avplay - AVplay media player avplay [options] input_file AVplay is a very simple and portable media player using the Libav libraries and the SDL library. It is mostly used as a testbed for the various Libav APIs. All the numerical options, if not specified otherwise, accept in input a string representing a number, which may contain one of the SI unit prefixes, for example 'K', 'M', 'G'. If 'i' is appended after the prefix, binary prefixes are used, which are based on powers of 1024 instead of powers of 1000. The 'B' postfix multiplies the value by 8, and can be appended after a unit prefix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as number 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 specifer, which matches the second audio stream. Therefore it would select the ac3 codec for the second audio stream. A stream specifier can match several stream, the option is then applied to all of them. E.g. the stream specifier in "-b:a 128k" matches all audio streams. An empty stream specifier matches all streams, for example "-codec copy" or "-codec: copy" would copy all the streams without reencoding. Possible forms of stream specifiers are:

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.

i:stream_id

Match the stream by stream id (e.g. PID in MPEG-TS container).

m:key[:value]

Matches streams with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.

Note that in avconv, matching by metadata will only work properly for input files.

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:

-version

Show version.

-formats

Show available formats.

The fields preceding the format names have the following meanings:

-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.

-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:

-cpuflags mask (global)

Set a mask that's applied to autodetected CPU flags. This option is intended for testing. Do not use it unless you know what you're doing.

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:

        avconv -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)

-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 minimum/maximum bitrate. Lowering tolerance too much has an adverse effect on quality.

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

Possible values:

-me_method[:stream_specifier] integer (output,video )

set motion estimation method

Possible values:

-g[:stream_specifier] integer (output,video)

set the group of picture (GOP) 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 )

-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)

minimum video quantizer scale (VBR)

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

maximum video quantizer scale (VBR)

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

maximum difference between the quantizer scales (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

-bug[:stream_specifier] flags (input,video)

work around not autodetected encoder bugs

Possible values:

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

how strictly to follow the standards

Possible values:

-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:

-mpeg_quant[:stream_specifier] integer (output,video )

use MPEG quantizers instead of H.263

-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_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 maximum bitrate tolerance (in bits/s). Requires bufsize to be set.

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

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

-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:

-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:

-ec[:stream_specifier] flags (input,video)

set error concealment strategy

Possible values:

-pred[:stream_specifier] integer (output,video)

prediction method

Possible values:

-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:

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

visualize motion vectors (MVs)

Possible values:

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

full-pel ME compare function

Possible values:

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

sub-pel ME compare function

Possible values:

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

macroblock compare function

Possible values:

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

interlaced DCT compare function

Possible values:

-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:

-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

-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

-global_quality[:stream_specifier] integer (output,audio,video )

-coder[:stream_specifier] integer (output,video )

Possible values:

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

context model

-mbd[:stream_specifier] integer (output,video)

macroblock decision algorithm (high quality mode)

Possible values:

-sc_threshold[:stream_specifier] integer (output,video )

scene change threshold

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

minimum Lagrange factor (VBR)

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

maximum 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

-flags2[:stream_specifier] flags (input/output,audio,video )

Possible values:

-error[:stream_specifier] integer (output,video )

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

Possible values:

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

motion estimation 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:

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

Possible values:

-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:

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

increase the quantizer for macroblocks close to borders

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

minimum macroblock Lagrange factor (VBR)

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

maximum 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:

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

Possible values:

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

Possible values:

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

refine the two motion vectors used in bidirectional macroblocks

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

downscale frames for dynamic B-frame decision

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

minimum interval between IDR-frames (x264)

-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 )

adjust 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

-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 primaries

Possible values:

-color_trc[:stream_specifier] integer (input/output,video )

color transfert characteristic

Possible values:

-colorspace[:stream_specifier] integer (input/output,video )

colorspace

Possible values:

-color_range[:stream_specifier] integer (input/output,video )

color range

Possible values:

-chroma_sample_location[:stream_specifier] integer (input/output,video)

-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:

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

audio service type

Possible values:

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

Possible values:

-refcounted_frames[:stream_specifier] integer (input,audio,video )

-side_data_only_packets[:stream_specifier] integer (output,audio,video)

Format AVOptions¶

-probesize integer (input)

set probing size

-packetsize integer (output)

set packet size

-fflags flags (input/output)

Possible values:

-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

-f_err_detect flags (input)

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

Possible values:

-err_detect flags (input)

set error detection flags

Possible values:

-max_interleave_delta integer (output)

maximum buffering duration for interleaving

-f_strict integer (input/output)

how strictly to follow the standards (deprecated; use strict, save via avconv)

Possible values:

-strict integer (input/output)

how strictly to follow the standards

Possible values:

Main options¶

-x width

Force displayed width.

-y height

Force displayed height.

-s size

This option has been removed. Use private format options for specifying the input video size. For example with the rawvideo demuxer you need to specify the option video_size.

-an

Disable audio.

-vn

Disable video.

-ss pos

Seek to a given position in seconds.

-t duration

play <duration> seconds of audio/video

-bytes

Seek by bytes.

-nodisp

Disable graphical display.

-f fmt

Force format.

-window_title title

Set window title (default is the input filename).

-loop number

Loops movie playback <number> times. 0 means forever.

-vf filter_graph

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

Advanced options¶

-pix_fmt format

This option has been removed. Use private options for specifying the input pixel format. For example with the rawvideo demuxer you need to specify the option pixel_format.

-stats

Show the stream duration, the codec parameters, the current position in the stream and the audio/video synchronisation drift.

-bug

Work around bugs.

-fast

Non-spec-compliant optimizations.

-genpts

Generate pts.

-rtp_tcp

Force RTP/TCP protocol usage instead of RTP/UDP. It is only meaningful if you are streaming with the RTSP protocol.

-sync type

Set the master clock to audio ("type=audio"), video ("type=video") or external ("type=ext"). Default is audio. The master clock is used to control audio-video synchronization. Most media players use audio as master clock, but in some cases (streaming or high quality broadcast) it is necessary to change that. This option is mainly used for debugging purposes.

-threads count

Set the thread count.

-ast audio_stream_number

Select the desired audio stream number, counting from 0. The number refers to the list of all the input audio streams. If it is greater than the number of audio streams minus one, then the last one is selected, if it is negative the audio playback is disabled.

-vst video_stream_number

Select the desired video stream number, counting from 0. The number refers to the list of all the input video streams. If it is greater than the number of video streams minus one, then the last one is selected, if it is negative the video playback is disabled.

-sst subtitle_stream_number

Select the desired subtitle stream number, counting from 0. The number refers to the list of all the input subtitle streams. If it is greater than the number of subtitle streams minus one, then the last one is selected, if it is negative the subtitle rendering is disabled.

-autoexit

Exit when video is done playing.

-exitonkeydown

Exit if any key is pressed.

-exitonmousedown

Exit if any mouse button is pressed.

While playing¶

q, ESC

Quit.

f

Toggle full screen.

p, SPC

Pause.

a

Cycle audio channel.

v

Cycle video channel.

t

Cycle subtitle channel.

w

Show audio waves.

left/right

Seek backward/forward 10 seconds.

down/up

Seek backward/forward 1 minute.

PGDOWN/PGUP

Seek to the previous/next chapter.

mouse click

Seek to percentage in file corresponding to fraction of width.

When evaluating an arithmetic expression, Libav 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)

cosh(x)

tanh(x)

sin(x)

cos(x)

tan(x)

atan(x)

asin(x)

acos(x)

exp(x)

log(x)

abs(x)

squish(x)

gauss(x)

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)

max(x, y)

min(x, y)

eq(x, y)

gte(x, y)

gt(x, y)

lte(x, y)

lt(x, y)

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.

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.

