating a frame with unchanged height at half framerate.

drop_even, 2

Only output odd frames, even frames are dropped, generating a frame with unchanged height at half framerate.

pad, 3

Expand each frame to full height, but pad alternate lines with black, generating a frame with double height at the same input framerate.

interleave_top, 4

Interleave the upper field from odd frames with the lower field from even frames, generating a frame with unchanged height at half framerate.

interleave_bottom, 5

Interleave the lower field from odd frames with the upper field from even frames, generating a frame with unchanged height at half framerate.

interlacex2, 6

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
Transpose rows with columns in the input video and optionally flip it.

It accepts a parameter representing an integer, which can assume the values:
0, 4

Rotate by 90 degrees counterclockwise and vertically flip (default), that is:

        L.R     L.l
        . . −>  . .
        l.r     R.r

1, 5

Rotate by 90 degrees clockwise, that is:

        L.R     l.L
        . . −>  . .
        l.r     r.R

2, 6

Rotate by 90 degrees counterclockwise, that is:

        L.R     R.r
        . . −>  . .
        l.r     L.l

3, 7

Rotate by 90 degrees clockwise and vertically flip, that is:

        L.R     r.R
        . . −>  . .
        l.r     l.L

For values between 4−7 transposition is only done if the input video geometry is portrait and not landscape.

unsharp
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’.
luma_msize_x

Set the luma matrix horizontal size. It can be an integer between 3 and 13, default value is 5.

luma_msize_y

Set the luma matrix vertical size. It can be an integer between 3 and 13, default value is 5.

luma_amount

Set the luma effect strength. It can be a float number between −2.0 and 5.0, default value is 1.0.

chroma_msize_x

Set the chroma matrix horizontal size. It can be an integer between 3 and 13, default value is 5.

chroma_msize_y

Set the chroma matrix vertical size. It can be an integer between 3 and 13, default value is 5.

chroma_amount

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 B<ffmpeg>
        ffmpeg −i in.avi −vf "unsharp" out.mp4

vflip
Flip the input video vertically.

        ffmpeg −i in.avi −vf "vflip" out.avi

yadif
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:

	0		output 1 frame for each frame
	1		output 1 frame for each field
	2		like 0 but skips spatial interlacing check
	3		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:

	0		assume top field first
	1		assume bottom field first
	−1		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

	0		deinterlace all frames
	1		only deinterlace frames marked as interlaced

Default value is 0.

VIDEO SOURCES

Below is a description of the currently available video sources.

buffer
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 descroption of the accepted options follows.
video_size

Specify the size (width and height) of the buffered video frames.

pix_fmt

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.

time_base

Specify the timebase assumed by the timestamps of the buffered frames.

time_base

Specify the frame rate expected for the video stream.

pixel_aspect

Specify the sample aspect ratio assumed by the video frames.

sws_param

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 PixelFormat 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]

cellauto
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.
filename, f

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.

pattern, p

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.

rate, r

Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio

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.

random_seed, seed

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.

rule

Set the cellular automaton rule, it is a number ranging from 0 to 255. Default value is 110.

size, s

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).

scroll

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.

start_full, full

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.

stitch

If set to 1, stitch the left and right row edges together. This is the default behavior, for disabling set the value to 0.

Examples

•

Read the initial state from pattern, and specify an output of size 200x400.

        cellauto=f=pattern:s=200x400

•

Generate a random initial row with a width of 200 cells, with a fill ratio of 2/3:

        cellauto=ratio=2/3:s=200x200

•

Create a pattern generated by rule 18 starting by a single alive cell centered on an initial row with width 100:

        cellauto=p=@s=100x400:full=0:rule=18

•

Specify a more elaborated initial pattern:

        cellauto=p='@@ @ @@':s=100x400:full=0:rule=18

mandelbrot
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.
end_pts

Set the terminal pts value. Default value is 400.

end_scale

Set the terminal scale value. Must be a floating point value. Default value is 0.3.

inner

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:
black

Set black mode.

convergence

Show time until convergence.

mincol

Set color based on point closest to the origin of the iterations.

period

Set period mode.

Default value is mincol.

bailout

Set the bailout value. Default value is 10.0.

maxiter

Set the maximum of iterations performed by the rendering algorithm. Default value is 7189.

outer

Set outer coloring mode. It shall assume one of following values:
iteration_count

Set iteration cound mode.

normalized_iteration_count

set normalized iteration count mode.

