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AVPLAY(1) | AVPLAY(1) |
NAME¶
avplay - AVplay media playerSYNOPSIS¶
avplay [options] input_fileDESCRIPTION¶
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.OPTIONS¶
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.
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.
- decoder=decoder_name
- Print detailed information about the decoder named decoder_name. Use the -decoders option to get a list of all decoders.
- encoder=encoder_name
- Print detailed information about the encoder named encoder_name. Use the -encoders option to get a list of all encoders.
- demuxer=demuxer_name
- Print detailed information about the demuxer named demuxer_name. Use the -formats option to get a list of all demuxers and muxers.
- muxer=muxer_name
- Print detailed information about the muxer named muxer_name. Use the -formats option to get a list of all muxers and demuxers.
- filter=filter_name
- Print detailed information about the filter name filter_name. Use the -filters option to get a list of all filters.
- -version
- Show version.
- -formats
- Show available formats.
- D
- Decoding available
- E
- Encoding available
- -codecs
- Show all codecs known to libavcodec.
- -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:
- quiet
- panic
- fatal
- error
- warning
- info
- verbose
- debug
- -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.
avconv -i input.flac -id3v2_version 3 out.mp3All 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:
- unaligned
- allow decoders to produce unaligned output
- mv4
- use four motion vectors per macroblock (MPEG-4)
- qpel
- use 1/4-pel motion compensation
- loop
- use loop filter
- qscale
- use fixed qscale
- gmc
- use gmc
- mv0
- always try a mb with mv=<0,0>
- input_preserved
- pass1
- use internal 2-pass ratecontrol in first pass mode
- pass2
- use internal 2-pass ratecontrol in second pass mode
- gray
- only decode/encode grayscale
- emu_edge
- do not draw edges
- psnr
- error[?] variables will be set during encoding
- truncated
- naq
- normalize adaptive quantization
- ildct
- use interlaced DCT
- low_delay
- force low delay
- global_header
- place global headers in extradata instead of every keyframe
- bitexact
- use only bitexact functions (except (I)DCT)
- aic
- H.263 advanced intra coding / MPEG-4 AC prediction
- ilme
- interlaced motion estimation
- cgop
- closed GOP
- output_corrupt
- Output even potentially corrupted frames
- -me_method[:stream_specifier] integer ( output,video)
- set motion estimation method
- zero
- zero motion estimation (fastest)
- full
- full motion estimation (slowest)
- epzs
- EPZS motion estimation (default)
- esa
- esa motion estimation (alias for full)
- tesa
- tesa motion estimation
- dia
- diamond motion estimation (alias for EPZS)
- log
- log motion estimation
- phods
- phods motion estimation
- x1
- X1 motion estimation
- hex
- hex motion estimation
- umh
- umh motion estimation
- -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
- autodetect
- old_msmpeg4
- some old lavc-generated MSMPEG4v3 files (no autodetection)
- xvid_ilace
- Xvid interlacing bug (autodetected if FOURCC == XVIX)
- ump4
- (autodetected if FOURCC == UMP4)
- no_padding
- padding bug (autodetected)
- amv
- ac_vlc
- illegal VLC bug (autodetected per FOURCC)
- qpel_chroma
- std_qpel
- old standard qpel (autodetected per FOURCC/version)
- qpel_chroma2
- direct_blocksize
- direct-qpel-blocksize bug (autodetected per FOURCC/version)
- edge
- edge padding bug (autodetected per FOURCC/version)
- hpel_chroma
- dc_clip
- ms
- work around various bugs in Microsoft's broken decoders
- trunc
- truncated frames
- -strict[:stream_specifier] integer (input/output,audio,video)
- how strictly to follow the standards
- very
- strictly conform to a older more strict version of the spec or reference software
- strict
- strictly conform to all the things in the spec no matter what the consequences
- normal
- unofficial
- allow unofficial extensions
- experimental
- allow non-standardized experimental things
- -b_qoffset[:stream_specifier] float (output,video)
- QP offset between P- and B-frames
- -err_detect[:stream_specifier] flags (input,audio,video)
- set error detection flags
- crccheck
- verify embedded CRCs
- bitstream
- detect bitstream specification deviations
- buffer
- detect improper bitstream length
- explode
- abort decoding on minor error detection
- -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
- auto
- autoselect a good one (default)
- fastint
- fast integer
- int
- accurate integer
- mmx
- altivec
- faan
- floating point AAN DCT
- -lumi_mask[:stream_specifier] float (output,video)
- compresses bright areas stronger than medium ones
- -tcplx_mask[:stream_specifier] float (output,video)
- temporal complexity masking
- -scplx_mask[:stream_specifier] float (output,video)