Note that: "*" works like AND "+" works like OR thus

        if A then B else C

is equivalent to

        A*B + not(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 are configured elements in Libav which allow the decoding of multimedia streams. When you configure your Libav 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 "-decoders" of the av* tools will display the list of enabled decoders. A description of some of the currently available audio decoders follows.

ac3¶

AC-3 audio decoder. This decoder implements part of ATSC A/52:2010 and ETSI TS 102 366, as well as the undocumented RealAudio 3 (a.k.a. dnet). AC-3 Decoder Options

-drc_scale value

Dynamic Range Scale Factor. The factor to apply to dynamic range values from the AC-3 stream. This factor is applied exponentially. There are 3 notable scale factor ranges:

Demuxers are configured elements in Libav which allow to read the multimedia streams from a particular type of file. When you configure your Libav 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 av* 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 pattern may contain the string "%d" or "%0 Nd", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0 Nd" is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character '%' can be specified in the pattern with the string "%%". If the pattern contains "%d" or "%0 Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between 0 and 4, all the following numbers must be sequential. This limitation may be hopefully fixed. The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files. For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc.; the pattern "i%%m%%g-%d.jpg" will match a sequence of filenames of the form i%m%g-1.jpg, i%m%g-2.jpg, ..., i%m%g-10.jpg, etc. The size, the pixel format, and the format of each image must be the same for all the files in the sequence. The following example shows how to use avconv for creating a video from the images in the file sequence img-001.jpeg, img-002.jpeg, ..., assuming an input framerate of 10 frames per second:

        avconv -i 'img-%03d.jpeg' -r 10 out.mkv

Note that the pattern must not necessarily contain "%d" or "%0 Nd", for example to convert a single image file img.jpeg you can employ the command:

        avconv -i img.jpeg img.png

-pixel_format format

Set the pixel format (for raw image)

-video_size size

Set the frame size (for raw image)

-framerate rate

Set the frame rate

-loop bool

Loop over the images

-start_number start

Specify the first number in the sequence

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 avplay), 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".

flv¶

Adobe Flash Video Format demuxer. This demuxer is used to demux FLV files and RTMP network streams.

-flv_metadata bool

Allocate the streams according to the onMetaData array content.

asf¶

Advanced Systems Format demuxer. This demuxer is used to demux ASF files and MMS network streams.

-no_resync_search bool

Do not try to resynchronize by looking for a certain optional start code.

Muxers are configured elements in Libav which allow writing multimedia streams to a particular type of file. When you configure your Libav 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 av* tools will display the list of enabled muxers. A description of some of the currently available muxers follows.

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:

        avconv -i INPUT -f crc out.crc

You can print the CRC to stdout with the command:

        avconv -i INPUT -f crc -

You can select the output format of each frame with avconv 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:

        avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc -

See also the framecrc muxer.

framecrc¶

Per-frame CRC (Cyclic Redundancy Check) testing format. This muxer computes and prints the Adler-32 CRC for each decoded audio and video frame. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC. The output of the muxer consists of a line for each audio and video frame of the form: stream_index, frame_dts, frame_size, 0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the decoded frame. For example to compute the CRC of each decoded frame in the input, and store it in the file out.crc:

        avconv -i INPUT -f framecrc out.crc

You can print the CRC of each decoded frame to stdout with the command:

        avconv -i INPUT -f framecrc -

You can select the output format of each frame with avconv by specifying the audio and video codec and format. For example, to compute the CRC of each decoded input audio frame converted to PCM unsigned 8-bit and of each decoded input video frame converted to MPEG-2 video, use the command:

        avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc -

See also the crc muxer.

hls¶

Apple HTTP Live Streaming muxer that segments MPEG-TS according to the HTTP Live Streaming specification. It creates a playlist file and numbered segment files. The output filename specifies the playlist filename; the segment filenames receive the same basename as the playlist, a sequential number and a .ts extension.

        avconv -i in.nut out.m3u8

-hls_time seconds

Set the segment length in seconds.

-hls_list_size size

Set the maximum number of playlist entries.

-hls_wrap wrap

Set the number after which index wraps.

-start_number number

Start the sequence from number.

-hls_base_url baseurl

Append baseurl to every entry in the playlist. Useful to generate playlists with absolute paths.

image2¶

Image file muxer. The image file muxer writes video frames to image files. The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0 Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0 Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character '%' can be specified in the pattern with the string "%%". If the pattern contains "%d" or "%0 Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential. The pattern may contain a suffix which is used to automatically determine the format of the image files to write. For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form img-001.bmp, img-002.bmp, ..., img-010.bmp, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form img%-1.jpg, img%-2.jpg, ..., img%-10.jpg, etc. The following example shows how to use avconv for creating a sequence of files img-001.jpeg, img-002.jpeg, ..., taking one image every second from the input video:

        avconv -i in.avi -vsync 1 -r 1 -f image2 'img-%03d.jpeg'

Note that with avconv, if the format is not specified with the "-f" option and the output filename specifies an image file format, the image2 muxer is automatically selected, so the previous command can be written as:

        avconv -i in.avi -vsync 1 -r 1 'img-%03d.jpeg'

Note also that the pattern must not necessarily contain "%d" or "%0 Nd", for example to create a single image file img.jpeg from the input video you can employ the command:

        avconv -i in.avi -f image2 -frames:v 1 img.jpeg

-start_number number

Start the sequence from number.

-update number

If number is nonzero, the filename will always be interpreted as just a filename, not a pattern, and this file will be continuously overwritten with new images.

matroska¶

Matroska container muxer. This muxer implements the matroska and webm container specs. The recognized metadata settings in this muxer are:

title=title name

Name provided to a single track

language=language name

Specifies the language of the track in the Matroska languages form

STEREO_MODE=mode

Stereo 3D video layout of two views in a single video track

For example a 3D WebM clip can be created using the following command line:

        avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm

This muxer supports the following options:

reserve_index_space

By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases -- e.g. streaming where seeking is possible but slow -- it is useful to put the index at the beginning of the file.

If this option is set to a non-zero value, the muxer will reserve a given amount of space in the file header and then try to write the cues there when the muxing finishes. If the available space does not suffice, muxing will fail. A safe size for most use cases should be about 50kB per hour of video.

Note that cues are only written if the output is seekable and this option will have no effect if it is not.

mov, mp4, ismv¶

The mov/mp4/ismv muxer supports fragmentation. Normally, a MOV/MP4 file has all the metadata about all packets stored in one location (written at the end of the file, it can be moved to the start for better playback using the qt-faststart tool). A fragmented file consists of a number of fragments, where packets and metadata about these packets are stored together. Writing a fragmented file has the advantage that the file is decodable even if the writing is interrupted (while a normal MOV/MP4 is undecodable if it is not properly finished), and it requires less memory when writing very long files (since writing normal MOV/MP4 files stores info about every single packet in memory until the file is closed). The downside is that it is less compatible with other applications. Fragmentation is enabled by setting one of the AVOptions that define how to cut the file into fragments:

-movflags frag_keyframe

Start a new fragment at each video keyframe.

-frag_duration duration

Create fragments that are duration microseconds long.

-frag_size size

Create fragments that contain up to size bytes of payload data.

-movflags frag_custom

Allow the caller to manually choose when to cut fragments, by calling "av_write_frame(ctx, NULL)" to write a fragment with the packets written so far. (This is only useful with other applications integrating libavformat, not from avconv.)

-min_frag_duration duration

Don't create fragments that are shorter than duration microseconds long.

If more than one condition is specified, fragments are cut when one of the specified conditions is fulfilled. The exception to this is "-min_frag_duration", which has to be fulfilled for any of the other conditions to apply. Additionally, the way the output file is written can be adjusted through a few other options:

-movflags empty_moov

Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration.

Files written with this option set do not work in QuickTime. This option is implicitly set when writing ismv (Smooth Streaming) files.

-movflags separate_moof

Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.

This option is implicitly set when writing ismv (Smooth Streaming) files.

-movflags faststart

Run a second pass moving the index (moov atom) to the beginning of the file. This operation can take a while, and will not work in various situations such as fragmented output, thus it is not enabled by default.

-movflags disable_chpl

Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can cause failures when the file is reprocessed with certain tagging programs.

Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example:

        avconv -re <<normal input/transcoding options>> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1)

mp3¶

The MP3 muxer writes a raw MP3 stream with an ID3v2 header at the beginning and optionally an ID3v1 tag at the end. ID3v2.3 and ID3v2.4 are supported, the "id3v2_version" option controls which one is used. Setting "id3v2_version" to 0 will disable the ID3v2 header completely. The legacy ID3v1 tag is not written by default, but may be enabled with the "write_id3v1" option. The muxer may also write a Xing frame at the beginning, which contains the number of frames in the file. It is useful for computing duration of VBR files. The Xing frame is written if the output stream is seekable and if the "write_xing" option is set to 1 (the default). The muxer supports writing ID3v2 attached pictures (APIC frames). The pictures are supplied to the muxer in form of a video stream with a single packet. There can be any number of those streams, each will correspond to a single APIC frame. The stream metadata tags title and comment map to APIC description and picture type respectively. See < http://id3.org/id3v2.4.0-frames> for allowed picture types. Note that the APIC frames must be written at the beginning, so the muxer will buffer the audio frames until it gets all the pictures. It is therefore advised to provide the pictures as soon as possible to avoid excessive buffering. Examples: Write an mp3 with an ID3v2.3 header and an ID3v1 footer:

        avconv -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3

Attach a picture to an mp3:

        avconv -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover"
        -metadata:s:v comment="Cover (Front)" out.mp3

Write a "clean" MP3 without any extra features:

        avconv -i input.wav -write_xing 0 -id3v2_version 0 out.mp3

mpegts¶

MPEG transport stream muxer. This muxer implements ISO 13818-1 and part of ETSI EN 300 468. The muxer options are:

-mpegts_original_network_id number

Set the original_network_id (default 0x0001). This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path Original_Network_ID, Transport_Stream_ID.