Default value is normalized_iteration_count.

rate, r

Set frame rate, expressed as number of frames per second. Default value is "25".

size, s

Set frame size. Default value is "640x480".

start_scale

Set the initial scale value. Default value is 3.0.

start_x

Set the initial x position. Must be a floating point value between −100 and 100. Default value is −0.743643887037158704752191506114774.

start_y

Set the initial y position. Must be a floating point value between −100 and 100. Default value is −0.131825904205311970493132056385139.

mptestsrc
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.
rate, r

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".

duration, d

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.

test, t

Set the number or the name of the test to perform. Supported tests are:
dc_luma
dc_chroma
freq_luma
freq_chroma
amp_luma
amp_chroma

	cbp
	mv

ring1
ring2

all

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.

frei0r_src
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

life
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.
filename, f

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.

rate, r

Set the video rate, that is the number of frames generated per second. Default is 25.

random_fill_ratio, ratio

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.

random_seed, seed

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.

rule

Set the life rule.

A rule can be specified with a code of the kind "S NS /B NB ", 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.

size, s

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).

stitch

If set to 1, stitch the left and right grid edges together, and the top and bottom edges also. Defaults to 1.

mold

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.

life_color

Set the color of living (or new born) cells.

death_color

Set the color of dead cells. If mold is set, this is the first color used to represent a dead cell.

mold_color

Set mold color, for definitely dead and moldy cells.

Examples

•

Read a grid from pattern, and center it on a grid of size 300x300 pixels:

        life=f=pattern:s=300x300

•

Generate a random grid of size 200x200, with a fill ratio of 2/3:

        life=ratio=2/3:s=200x200

•

Specify a custom rule for evolving a randomly generated grid:

        life=rule=S14/B34

•

Full example with slow death effect (mold) using 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

color, nullsrc, rgbtestsrc, smptebars, testsrc
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.
color, c

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".

size, s

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".

rate, r

sar

Set the sample aspect ratio of the sourced video.

duration, d

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.

decimals, n

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

VIDEO SINKS

Below is a description of the currently available video sinks.

buffersink
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.

nullsink
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.

MULTIMEDIA FILTERS

Below is a description of the currently available multimedia filters.

asendcmd, sendcmd
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:
commands, c

Set the commands to be read and sent to the other filters.

filename, f

Set the filename of the commands to be read and sent to the other filters.

Commands syntax

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:
enter

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.

leave

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>]

Examples

•

Specify audio tempo change at second 4:

        asendcmd=c='4.0 atempo tempo 1.5',atempo

•

Specify a list of drawtext and hue commands in a file.

        # 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

asetpts, setpts
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

frame rate, only defined for constant frame-rate video

	PTS		the presentation timestamp in input
	N		the count of the input frame, starting from 0.

NB_CONSUMED_SAMPLES

the number of consumed samples, not including the current frame (only audio)

NB_SAMPLES

the number of samples in the current frame (only audio)

SAMPLE_RATE

audio sample rate

STARTPTS

the PTS of the first frame

STARTT

the time in seconds of the first frame

INTERLACED

tell if the current frame is interlaced

	T		the time in seconds of the current frame
	TB		the time base
	POS		original position in the file of the frame, or undefined if undefined for the current frame

PREV_INPTS

previous input PTS

PREV_INT

previous input time in seconds

PREV_OUTPTS

previous output PTS

PREV_OUTT

previous output time in seconds

Examples

•

Start counting PTS from zero

        setpts=PTS−STARTPTS

•

Apply fast motion effect:

        setpts=0.5*PTS

•

Apply slow motion effect:

        setpts=2.0*PTS

•

Set fixed rate of 25 frames per second:

        setpts=N/(25*TB)

•

Set fixed rate 25 fps with some jitter:

        setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'

•

Apply an offset of 10 seconds to the input PTS:

        setpts=PTS+10/TB

settb, asettb
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".

Examples

•

Set the timebase to 1/25:

        settb=1/25

•

Set the timebase to 1/10:

        settb=0.1

•

Set the timebase to 1001/1000:

        settb=1+0.001

•

Set the timebase to 2*intb:

        settb=2*intb

•

Set the default timebase value:

        settb=AVTB

concat
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:

	n		Set the number of segments. Default is 2.
	v		Set the number of output video streams, that is also the number of video streams in each segment. Default is 1.
	a		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 nx(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:

•

Concatenate an opening, an episode and an ending, all in bilingual version (video in stream 0, audio in streams 1 and 2):

        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

•

Concatenate two parts, handling audio and video separately, using the (a)movie sources, and adjusting the resolution:

        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.

showspectrum
Convert input audio to a video output, representing the audio frequency spectrum.

The filter accepts the following named parameters:
size, s

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.

showwaves
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.

rate, r

Set the (approximate) output frame rate. This is done by setting the option n. Default value is "25".

size, s

Specify the video size for the output. Default value is "600x240".

Some examples follow.