- spatial complexity masking
- -p_mask[:stream_specifier] float (output,video )
- inter masking
- -dark_mask[:stream_specifier] float (output,video)
- compresses dark areas stronger than medium ones
- -idct[:stream_specifier] integer (input/output,video )
- select IDCT implementation
- auto
- int
- simple
- simplemmx
- arm
- altivec
- sh4
- simplearm
- simplearmv5te
- simplearmv6
- simpleneon
- simplealpha
- ipp
- xvidmmx
- faani
- floating point AAN IDCT
- -ec[:stream_specifier] flags (input,video )
- set error concealment strategy
- guess_mvs
- iterative motion vector (MV) search (slow)
- deblock
- use strong deblock filter for damaged MBs
- -pred[:stream_specifier] integer (output,video )
- prediction method
- left
- plane
- median
- -aspect[:stream_specifier] rational number (output,video)
- sample aspect ratio
- -debug[:stream_specifier] flags (input/output,audio,video,subtitles )
- print specific debug info
- pict
- picture info
- rc
- rate control
- bitstream
- mb_type
- macroblock (MB) type
- qp
- per-block quantization parameter (QP)
- mv
- motion vector
- dct_coeff
- skip
- startcode
- pts
- er
- error recognition
- mmco
- memory management control operations (H.264)
- bugs
- vis_qp
- visualize quantization parameter (QP), lower QP are tinted greener
- vis_mb_type
- visualize block types
- buffers
- picture buffer allocations
- thread_ops
- threading operations
- -vismv[:stream_specifier] integer (input,video )
- visualize motion vectors (MVs)
- pf
- forward predicted MVs of P-frames
- bf
- forward predicted MVs of B-frames
- bb
- backward predicted MVs of B-frames
- -cmp[:stream_specifier] integer (output,video )
- full-pel ME compare function
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -subcmp[:stream_specifier] integer (output,video)
- sub-pel ME compare function
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -mbcmp[:stream_specifier] integer (output,video )
- macroblock compare function
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -ildctcmp[:stream_specifier] integer (output,video)
- interlaced DCT compare function
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -dia_size[:stream_specifier] integer (output,video)
- diamond type & size for motion estimation
- -last_pred[:stream_specifier] integer ( output,video)
- amount of motion predictors from the previous frame
- -preme[:stream_specifier] integer (output,video )
- pre motion estimation
- -precmp[:stream_specifier] integer (output,video)
- pre motion estimation compare function
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -pre_dia_size[:stream_specifier] integer ( output,video)
- diamond type & size for motion estimation pre-pass
- -subq[:stream_specifier] integer (output,video )
- sub-pel motion estimation quality
- -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:
- vlc
- variable length coder / Huffman coder
- ac
- arithmetic coder
- raw
- raw (no encoding)
- rle
- run-length coder
- deflate
- deflate-based coder
- -context[:stream_specifier] integer (output,video)
- context model
- -mbd[:stream_specifier] integer (output,video )
- macroblock decision algorithm (high quality mode)
- simple
- use mbcmp (default)
- bits
- use fewest bits
- rd
- use best rate distortion
- -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:
- fast
- allow non-spec-compliant speedup tricks
- noout
- skip bitstream encoding
- ignorecrop
- ignore cropping information from sps
- local_header
- place global headers at every keyframe instead of in extradata
- -error[:stream_specifier] integer (output,video )
- -threads[:stream_specifier] integer (input/output,video)
- Possible values:
- auto
- autodetect a suitable number of threads to use
- -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:
- unknown
- aac_main
- aac_low
- aac_ssr
- aac_ltp
- aac_he
- aac_he_v2
- aac_ld
- aac_eld
- mpeg2_aac_low
- mpeg2_aac_he
- dts
- dts_es
- dts_96_24
- dts_hd_hra
- dts_hd_ma
- -level[:stream_specifier] integer (output,audio,video )
- Possible values:
- unknown
- -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
- sad
- sum of absolute differences, fast (default)
- sse
- sum of squared errors
- satd
- sum of absolute Hadamard transformed differences
- dct
- sum of absolute DCT transformed differences
- psnr
- sum of squared quantization errors (avoid, low quality)
- bit
- number of bits needed for the block
- rd
- rate distortion optimal, slow
- zero
- 0
- vsad
- sum of absolute vertical differences
- vsse
- sum of squared vertical differences
- nsse
- noise preserving sum of squared differences
- dctmax
- chroma
- -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:
- none
- default
- noref
- bidir
- nokey
- all
- -skip_idct[:stream_specifier] integer ( input,video)
- Possible values:
- none
- default
- noref
- bidir
- nokey
- all