-mpegts_transport_stream_id number

Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB.

-mpegts_service_id number

Set the service_id (default 0x0001) also known as program in DVB.

-mpegts_pmt_start_pid number

Set the first PID for PMT (default 0x1000, max 0x1f00).

-mpegts_start_pid number

Set the first PID for data packets (default 0x0100, max 0x0f00).

-muxrate number

Set a constant muxrate (default VBR).

-pcr_period numer

Override the default PCR retransmission time (default 20ms), ignored if variable muxrate is selected.

The recognized metadata settings in mpegts muxer are "service_provider" and "service_name". If they are not set the default for "service_provider" is "Libav" and the default for "service_name" is "Service01".

        avconv -i file.mpg -c copy \
             -mpegts_original_network_id 0x1122 \
             -mpegts_transport_stream_id 0x3344 \
             -mpegts_service_id 0x5566 \
             -mpegts_pmt_start_pid 0x1500 \
             -mpegts_start_pid 0x150 \
             -metadata service_provider="Some provider" \
             -metadata service_name="Some Channel" \
             -y out.ts

null¶

Null muxer. This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes. For example to benchmark decoding with avconv you can use the command:

        avconv -benchmark -i INPUT -f null out.null

Note that the above command does not read or write the out.null file, but specifying the output file is required by the avconv syntax. Alternatively you can write the command as:

        avconv -benchmark -i INPUT -f null -

nut¶

-syncpoints flags

Change the syncpoint usage in nut:

        avconv -i INPUT -f_strict experimental -syncpoints none - | processor

ogg¶

Ogg container muxer.

-page_duration duration

Preferred page duration, in microseconds. The muxer will attempt to create pages that are approximately duration microseconds long. This allows the user to compromise between seek granularity and container overhead. The default is 1 second. A value of 0 will fill all segments, making pages as large as possible. A value of 1 will effectively use 1 packet-per-page in most situations, giving a small seek granularity at the cost of additional container overhead.

segment¶

Basic stream segmenter. The segmenter muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to image2. Every segment starts with a video keyframe, if a video stream is present. The segment muxer works best with a single constant frame rate video. Optionally it can generate a flat list of the created segments, one segment per line.

segment_format format

Override the inner container format, by default it is guessed by the filename extension.

segment_time t

Set segment duration to t seconds.

segment_list name

Generate also a listfile named name.

segment_list_type type

Select the listing format.

segment_list_size size

Overwrite the listfile once it reaches size entries.

segment_list_entry_prefix prefix

Prepend prefix to each entry. Useful to generate absolute paths.

segment_wrap limit

Wrap around segment index once it reaches limit.

        avconv -i in.mkv -c copy -map 0 -f segment -list out.list out%03d.nut

Input devices are configured elements in Libav which allow to access the data coming from a multimedia device attached to your system. When you configure your Libav 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 av* 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 avconv from an ALSA device with card id 0, you may run the command:

        avconv -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.

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 avconv:

        avconv -f fbdev -r 10 -i /dev/fb0 out.avi

You can take a single screenshot image with the command:

        avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg

See also < http://linux-fbdev.sourceforge.net/>, and fbset(1).

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 Libav 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 avconv.

        # Create a JACK writable client with name "libav".
        $ avconv -f jack -i libav -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
        libav:input_1
        metro:120_bpm
        
        # Connect metro to the avconv writable client.
        $ jack_connect metro:120_bpm libav:input_1

For more information read: < http://jackaudio.org/>

libdc1394¶

IIDC1394 input device, based on libdc1394 and libraw1394.

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 avconv use the command:

        avconv -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.

        avconv -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 "libav" 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 avconv use the command:

        avconv -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 avconv and avplay:

        # Grab and show the input of a video4linux2 device.
        avplay -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.
        avconv -f video4linux2 -input_format mjpeg -i /dev/video0 out.mpeg

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 avconv:

        avconv -f x11grab -r 25 -s cif -i :0.0 out.mpg
        
        # Grab at position 10,20.
        avconv -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg

follow_mouse AVOption The syntax is:

        -follow_mouse centered|<PIXELS>

When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within PIXELS (greater than zero) to the edge of region. For example:

        avconv -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg
        
        # Follows only when the mouse pointer reaches within 100 pixels to edge
        avconv -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg

show_region AVOption The syntax is:

        -show_region 1

If show_region AVOption is specified with 1, then the grabbing region will be indicated on screen. With this option, it's easy to know what is being grabbed if only a portion of the screen is grabbed. For example:

        avconv -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg
        
        # With follow_mouse
        avconv -f x11grab -follow_mouse centered -show_region 1  -r 25 -s cif -i :0.0 out.mpg

Output devices are configured elements in Libav which allow to write multimedia data to an output device attached to your system. When you configure your Libav 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 av* 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.

oss¶

OSS (Open Sound System) output device.

sndio¶

sndio audio output device. Protocols are configured elements in Libav which allow to access resources which require the use of a particular protocol. When you configure your Libav 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 av* tools will display the list of supported protocols. A description of the currently available protocols follows.

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 avplay use the command:

        avplay 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 avconv use the command:

        avconv -i file:input.mpeg output.mpeg

The av* 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). This protocol accepts the following options:

chunked_post

If set to 1 use chunked Transfer-Encoding for posts, default is 1.

content_type

Set a specific content type for the POST messages.

headers

Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.

multiple_requests

Use persistent connections if set to 1, default is 0.

post_data

Set custom HTTP post data.

user_agent

Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.

mime_type

Export the MIME type.

icy

If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the icy_metadata_headers and icy_metadata_packet options. The default is 1.

icy_metadata_headers

If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.

icy_metadata_packet

If the server supports ICY metadata, and icy was set to 1, this contains the last non-empty metadata packet sent by the server. It should be polled in regular intervals by applications interested in mid-stream metadata updates.

offset

Set initial byte offset.

end_offset

Try to limit the request to bytes preceding this offset.

Icecast¶

Icecast (stream to Icecast servers) This protocol accepts the following options:

ice_genre

Set the stream genre.

ice_name

Set the stream name.

ice_description

Set the stream description.

ice_url

Set the stream website URL.

ice_public

Set if the stream should be public or not. The default is 0 (not public).

user_agent

Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.

password

Set the Icecast mountpoint password.

content_type

Set the stream content type. This must be set if it is different from audio/mpeg.

legacy_icecast

This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.

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.
        avconv -i input.flv -f avi -y md5:output.avi.md5
        
        # Write the MD5 hash of the encoded AVI file to stdout.
        avconv -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 avconv:

        cat test.wav | avconv -i pipe:0
        # ...this is the same as...
        cat test.wav | avconv -i pipe:

For writing to stdout with avconv:

        avconv -i test.wav -f avi pipe:1 | cat > test.avi
        # ...this is the same as...
        avconv -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://[<username>:<password>@]<server>[:<port>][/<app>][/<instance>][/<playpath>]

The accepted parameters are:

username

An optional username (mostly for publishing).

password

An optional password (mostly for publishing).

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. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)

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 avplay a multimedia resource named "sample" from the application "vod" from an RTMP server "myserver":

        avplay rtmp://myserver/vod/sample

To publish to a password protected server, passing the playpath and app names separately:

        avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/

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.

librtmp 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 avconv:

        avconv -re -i myfile -f flv rtmp://myserver/live/mystream

To play the same stream using avplay:

        avplay "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
RTSP server ("http://github.com/revmischa/rtsp-server")). The required syntax for a RTSP url is:

        rtsp://<hostname>[:<port>]/<path>

The following options (set on the avconv/avplay 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 avplay, 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:

        avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4

To watch a stream tunneled over HTTP:

        avplay -rtsp_transport http rtsp://server/video.mp4

To send a stream in realtime to a RTSP server, for others to watch:

        avconv -re -i <input> -f rtsp -muxdelay 0.1 rtsp://server/live.sdp

To receive a stream in realtime:

        avconv -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:

        avconv -re -i <input> -f sap sap://224.0.0.255?same_port=1

Similarly, for watching in avplay:

        avconv -re -i <input> -f sap sap://224.0.0.255

And for watching in avplay, over IPv6:

        avconv -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:

        avplay sap://

To play back the first stream announced on one the default IPv6 SAP multicast address:

        avplay 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

        avconv -i <input> -f <format> tcp://<hostname>:<port>?listen
        avplay tcp://<hostname>:<port>
    

tls¶

Transport Layer Security (TLS) / Secure Sockets Layer (SSL) The required syntax for a TLS url is:

        tls://<hostname>:<port>

The following parameters can be set via command line options (or in code via "AVOption"s):

ca_file

A file containing certificate authority (CA) root certificates to treat as trusted. If the linked TLS library contains a default this might not need to be specified for verification to work, but not all libraries and setups have defaults built in.

tls_verify=1|0

If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With GnuTLS, the host name is validated as well.)