•

Output the input file audio and the corresponding video representation at the same time:

        amovie=a.mp3,asplit[out0],showwaves[out1]

•

Create a synthetic signal and show it with showwaves, forcing a framerate of 30 frames per second:

        aevalsrc=sin(1*2*PI*t)*sin(880*2*PI*t):cos(2*PI*200*t),asplit[out0],showwaves=r=30[out1]

MULTIMEDIA SOURCES

Below is a description of the currently available multimedia sources.

amovie
This is the same as src_movie source, except it selects an audio stream by default.

movie
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.
format_name, f

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.

seek_point, sp

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".

streams, s

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".

stream_index, si

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.

loop

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.

•

Skip 3.2 seconds from the start of the avi file in.avi, and overlay it on top of the input labelled as "in":

        movie=in.avi:seek_point=3.2, scale=180:−1, setpts=PTS−STARTPTS [movie];
        [in] setpts=PTS−STARTPTS, [movie] overlay=16:16 [out]

•

Read from a video4linux2 device, and overlay it on top of the input labelled as "in":

        movie=/dev/video0:f=video4linux2, scale=180:−1, setpts=PTS−STARTPTS [movie];
        [in] setpts=PTS−STARTPTS, [movie] overlay=16:16 [out]

•

Read the first video stream and the audio stream with id 0x81 from dvd.vob; the video is connected to the pad named "video" and the audio is connected to the pad named "audio":

        movie=dvd.vob:s=v:0+#0x81 [video] [audio]

METADATA

FFmpeg is able to dump metadata from media files into a simple UTF−8−encoded INI-like text file and then load it back using the metadata muxer/demuxer.

The file format is as follows:

	1.		A file consists of a header and a number of metadata tags divided into sections, each on its own line.
	2.		The header is a ’;FFMETADATA’ string, followed by a version number (now 1).
	3.		Metadata tags are of the form ’key=value’
	4.		Immediately after header follows global metadata
	5.		After global metadata there may be sections with per−stream/per−chapter metadata.
	6.		A section starts with the section name in uppercase (i.e. STREAM or CHAPTER ) in brackets (’[’, ’]’) and ends with next section or end of file.
	7.		At the beginning of a chapter section there may be an optional timebase to be used for start/end values. It must be in form ’TIMEBASE=num/den’, where num and den are integers. If the timebase is missing then start/end times are assumed to be in milliseconds. Next a chapter section must contain chapter start and end times in form ’START=num’, ’END=num’, where num is a positive integer.
	8.		Empty lines and lines starting with ’;’ or ’#’ are ignored.
	9.		Metadata keys or values containing special characters (’=’, ’;’, ’#’, ’\’ and a newline) must be escaped with a backslash ’\’.
	10.		Note that whitespace in metadata (e.g. foo = bar) is considered to be a part of the tag (in the example above key is ’foo ’, value is ’ bar’).

A ffmetadata file might look like this:

        ;FFMETADATA1
        title=bike\\shed
        ;this is a comment
        artist=FFmpeg troll team
        [CHAPTER]
        TIMEBASE=1/1000
        START=0
        #chapter ends at 0:01:00
        END=60000
        title=chapter \#1
        [STREAM]
        title=multi\
        line

ffmpeg(1)

	•		Size cannot exceed 256 pixels in any dimension
	•		Only BMP and PNG images can be stored
	•		If a BMP image is used, it must be one of the following pixel formats:

	•		If a BMP image is used, it must use the BITMAPINFOHEADER DIB header
	•		If a PNG image is used, it must use the rgba pixel format

GNU/Linux man pages

Livre :Expressions régulières,Syntaxe et mise en oeuvre :

ISBN : 978-2-7460-9712-4 EAN : 9782746097124 (Editions ENI)

GNU/Linux

ffmpeg(1)

FFMPEG

NAME

SYNOPSIS

DESCRIPTION

DETAILED DESCRIPTION

STREAM SELECTION

OPTIONS

TIPS

EXAMPLES

SYNTAX

EXPRESSION EVALUATION

DECODERS

VIDEO DECODERS

AUDIO DECODERS

ENCODERS

AUDIO ENCODERS

VIDEO ENCODERS

DEMUXERS

MUXERS

INPUT DEVICES

OUTPUT DEVICES

PROTOCOLS

BITSTREAM FILTERS

FILTERING INTRODUCTION

GRAPH

FILTERGRAPH DESCRIPTION

AUDIO FILTERS

AUDIO SOURCES

AUDIO SINKS

VIDEO FILTERS

VIDEO SOURCES

VIDEO SINKS

MULTIMEDIA FILTERS

MULTIMEDIA SOURCES

METADATA

Livre :
Expressions régulières,
Syntaxe et mise en oeuvre :

ISBN : 978-2-7460-9712-4
EAN : 9782746097124
(Editions ENI)