- -skip_frame[:stream_specifier] integer ( input,video)
- Possible values:
- none
- default
- noref
- bidir
- nokey
- all
- -bidir_refine[:stream_specifier] integer ( output,video)
- refine the two motion vectors used in bidirectional macroblocks
- -brd_scale[:stream_specifier] integer ( output,video)
- 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_trc[:stream_specifier] integer ( input/output,video)
- -colorspace[:stream_specifier] integer ( input/output,video)
- -color_range[:stream_specifier] integer ( input/output,video)
- -chroma_sample_location[:stream_specifier] integer (input/output,video)
- -slices[:stream_specifier] integer (output,video)
- number of slices, used in parallelized encoding
- -thread_type[:stream_specifier] flags ( input/output,video)
- select multithreading type
- slice
- frame
- -audio_service_type[:stream_specifier] integer (output,audio)
- audio service type
- ma
- Main Audio Service
- ef
- Effects
- vi
- Visually Impaired
- hi
- Hearing Impaired
- di
- Dialogue
- co
- Commentary
- em
- Emergency
- vo
- Voice Over
- ka
- Karaoke
- -request_sample_fmt[:stream_specifier] integer (input,audio)
- Possible values:
- u8
- 8-bit unsigned integer
- s16
- 16-bit signed integer
- s32
- 32-bit signed integer
- flt
- 32-bit float
- dbl
- 64-bit double
- u8p
- 8-bit unsigned integer planar
- s16p
- 16-bit signed integer planar
- s32p
- 32-bit signed integer planar
- fltp
- 32-bit float planar
- dblp
- 64-bit double planar
- -refcounted_frames[:stream_specifier] integer (input,audio,video)
Format AVOptions¶
- -probesize integer (input)
- set probing size
- -packetsize integer (output)
- set packet size
- -fflags flags (input/output)
- Possible values:
- flush_packets
- reduce the latency by flushing out packets immediately
- ignidx
- ignore index
- genpts
- generate pts
- nofillin
- do not fill in missing values that can be exactly calculated
- noparse
- disable AVParsers, this needs nofillin too
- igndts
- ignore dts
- discardcorrupt
- discard corrupted frames
- nobuffer
- reduce the latency introduced by optional buffering
- -analyzeduration integer (input)
- how many microseconds are analyzed to estimate duration
- -cryptokey hexadecimal string (input )
- decryption key
- -indexmem integer (input)
- max memory used for timestamp index (per stream)
- -rtbufsize integer (input)
- max memory used for buffering real-time frames
- -fdebug flags (input/output)
- print specific debug info
- ts
- -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)
- crccheck
- verify embedded CRCs
- bitstream
- detect bitstream specification deviations
- buffer
- detect improper bitstream length
- explode
- abort decoding on minor error detection
- -err_detect flags (input)
- set error detection flags
- crccheck
- verify embedded CRCs
- bitstream
- detect bitstream specification deviations
- buffer
- detect improper bitstream length
- explode
- abort decoding on minor error detection
- -max_interleave_delta integer (output )
- maximum buffering duration for interleaving
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.
EXPRESSION EVALUATION¶
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.
if A then B else Cis equivalent to
A*B + not(A)*CIn your C code, you can extend the list of unary and binary functions, and define recognized constants, so that they are available for your expressions. The evaluator also recognizes the International System number postfixes. If 'i' is appended after the postfix, powers of 2 are used instead of powers of 10. The 'B' postfix multiplies the value for 8, and can be appended after another postfix or used alone. This allows using for example 'KB', 'MiB', 'G' and 'B' as postfix. Follows the list of available International System postfixes, with indication of the corresponding powers of 10 and of 2.
- y
- -24 / -80
- z
- -21 / -70
- a
- -18 / -60
- f
- -15 / -50
- p
- -12 / -40
- n
- -9 / -30
- u
- -6 / -20
- m
- -3 / -10
- c
- -2
- d
- -1
- h
- 2
- k
- 3 / 10
- K
- 3 / 10
- M
- 6 / 20
- G
- 9 / 30
- T
- 12 / 40
- P
- 15 / 40
- E
- 18 / 50
- Z
- 21 / 60
- Y
- 24 / 70
DECODERS¶
Decoders are configured elements in 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.AUDIO 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:
- drc_scale == 0
- DRC disabled. Produces full range audio.
- 0 < drc_scale <= 1
- DRC enabled. Applies a fraction of the stream DRC value. Audio reproduction is between full range and full compression.
- drc_scale > 1
- DRC enabled. Applies drc_scale asymmetrically. Loud sounds are fully compressed. Soft sounds are enhanced.
DEMUXERS¶
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.mkvNote 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¶
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.crcYou 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.crcYou 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.