This is disabled by default since it requires a CA database to be provided by the caller in many cases.

cert_file

A file containing a certificate to use in the handshake with the peer. (When operating as server, in listen mode, this is more often required by the peer, while client certificates only are mandated in certain setups.)

key_file

A file containing the private key for the certificate.

listen=1|0

If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.

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. Follow the list of supported options.

buffer_size=size

set the UDP buffer size in bytes

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.

Some usage examples of the udp protocol with avconv follow. To stream over UDP to a remote endpoint:

        avconv -i <input> -f <format> udp://<hostname>:<port>

To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:

        avconv -i <input> -f mpegts udp://<hostname>:<port>?pkt_size=188&buffer_size=65535

To receive over UDP from a remote endpoint:

        avconv -i udp://[<multicast-address>]:<port>

unix¶

Unix local socket The required syntax for a Unix socket URL is:

        unix://<filepath>

The following parameters can be set via command line options (or in code via "AVOption"s):

timeout

Timeout in ms.

listen

Create the Unix socket in listening mode.

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 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", and a filter with no output pads is called a "sink".

Filtergraph syntax¶

A filtergraph has a textual representation, which is recognized by the -filter/ -vf and -filter_complex options in avconv and -vf in avplay, and by the "avfilter_graph_parse()"/"avfilter_graph_parse2()" functions defined in libavfilter/avfilter.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. It may have one of two forms:

•

A ':'-separated list of key=value pairs.

•

A ':'-separated list of value. In this case, the keys are assumed to be the option names in the order they are declared. E.g. the "fade" filter declares three options in this order -- type, start_frame and nb_frames. Then the parameter list in:0:30 means that the value in is assigned to the option type, 0 to start_frame and 30 to nb_frames.

If the option value itself is a list of items (e.g. the "format" filter takes a list of pixel formats), the items in the list are usually separated by '|'. 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. Here is a BNF description of the filtergraph syntax:

        <NAME>             ::= sequence of alphanumeric characters and '_'
        <LINKLABEL>        ::= "[" <NAME> "]"
        <LINKLABELS>       ::= <LINKLABEL> [<LINKLABELS>]
        <FILTER_ARGUMENTS> ::= sequence of chars (possibly quoted)
        <FILTER>           ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>]
        <FILTERCHAIN>      ::= <FILTER> [,<FILTERCHAIN>]
        <FILTERGRAPH>      ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]

When you configure your Libav 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.

aformat¶

Convert the input audio to one of the specified formats. The framework will negotiate the most appropriate format to minimize conversions. It accepts the following parameters:

sample_fmts

A '|'-separated list of requested sample formats.

sample_rates

A '|'-separated list of requested sample rates.

channel_layouts

A '|'-separated list of requested channel layouts.

If a parameter is omitted, all values are allowed. Force the output to either unsigned 8-bit or signed 16-bit stereo

        aformat=sample_fmts=u8|s16:channel_layouts=stereo

amix¶

Mixes multiple audio inputs into a single output. For example

        avconv -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. It accepts the following parameters:

inputs

The number of inputs. If unspecified, it defaults to 2.

duration

How to determine the end-of-stream.

dropout_transition

The 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.

asetpts¶

Change the PTS (presentation timestamp) of the input audio frames. It accepts the following parameters:

expr

The expression which is evaluated for each frame to construct its timestamp.

The expression is evaluated through the eval API and can contain the following constants:

PTS

the presentation timestamp in input

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

N

The number of audio samples passed through the filter so far, starting at 0.

S

The number of audio samples in the current frame.

SR

The audio sample rate.

STARTPTS

The PTS of the first frame.

PREV_INPTS

The previous input PTS.

PREV_OUTPTS

The previous output PTS.

RTCTIME

The wallclock (RTC) time in microseconds.

RTCSTART

The wallclock (RTC) time at the start of the movie in microseconds.

Some examples:

        # Start counting PTS from zero
        asetpts=expr=PTS-STARTPTS
        
        # Generate timestamps by counting samples
        asetpts=expr=N/SR/TB
        
        # Generate timestamps from a "live source" and rebase onto the current timebase
        asetpts='(RTCTIME - RTCSTART) / (TB * 1000000)"

asettb¶

Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration. This filter accepts the following parameters:

expr

The expression which is evaluated into the output timebase.

The expression can contain the constants PI, E, PHI, AVTB (the default timebase), intb (the input timebase), and sr (the sample rate, audio only). The default value for the input is intb. Some examples:

        # Set the timebase to 1/25:
        settb=1/25
        
        # Set the timebase to 1/10:
        settb=0.1
        
        # Set the timebase to 1001/1000:
        settb=1+0.001
        
        # Set the timebase to 2*intb:
        settb=2*intb
        
        # Set the default timebase value:
        settb=AVTB
        
        # Set the timebase to twice the sample rate:
        asettb=sr*2

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. It accepts the following parameters:

n

The (sequential) number of the input frame, starting from 0.

pts

The presentation timestamp of the input frame, in time base units; the time base depends on the filter input pad, and is usually 1/ sample_rate.

pts_time

The presentation timestamp of the input frame in seconds.

fmt

The sample format.

chlayout

The channel layout.

rate

The sample rate for the audio frame.

nb_samples

The number of samples (per channel) in the frame.

checksum

The Adler-32 checksum (printed in hexadecimal) of the audio data. For planar audio, the data is treated as if all the planes were concatenated.

plane_checksums

A list of Adler-32 checksums for each data plane.

asplit¶

Split input audio into several identical outputs. It accepts a single parameter, which specifies the number of outputs. If unspecified, it defaults to 2. For example,

        avconv -i INPUT -filter_complex asplit=5 OUTPUT

will create 5 copies of the input audio.

asyncts¶

Synchronize audio data with timestamps by squeezing/stretching it and/or dropping samples/adding silence when needed. It accepts the following 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

The minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. The default value is 0.1. If you get an imperfect sync with this filter, try setting this parameter to 0.

max_comp

The maximum compensation in samples per second. Only relevant with compensate=1. The default value is 500.

first_pts

Assume that the first PTS should be this value. The time base is 1 / sample rate. This allows for padding/trimming at the start of the 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 or to trim any samples with a negative PTS due to encoder delay.

atrim¶

Trim the input so that the output contains one continuous subpart of the input. It accepts the following parameters:

start

Timestamp (in seconds) of the start of the section to keep. I.e. the audio sample with the timestamp start will be the first sample in the output.

end

Timestamp (in seconds) of the first audio sample that will be dropped. I.e. the audio sample immediately preceding the one with the timestamp end will be the last sample in the output.

start_pts

Same as start, except this option sets the start timestamp in samples instead of seconds.

end_pts

Same as end, except this option sets the end timestamp in samples instead of seconds.

duration

The maximum duration of the output in seconds.

start_sample

The number of the first sample that should be output.

end_sample

The number of the first sample that should be dropped.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _sample options simply count the samples that pass through the filter. So start/end_pts and start/end_sample will give different results when the timestamps are wrong, inexact or do not start at zero. Also note that this filter does not modify the timestamps. If you wish to have the output timestamps start at zero, insert the asetpts filter after the atrim filter. If multiple start or end options are set, this filter tries to be greedy and keep all samples that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple atrim filters. The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time. Examples:

•

Drop everything except the second minute of input:

        avconv -i INPUT -af atrim=60:120
    

•

Keep only the first 1000 samples:

        avconv -i INPUT -af atrim=end_sample=1000
    

bs2b¶

Bauer stereo to binaural transformation, which improves headphone listening of stereo audio records. It accepts the following parameters:

profile

Pre-defined crossfeed level.

fcut

Cut frequency (in Hz).

feed

Feed level (in Hz).

channelsplit¶

Split each channel from an input audio stream into a separate output stream. It accepts the following parameters:

channel_layout

The channel layout of the input stream. The default is "stereo".

For example, assuming a stereo input MP3 file,

        avconv -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. Split a 5.1 WAV file into per-channel files:

        avconv -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. It accepts the following parameters:

channel_layout

The channel layout of the output stream.

map

Map channels from input to output. The argument is a '|'-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 indices. For example, assuming a 5.1+downmix input MOV file,

        avconv -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

        avconv -i in.wav -filter 'channelmap=1|2|0|5|3|4:channel_layout=5.1' out.wav

compand¶

Compress or expand the audio's dynamic range. It accepts the following parameters:

attacks

decays

A list of times in seconds for each channel over which the instantaneous level of the input signal is averaged to determine its volume. attacks refers to increase of volume and decays refers to decrease of volume. For most situations, the attack time (response to the audio getting louder) should be shorter than the decay time, because the human ear is more sensitive to sudden loud audio than sudden soft audio. A typical value for attack is 0.3 seconds and a typical value for decay is 0.8 seconds.

points

A list of points for the transfer function, specified in dB relative to the maximum possible signal amplitude. Each key points list must be defined using the following syntax: "x0/y0|x1/y1|x2/y2|...."

The input values must be in strictly increasing order but the transfer function does not have to be monotonically rising. The point "0/0" is assumed but may be overridden (by "0/out-dBn"). Typical values for the transfer function are "-70/-70|-60/-20".

soft-knee

Set the curve radius in dB for all joints. It defaults to 0.01.

gain

Set the additional gain in dB to be applied at all points on the transfer function. This allows for easy adjustment of the overall gain. It defaults to 0.

volume

Set an initial volume, in dB, to be assumed for each channel when filtering starts. This permits the user to supply a nominal level initially, so that, for example, a very large gain is not applied to initial signal levels before the companding has begun to operate. A typical value for audio which is initially quiet is -90 dB. It defaults to 0.

delay

Set a delay, in seconds. The input audio is analyzed immediately, but audio is delayed before being fed to the volume adjuster. Specifying a delay approximately equal to the attack/decay times allows the filter to effectively operate in predictive rather than reactive mode. It defaults to 0.

Examples

•

Make music with both quiet and loud passages suitable for listening to in a noisy environment:

        compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
    

•

A noise gate for when the noise is at a lower level than the signal:

        compand=.1|.1:.2|.2:-900/-900|-50.1/-900|-50/-50:.01:0:-90:.1
    

•

Here is another noise gate, this time for when the noise is at a higher level than the signal (making it, in some ways, similar to squelch):

        compand=.1|.1:.1|.1:-45.1/-45.1|-45/-900|0/-900:.01:45:-90:.1
    

join¶

Join multiple input streams into one multi-channel stream. It accepts the following parameters:

inputs

The number of input streams. It defaults to 2.

channel_layout

The desired output channel layout. It defaults to stereo.

map

Map channels from inputs to output. The argument is a '|'-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 they 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. Join 3 inputs (with properly set channel layouts):

        avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUT

Build a 5.1 output from 6 single-channel streams:

        avconv -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. It is not meant to be used directly; it is inserted automatically by libavfilter whenever conversion is needed. Use the aformat filter to force a specific conversion.

volume¶

Adjust the input audio volume. It accepts the following parameters:

volume

This expresses how the audio volume will be increased or decreased.

Output values are clipped to the maximum value.

The output audio volume is given by the relation:

        <output_volume> = <volume> * <input_volume>
    

The default value for volume is 1.0.

precision

This parameter represents the mathematical precision.

It determines which input sample formats will be allowed, which affects the precision of the volume scaling.

replaygain

Choose the behaviour on encountering ReplayGain side data in input frames.

replaygain_preamp

Pre-amplification gain in dB to apply to the selected replaygain gain.

Default value for replaygain_preamp is 0.0.

replaygain_noclip

Prevent clipping by limiting the gain applied.

Default value for replaygain_noclip is 1.

Examples

•

Halve the input audio volume:

        volume=volume=0.5
        volume=volume=1/2
        volume=volume=-6.0206dB
    

•

Increase input audio power by 6 decibels using fixed-point precision:

        volume=volume=6dB:precision=fixed
    

Below is a description of the currently available audio sources.

anullsrc¶

The null audio source; it never returns audio frames. It is mainly useful as a template and for use in analysis / debugging tools. It accepts, as an optional parameter, a string of the form sample_rate:channel_layout. sample_rate specifies the sample rate, and defaults to 44100. channel_layout specifies the channel layout, and can be either an integer or a string representing a channel layout. The default value of channel_layout is 3, which corresponds to CH_LAYOUT_STEREO. Check the channel_layout_map definition in libavutil/channel_layout.c for the mapping between strings and channel layout values. Some examples:

        # Set the sample rate to 48000 Hz and the channel layout to CH_LAYOUT_MONO
        anullsrc=48000:4
        
        # The same as above
        anullsrc=48000: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; it is for insertion by calling programs, through the interface defined in libavfilter/buffersrc.h. It accepts the following parameters:

time_base

The 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

The audio sample rate.

sample_fmt

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

channel_layout

The channel layout of the audio data, in the form that can be accepted by "av_get_channel_layout()".

All the parameters need to be explicitly defined. Below is a description of the currently available audio sinks.

anullsink¶

Null audio sink; do absolutely nothing with the input audio. It is mainly useful as a template and for use 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. It does not accept any parameters. When you configure your Libav 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.

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. It accepts the following parameters:

amount

The percentage of the pixels that have to be below the threshold; it defaults to 98.

threshold

The threshold below which a pixel value is considered black; it defaults to 32.

boxblur¶

Apply a boxblur algorithm to the input video. It accepts the following parameters:

luma_radius

luma_power

chroma_radius

chroma_power

alpha_radius

alpha_power

The 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

The 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:

•

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

        boxblur=luma_radius=2:luma_power=1
    

•

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

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

•

Set the luma and chroma radii to a fraction of the video dimension:

        boxblur=luma_radius=min(h,w)/10:luma_power=1:chroma_radius=min(cw,ch)/10:chroma_power=1
    

copy¶

Copy the input source unchanged to the output. This is mainly useful for testing purposes.

crop¶

Crop the input video to given dimensions. It accepts the following parameters:

out_w

The width of the output video.

out_h

The height of the output video.

x

The horizontal position, in the input video, of the left edge of the output video.

y

The vertical position, in the input video, of the top edge of the output video.

The parameters are expressions containing the following constants:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

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

These are the same as in_w and in_h.

out_w, out_h

The output (cropped) width and height.

ow, oh

These are the same as out_w and out_h.

n

The number of the input frame, starting from 0.

t

The timestamp expressed in seconds. It's 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 only evaluated during 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. Some examples:

        # Crop the central input area with size 100x100
        crop=out_w=100:out_h=100
        
        # Crop the central input area with size 2/3 of the input video
        "crop=out_w=2/3*in_w:out_h=2/3*in_h"
        
        # Crop the input video central square
        crop=out_w=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=out_w=in_w-100:out_h=in_h-100:x=100:y=100
        
        # Crop 10 pixels from the left and right borders, and 20 pixels from
        # the top and bottom borders
        "crop=out_w=in_w-2*10:out_h=in_h-2*20"
        
        # Keep only the bottom right quarter of the input image
        "crop=out_w=in_w/2:out_h=in_h/2:x=in_w/2:y=in_h/2"
        
        # Crop height for getting Greek harmony
        "crop=out_w=in_w:out_h=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=out_w=in_w/2:out_h=in_h/2:x=(in_w-out_w)/2+((in_w-out_w)/2)*sin(t*10):y=(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 the crop size. It calculates the necessary cropping parameters and prints the recommended parameters via the logging system. The detected dimensions correspond to the non-black area of the input video. It accepts the following parameters:

limit

The threshold, an optional parameter between nothing (0) and everything (255). It defaults to 24.

round

The value which the width/height should be divisible by. It 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

A counter that determines how many frames cropdetect will reset the previously detected largest video area after. It will then start over and detect the current optimal crop area. It defaults to 0.

This can be useful when channel logos distort the video area. 0 indicates 'never reset', and returns the largest area encountered during playback.

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). It accepts the following parameters:

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.

An example:

•

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

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

drawbox¶

Draw a colored box on the input image. It accepts the following parameters:

x, y

Specify the top left corner coordinates of the box. It defaults to 0.

width, height

Specify the width and height of the box; if 0 they are interpreted as the input width and height. It defaults 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.

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=x=10:y=20:width=200:height=60:color=red@0.5"

drawtext¶

Draw a text string or text from a specified file on top of a video, using the libfreetype library. To enable compilation of this filter, you need to configure Libav with "--enable-libfreetype". To enable default font fallback and the font option you need to configure Libav with "--enable-libfontconfig". The filter also recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime(). It accepts the following parameters:

font

The font family to be used for drawing text. By default Sans.

fontfile

The font file to be used for drawing text. The path must be included. This parameter is mandatory if the fontconfig support is disabled.

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 offsets where text will be drawn within the video frame. It is relative to the top/left border of the output image. They accept expressions similar to the overlay filter:

fontsize

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

fontcolor

The color to be used for drawing fonts. It is 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".

boxcolor

The color to be used for drawing box around text. It is 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".

box

Used to draw a box around text using the background color. The value must be either 1 (enable) or 0 (disable). The default value of box is 0.

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. The default value for both is "0".

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".

ft_load_flags

The 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:

tabsize

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

fix_bounds

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

For example the command:

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

will draw "Test Text" with font FreeSerif, using the default values for the optional parameters. The command:

        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"

will 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%. Note that the double quotes are not necessary if spaces are not used within the parameter list. For more information about libfreetype, check: < http://www.freetype.org/>.

fade¶

Apply a fade-in/out effect to the input video. It accepts the following parameters:

type

The effect type can be either "in" for a fade-in, or "out" for a fade-out effect.

start_frame

The number of the frame to start applying the fade effect at.

nb_frames

The number of frames that the fade effect lasts. 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.

Some examples:

        # Fade in the first 30 frames of video
        fade=type=in:nb_frames=30
        
        # Fade out the last 45 frames of a 200-frame video
        fade=type=out:start_frame=155:nb_frames=45
        
        # Fade in the first 25 frames and fade out the last 25 frames of a 1000-frame video
        fade=type=in:start_frame=0:nb_frames=25, fade=type=out:start_frame=975:nb_frames=25
        
        # Make the first 5 frames black, then fade in from frame 5-24
        fade=type=in:start_frame=5:nb_frames=20

fieldorder¶

Transform the field order of the input video. It accepts the following parameters:

order

The output field order. Valid values are tff for top field first or bff for bottom field first.

The default value is "tff". The transformation is done 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. It is very useful when converting to or from PAL DV material, which is bottom field first. For example:

        ./avconv -i in.vob -vf "fieldorder=order=bff" out.dv

fifo¶

Buffer input images and send them when they are requested. It is mainly useful when auto-inserted by the libavfilter framework. It does not take parameters.

format¶

Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is suitable as input to the next filter. It accepts the following parameters:

pix_fmts

A '|'-separated list of pixel format names, such as "pix_fmts=yuv420p|monow|rgb24".

Some examples:

        # Convert the input video to the "yuv420p" format
        format=pix_fmts=yuv420p
        
        # Convert the input video to any of the formats in the list
        format=pix_fmts=yuv420p|yuv444p|yuv410p

fps¶

Convert the video to specified constant framerate by duplicating or dropping frames as necessary. It accepts the following parameters:

fps

The desired output framerate.

start_time

Assume the first PTS should be the given value, in seconds. 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 duplicates of the first frame if a video stream starts after the audio stream or to trim any frames with a negative PTS.

framepack¶

Pack two different video streams into a stereoscopic video, setting proper metadata on supported codecs. The two views should have the same size and framerate and processing will stop when the shorter video ends. Please note that you may conveniently adjust view properties with the scale and fps filters. It accepts the following parameters:

format

The desired packing format. Supported values are:

Some examples:

        # Convert left and right views into a frame-sequential video
        avconv -i LEFT -i RIGHT -filter_complex framepack=frameseq OUTPUT
        
        # Convert views into a side-by-side video with the same output resolution as the input
        avconv -i LEFT -i RIGHT -filter_complex [0:v]scale=w=iw/2[left],[1:v]scale=w=iw/2[right],[left][right]framepack=sbs OUTPUT

frei0r¶

Apply a frei0r effect to the input video. To enable the compilation of this filter, you need to install the frei0r header and configure Libav with --enable-frei0r. It accepts the following parameters:

filter_name

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

filter_params

A '|'-separated list of parameters to pass to the frei0r effect.

A frei0r effect parameter can be a boolean (its value is either "y" or "n"), a double, a color (specified as R/G/B, where R, G, and B are floating point numbers between 0.0 and 1.0, inclusive) or by an "av_parse_color()" color description), a position (specified as X/Y, where X and Y are floating point numbers) and/or a string. The number and types of parameters depend on the loaded effect. If an effect parameter is not specified, the default value is set. Some examples:

        # Apply the distort0r effect, setting the first two double parameters
        frei0r=filter_name=distort0r:filter_params=0.5|0.01
        
        # Apply the colordistance effect, taking a color as the first parameter
        frei0r=colordistance:0.2/0.3/0.4
        frei0r=colordistance:violet
        frei0r=colordistance:0x112233
        
        # Apply the perspective effect, specifying the top left and top right
        # image positions
        frei0r=perspective:0.2/0.2|0.8/0.2

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

gradfun¶

Fix the banding artifacts that are sometimes introduced into nearly flat regions by truncation to 8bit colordepth. Interpolate the gradients that should go where the bands are, and dither them. It 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. It accepts the following parameters:

strength

The maximum amount by which the filter will change any one pixel. This is also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 64; the default value is 1.2. Out-of-range values will be clipped to the valid range.

radius

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; the default value is 16. Out-of-range values will be clipped to the valid range.

        # Default parameters
        gradfun=strength=1.2:radius=16
        
        # Omitting the radius
        gradfun=1.2

hflip¶

Flip the input video horizontally. For example, to horizontally flip the input video with avconv:

        avconv -i in.avi -vf "hflip" out.avi

hqdn3d¶

This is a high precision/quality 3d denoise filter. It 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

A non-negative floating point number which specifies spatial luma strength. It defaults to 4.0.

chroma_spatial

A non-negative floating point number which specifies spatial chroma strength. It defaults to 3.0* luma_spatial/4.0.

luma_tmp

A floating point number which specifies luma temporal strength. It defaults to 6.0* luma_spatial/4.0.

chroma_tmp

A floating point number which specifies chroma temporal strength. It defaults to luma_tmp*chroma_spatial/luma_spatial.

interlace¶

Simple interlacing filter from progressive contents. This interleaves upper (or lower) lines from odd frames with lower (or upper) lines from even frames, halving the frame rate and preserving image height.

           Original        Original             New Frame
           Frame 'j'      Frame 'j+1'             (tff)
          ==========      ===========       ==================
            Line 0  -------------------->    Frame 'j' Line 0
            Line 1          Line 1  ---->   Frame 'j+1' Line 1
            Line 2 --------------------->    Frame 'j' Line 2
            Line 3          Line 3  ---->   Frame 'j+1' Line 3
             ...             ...                   ...
        New Frame + 1 will be generated by Frame 'j+2' and Frame 'j+3' and so on

It accepts the following optional parameters:

scan

This determines whether the interlaced frame is taken from the even (tff - default) or odd (bff) lines of the progressive frame.

lowpass

Enable (default) or disable the vertical lowpass filter to avoid twitter interlacing and reduce moire patterns.

lut, lutrgb, lutyuv¶

Compute a look-up table for binding each pixel component input value to an output value, and apply it to the input video. lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video. These filters accept the following parameters:

c0 (first pixel component)

c1 (second pixel component)

c2 (third pixel component)

c3 (fourth pixel component, corresponds to the alpha component)

r (red component)

g (green component)

b (blue component)

a (alpha component)

y (Y/luminance component)

u (U/Cb component)

v (V/Cr component)

Each of them specifies the expression to use for computing the lookup table for the corresponding pixel component values. The exact component associated to each of the c* options depends on the format in input. The lut filter requires either YUV or RGB pixel formats in input, lutrgb requires RGB pixel formats in input, and lutyuv requires YUV. The expressions can contain the following constants and functions:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

w, h

The input width and height.

val

The input value for the pixel component.

clipval

The input value, clipped to the minval-maxval range.

maxval

The maximum value for the pixel component.

minval

The minimum value for the pixel component.

negval

The negated value for the pixel component value, clipped to the minval- maxval range; it corresponds to the expression "maxval-clipval+minval".

clip(val)

The computed value in val, clipped to the minval-maxval range.

gammaval(gamma)

The computed gamma correction value of the pixel component value, clipped to the minval-maxval range. It corresponds to the expression "pow((clipval-minval)/(maxval-minval), gamma)*(maxval-minval)+minval"

All expressions default to "val". Some examples:

        # 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=negval
        
        # Remove chroma components, turning 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 factor of 0.5
        lutyuv=y=gammaval(0.5)

negate¶

Negate input video. It 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. It accepts the following parameters:

pix_fmts

A '|'-separated list of pixel format names, such as apix_fmts=yuv420p|monow|rgb24".

Some examples:

        # Force libavfilter to use a format different from "yuv420p" for the
        # input to the vflip filter
        noformat=pix_fmts=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 a video transform using libopencv. To enable this filter, install the libopencv library and headers and configure Libav with --enable-libopencv. It accepts the following parameters:

filter_name

The name of the libopencv filter to apply.

filter_params

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> Several libopencv filters are supported; see the following subsections. dilate Dilate an image by using a specific structuring element. It corresponds to the libopencv function "cvDilate". It accepts the parameters: struct_el|nb_iterations. struct_el represents a structuring element, and has the syntax: colsx rows+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. shape must be "rect", "cross", "ellipse", or "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. Some examples:

        # Use the default values
        ocv=dilate
        
        # Dilate using a structuring element with a 5x5 cross, iterating two times
        ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2
        
        # Read the shape from the file diamond.shape, iterating two times.
        # The file diamond.shape may contain a pattern of characters like this
        #   *
        #  ***
        # *****
        #  ***
        #   *
        # The specified columns and rows are ignored
        # but the anchor point coordinates are not
        ocv=dilate:0x0+2x2/custom=diamond.shape|2

erode Erode an image by using a specific structuring element. It corresponds to the libopencv function "cvErode". It 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 must be one of the following values: "blur", "blur_no_scale", "median", "gaussian", or "bilateral". The default value is "gaussian". The meaning of param1, param2, param3, and param4 depend on the smooth type. param1 and param2 accept integer positive values or 0. param3 and param4 accept floating point 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 has one output. The first input is the "main" video on which the second input is overlayed. It accepts the following parameters:

x

The horizontal position of the left edge of the overlaid video on the main video.

y

The vertical position of the top edge of the overlaid video on the main video.

The parameters are expressions containing the following parameters:

main_w, main_h

The main input width and height.

W, H

These are the same as main_w and main_h.

overlay_w, overlay_h

The overlay input width and height.

w, h

These are the same as overlay_w and overlay_h.

eof_action

The action to take when EOF is encountered on the secondary input; it accepts one of the following values:

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 the example for the movie filter does. Some examples:

        # Draw the overlay at 10 pixels from the bottom right
        # corner of the main video
        overlay=x=main_w-overlay_w-10:y=main_h-overlay_h-10
        
        # Insert a transparent PNG logo in the bottom left corner of the input
        avconv -i input -i logo -filter_complex 'overlay=x=10:y=main_h-overlay_h-10' output
        
        # Insert 2 different transparent PNG logos (second logo on bottom
        # right corner)
        avconv -i input -i logo1 -i logo2 -filter_complex
        'overlay=x=10:y=H-h-10,overlay=x=W-w-10:y=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]
        
        # Mask 10-20 seconds of a video by applying the delogo filter to a section
        avconv -i test.avi -codec:v:0 wmv2 -ar 11025 -b:v 9000k
        -vf '[in]split[split_main][split_delogo];[split_delogo]trim=start=360:end=371,delogo=0:0:640:480[delogoed];[split_main][delogoed]overlay=eof_action=pass[out]'
        masked.avi

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 place the original input at the provided x, y coordinates. It accepts the following 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 to place the input image at within 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 an 0xRRGGBB[AA] sequence.

The default value of color is "black".

The parameters width, height, x, and y are expressions containing the following constants:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

in_w, in_h

The input video width and height.

iw, ih

These are the same as in_w and in_h.

out_w, out_h

The output width and height (the size of the padded area), as specified by the width and height expressions.

ow, oh

These are the same as out_w and out_h.

x, y

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

a

The input display aspect ratio, same as iw / ih.

hsub, vsub

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

Some examples:

        # Add paddings with the color "violet" to the input video. The output video
        # size is 640x480, and the top-left corner of the input video is placed at
        # column 0, row 40
        pad=width=640:height=480:x=0:y=40:color=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"
        
        # Double the 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.

scale¶

Scale the input video and/or convert the image format. It accepts the following parameters:

w

The output video width.

h

The output video height.

The parameters w and h are expressions containing the following constants:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

in_w, in_h

The input width and height.

iw, ih

These are the same as in_w and in_h.

out_w, out_h

The output (cropped) width and height.

ow, oh

These are the same as out_w and out_h.

a

This is the same as iw / ih.

sar

input sample aspect ratio

dar

The input display aspect ratio; it is the same as ( iw / ih) * sar.

hsub, vsub

The 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 w or h is 0, the respective input size is used for the output. If the value for w or h 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 w and h is 0. Some examples:

        # Scale the input video to a size of 200x100
        scale=w=200:h=100
        
        # Scale the input to 2x
        scale=w=2*iw:h=2*ih
        # The above is the same as
        scale=2*in_w:2*in_h
        
        # Scale the input to half the original size
        scale=w=iw/2:h=ih/2
        
        # Increase the width, and set the height to the same size
        scale=3/2*iw:ow
        
        # Seek 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=w=3/2*oh:h=3/5*ih
        
        # Increase the size, making 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,
        # keeping the same aspect ratio as the input
        scale=w='min(500, iw*3/2):h=-1'

select¶

Select frames to pass in output. It accepts the following parameters:

expr

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:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

n

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. It's NAN if undefined.

TB

The timebase of the input timestamps.

pts

The PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units. It's NAN if undefined.

t

The PTS of the filtered video frame, expressed in seconds. It's NAN if undefined.

prev_pts

The PTS of the previously filtered video frame. It's NAN if undefined.

prev_selected_pts

The PTS of the last previously filtered video frame. It's NAN if undefined.

prev_selected_t

The PTS of the last previously selected video frame. It's NAN if undefined.

start_pts

The PTS of the first video frame in the video. It's NAN if undefined.

start_t

The time of the first video frame in the video. It's NAN if undefined.

pict_type

The type of the filtered frame. It can assume one of the following values:

interlace_type

The frame interlace type. It can assume one of the following values:

key

This is 1 if the filtered frame is a key-frame, 0 otherwise.

The default value of the select expression is "1". Some examples:

        # Select all the frames in input
        select
        
        # The above is the same as
        select=expr=1
        
        # Skip all frames
        select=expr=0
        
        # Select only I-frames
        select='expr=eq(pict_type,I)'
        
        # Select one frame per 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)'

setdar¶

Set 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 this 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. It accepts the following parameters:

dar

The output display aspect ratio.

The parameter dar is an expression containing the following constants:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

w, h

The input width and height.

a

This is the same as w / h.

sar

The input sample aspect ratio.

dar

The input display aspect ratio. It is the same as ( w / h) * sar.

hsub, vsub

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

To change the display aspect ratio to 16:9, specify:

        setdar=dar=16/9
        # The above is equivalent to
        setdar=dar=1.77777

Also see the the setsar filter documentation.

setpts¶

Change the PTS (presentation timestamp) of the input video frames. It accepts the following parameters:

expr

The expression which is evaluated for each frame to construct its timestamp.

The expression is evaluated through the eval API and can contain the following constants:

PTS

The presentation timestamp in input.

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

N

The count of the input frame, starting from 0.

STARTPTS

The PTS of the first video frame.

INTERLACED

State whether the current frame is interlaced.

PREV_INPTS

The previous input PTS.

PREV_OUTPTS

The previous output PTS.

RTCTIME

The wallclock (RTC) time in microseconds.

RTCSTART

The wallclock (RTC) time at the start of the movie in microseconds.

TB

The timebase of the input timestamps.

Some examples:

        # Start counting the PTS from zero
        setpts=expr=PTS-STARTPTS
        
        # Fast motion
        setpts=expr=0.5*PTS
        
        # Slow motion
        setpts=2.0*PTS
        
        # Fixed rate 25 fps
        setpts=N/(25*TB)
        
        # Fixed rate 25 fps with some jitter
        setpts='1/(25*TB) * (N + 0.05 * sin(N*2*PI/25))'
        
        # Generate timestamps from a "live source" and rebase onto the current timebase
        setpts='(RTCTIME - RTCSTART) / (TB * 1000000)"

setsar¶

Set 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 following equation: DAR = HORIZONTAL_RESOLUTION / VERTICAL_RESOLUTION * SAR Keep in mind that the sample aspect ratio set by this filter may be changed by later filters in the filterchain, e.g. if another "setsar" or a "setdar" filter is applied. It accepts the following parameters:

sar

The output sample aspect ratio.

The parameter sar is an expression containing the following constants:

E, PI, PHI

These are approximated values for the mathematical constants e (Euler's number), pi (Greek pi), and phi (the golden ratio).

w, h

The input width and height.

a

These are the same as w / h.

sar

The input sample aspect ratio.

dar

The 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.

To change the sample aspect ratio to 10:11, specify:

        setsar=sar=10/11

settb¶

Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration. It accepts the following parameters:

expr

The expression which is evaluated into the output timebase.

The expression can contain the constants "PI", "E", "PHI", "AVTB" (the default timebase), and "intb" (the input timebase). The default value for the input is "intb". Some examples:

        # Set the timebase to 1/25
        settb=expr=1/25
        
        # Set the timebase to 1/10
        settb=expr=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

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. It accepts the following parameters:

n

The (sequential) number of the input frame, starting from 0.

pts

The 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

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

pos

The position of the frame in the input stream, or -1 if this information is unavailable and/or meaningless (for example in case of synthetic video).

fmt

The pixel format name.

sar

The sample aspect ratio of the input frame, expressed in the form num/ den.

s

The size of the input frame, expressed in the form widthxheight.

i

The type of interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first).

iskey

This is 1 if the frame is a key frame, 0 otherwise.

type

The picture type of the input frame ("I" for an I-frame, "P" for a P-frame, "B" for a B-frame, or "?" for an unknown type). Also refer to the documentation of the "AVPictureType" enum and of the "av_get_picture_type_char" function defined in libavutil/avutil.h.

checksum

The Adler-32 checksum of all the planes of the input frame.

plane_checksum

The Adler-32 checksum of each plane of the input frame, expressed in the form "[ c0 c1 c2 c3]".

shuffleplanes¶

Reorder and/or duplicate video planes. It accepts the following parameters:

map0

The index of the input plane to be used as the first output plane.

map1

The index of the input plane to be used as the second output plane.

map2

The index of the input plane to be used as the third output plane.

map3

The index of the input plane to be used as the fourth output plane.

The first plane has the index 0. The default is to keep the input unchanged. Swap the second and third planes of the input:

        avconv -i INPUT -vf shuffleplanes=0:2:1:3 OUTPUT

split¶

Split input video into several identical outputs. It accepts a single parameter, which specifies the number of outputs. If unspecified, it defaults to 2. Create 5 copies of the input video:

        avconv -i INPUT -filter_complex split=5 OUTPUT

transpose¶

Transpose rows with columns in the input video and optionally flip it. It accepts the following parameters:

dir

The direction of the transpose.

The direction can assume the following values:

cclock_flip

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

        L.R     L.l
        . . ->  . .
        l.r     R.r
    

clock

Rotate by 90 degrees clockwise, that is:

        L.R     l.L
        . . ->  . .
        l.r     r.R
    

cclock

Rotate by 90 degrees counterclockwise, that is:

        L.R     R.r
        . . ->  . .
        l.r     L.l
    

clock_flip

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

        L.R     r.R
        . . ->  . .
        l.r     l.L
    

trim¶

Trim the input so that the output contains one continuous subpart of the input. It accepts the following parameters:

start

The timestamp (in seconds) of the start of the kept section. The frame with the timestamp start will be the first frame in the output.

end

The timestamp (in seconds) of the first frame that will be dropped. The frame immediately preceding the one with the timestamp end will be the last frame in the output.

start_pts

This is the same as start, except this option sets the start timestamp in timebase units instead of seconds.

end_pts

This is the same as end, except this option sets the end timestamp in timebase units instead of seconds.

duration

The maximum duration of the output in seconds.

start_frame

The number of the first frame that should be passed to the output.

end_frame

The number of the first frame that should be dropped.

Note that the first two sets of the start/end options and the duration option look at the frame timestamp, while the _frame variants simply count the frames that pass through the filter. Also note that this filter does not modify the timestamps. If you wish for the output timestamps to start at zero, insert a setpts filter after the trim filter. If multiple start or end options are set, this filter tries to be greedy and keep all the frames that match at least one of the specified constraints. To keep only the part that matches all the constraints at once, chain multiple trim filters. The defaults are such that all the input is kept. So it is possible to set e.g. just the end values to keep everything before the specified time. Examples:

•

Drop everything except the second minute of input:

        avconv -i INPUT -vf trim=60:120
    

•

Keep only the first second:

        avconv -i INPUT -vf trim=duration=1
    

unsharp¶

Sharpen or blur the input video. It accepts the following parameters:

luma_msize_x

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

luma_msize_y

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

luma_amount

Set the luma effect strength. It must be a floating point number between -2.0 and 5.0. The default value is 1.0.

chroma_msize_x

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

chroma_msize_y

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

chroma_amount

Set the chroma effect strength. It must be a floating point number between -2.0 and 5.0. The default value is 0.0.

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'.

        # Strong luma sharpen effect parameters
        unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5
        
        # A strong blur of both luma and chroma parameters
        unsharp=7:7:-2:7:7:-2
        
        # Use the default values with B<avconv>
        ./avconv -i in.avi -vf "unsharp" out.mp4

vflip¶

Flip the input video vertically.

        ./avconv -i in.avi -vf "vflip" out.avi

yadif¶

Deinterlace the input video ("yadif" means "yet another deinterlacing filter"). It accepts the following parameters:

mode

The interlacing mode to adopt. It accepts one of the following values:

parity

The picture field parity assumed for the input interlaced video. It accepts one of the following values:

auto

Whether the deinterlacer should trust the interlaced flag and only deinterlace frames marked as interlaced.

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 the following parameters:

width

The input video width.

height

The input video height.

pix_fmt

The name of the input video pixel format.

time_base

The time base used for input timestamps.

sar

The sample (pixel) aspect ratio of the input video.

For example:

        buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=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).

color¶

Provide an uniformly colored input. It accepts the following parameters:

color

Specify the color of the source. It can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence, possibly followed by an alpha specifier. The default value is "black".

size

Specify the size of the sourced video, it may be a string of the form widthx height, or the name of a size abbreviation. The default value is "320x240".

framerate

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 floating point number or a valid video frame rate abbreviation. The default value is "25".

The following graph description will generate a red source with an opacity of 0.2, with size "qcif" and a frame rate of 10 frames per second, which will be overlayed over the source connected to the pad with identifier "in":

        "color=red@0.2:qcif:10 [color]; [in][color] overlay [out]"

movie¶

Read a video stream from a movie container. Note that this source is a hack that bypasses the standard input path. It can be useful in applications that do not support arbitrary filter graphs, but its use is discouraged in those that do. It should never be used with avconv; the -filter_complex option fully replaces it. It accepts the following parameters:

filename

The name of the resource to read (not necessarily a file; it can also be a device or a stream accessed through some protocol).

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. The default value is "0".

stream_index, si

Specifies the index of the video stream to read. If the value is -1, the most suitable video stream will be automatically selected. The default value is "-1".

It allows overlaying a second video on top of the main input of a filtergraph, as shown in this graph:

        input -----------> deltapts0 --> overlay --> output
                                            ^
                                            |
        movie --> scale--> deltapts1 -------+

Some examples:

        # Skip 3.2 seconds from the start of the AVI file in.avi, and overlay it
        # on top of the input labelled "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 "in"
        movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie];
        [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out]

nullsrc¶

Null video source: never return images. It is mainly useful as a template and to be employed in analysis / debugging tools. It accepts a string of the form width:height:timebase as an optional parameter. width and height specify the size of the configured source. The default values of width and height are respectively 352 and 288 (corresponding to the CIF size format). timebase specifies an arithmetic expression representing a timebase. The expression can contain the constants "PI", "E", "PHI", and "AVTB" (the default timebase), and defaults to the value "AVTB".

frei0r_src¶

Provide a frei0r source. To enable compilation of this filter you need to install the frei0r header and configure Libav with --enable-frei0r. This source accepts the following parameters:

size

The size of the video to generate. It may be a string of the form widthx height or a frame size abbreviation.

framerate

The framerate of the generated video. It may be a string of the form num/ den or a frame rate abbreviation.

filter_name

The name to the frei0r source to load. For more information regarding frei0r and how to set the parameters, read the frei0r section in the video filters documentation.

filter_params

A '|'-separated list of parameters to pass to the frei0r source.

An example:

        # Generate a frei0r partik0l source with size 200x200 and framerate 10
        # which is overlayed on the overlay filter main input
        frei0r_src=size=200x200:framerate=10:filter_name=partik0l:filter_params=1234 [overlay]; [in][overlay] overlay

rgbtestsrc, testsrc¶

The "rgbtestsrc" source generates an RGB test pattern useful for detecting RGB vs BGR issues. You should see a red, green and blue stripe from top to bottom. The "testsrc" source generates a test video pattern, showing a color pattern, a scrolling gradient and a timestamp. This is mainly intended for testing purposes. The sources accept the following parameters:

size, s

Specify the size of the sourced video, it may be a string of the form widthx height, or the name of a size abbreviation. The default value is "320x240".

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 floating point number or a valid video frame rate abbreviation. The default value is "25".

sar

Set the sample aspect ratio of the sourced video.

duration

Set the video duration of the sourced video. The accepted syntax is:

        [-]HH[:MM[:SS[.m...]]]
        [-]S+[.m...]
    

Also see the the "av_parse_time()" function.

If not specified, or the expressed duration is negative, the video is supposed to be generated forever.

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 framerate of 10 frames per second. 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 intended for programmatic use through the interface defined in libavfilter/buffersink.h.

nullsink¶

Null video sink: do absolutely nothing with the input video. It is mainly useful as a template and for use in analysis / debugging tools. avconv(1), avprobe(1) and the Libav HTML documentation The Libav developers