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
- mono
- video is not stereo
- left_right
- Both views are arranged side by side, Left-eye view is on the left
- bottom_top
- Both views are arranged in top-bottom orientation, Left-eye view is at bottom
- top_bottom
- Both views are arranged in top-bottom orientation, Left-eye view is on top
- checkerboard_rl
- Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
- checkerboard_lr
- Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
- row_interleaved_rl
- Each view is constituted by a row based interleaving, Right-eye view is first row
- row_interleaved_lr
- Each view is constituted by a row based interleaving, Left-eye view is first row
- col_interleaved_rl
- Both views are arranged in a column based interleaving manner, Right-eye view is first column
- col_interleaved_lr
- Both views are arranged in a column based interleaving manner, Left-eye view is first column
- anaglyph_cyan_red
- All frames are in anaglyph format viewable through red-cyan filters
- right_left
- Both views are arranged side by side, Right-eye view is on the left
- anaglyph_green_magenta
- All frames are in anaglyph format viewable through green-magenta filters
- block_lr
- Both eyes laced in one Block, Left-eye view is first
- block_rl
- Both eyes laced in one Block, Right-eye view is first
avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webmThis 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.
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.
- -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.
- -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.
- -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.
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.mp3Attach 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.mp3Write 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).
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.nullNote 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 -
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_size size
- Overwrite the listfile once it reaches size entries.
- 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¶
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.wavFor 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.aviYou can take a single screenshot image with the command:
avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpegSee 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_1For 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.wavFor 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.wavserver 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.mpgfollow_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.mpgshow_region AVOption The syntax is:
-show_region 1If 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¶
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¶
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.mpegNote 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.mpegThe 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.m3u8Using 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 0.
- 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.
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.aviNote 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.
- 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.
avplay rtmp://myserver/vod/sampleTo 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/mystreamTo 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'sRTSP 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.
- filter_src
- Accept packets only from negotiated peer address and port.
- listen
- Act as a server, listening for an incoming connection.
avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4To watch a stream tunneled over HTTP:
avplay -rtsp_transport http rtsp://server/video.mp4To 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.sdpTo 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.
avconv -re -i <input> -f sap sap://224.0.0.255?same_port=1Similarly, for watching in avplay:
avconv -re -i <input> -f sap sap://224.0.0.255And 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.)
- 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.
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=65535To 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.
FILTERGRAPH DESCRIPTION¶
A filtergraph is a directed graph of connected filters. It can contain cycles, and there can be multiple links between a pair of filters. Each link has one input pad on one side connecting it to one filter from which it takes its input, and one output pad on the other side connecting it to the one filter accepting its output. Each filter in a filtergraph is an instance of a filter class registered in the application, which defines the features and the number of input and output pads of the filter. A filter with no input pads is called a "source", a filter with no output pads is called a "sink".Filtergraph syntax¶
A filtergraph can be represented using a textual representation, which is recognized by the -filter/-vf and -filter_complex options in avconv and -vf in avplay, and by the "avfilter_graph_parse()"/"avfilter_graph_parse2()" function 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 the two allowed 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.
nullsrc, split[L1], [L2]overlay, nullsinkthe split filter instance has two output pads, and the overlay filter instance two input pads. The first output pad of split is labelled "L1", the first input pad of overlay is labelled "L2", and the second output pad of split is linked to the second input pad of overlay, which are both unlabelled. In a complete filterchain all the unlabelled filter input and output pads must be connected. A filtergraph is considered valid if all the filter input and output pads of all the filterchains are connected. Libavfilter will automatically insert scale filters where format conversion is required. It is possible to specify swscale flags for those automatically inserted scalers by prepending "sws_flags= flags;" to the filtergraph description. Follows a BNF description for the filtergraph syntax:
<NAME> ::= sequence of alphanumeric characters and '_' <LINKLABEL> ::= "[" <NAME> "]" <LINKLABELS> ::= <LINKLABEL> [<LINKLABELS>] <FILTER_ARGUMENTS> ::= sequence of chars (eventually quoted) <FILTER> ::= [<LINKLABELS>] <NAME> ["=" <FILTER_ARGUMENTS>] [<LINKLABELS>] <FILTERCHAIN> ::= <FILTER> [,<FILTERCHAIN>] <FILTERGRAPH> ::= [sws_flags=<flags>;] <FILTERCHAIN> [;<FILTERGRAPH>]
AUDIO FILTERS¶
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. The filter accepts the following named 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.
aformat=sample_fmts=u8|s16:channel_layouts=stereo
amix¶
Mixes multiple audio inputs into a single output. For exampleavconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex amix=inputs=3:duration=first:dropout_transition=3 OUTPUTwill mix 3 input audio streams to a single output with the same duration as the first input and a dropout transition time of 3 seconds. The filter accepts the following named parameters:
- inputs
- Number of inputs. If unspecified, it defaults to 2.
- duration
- How to determine the end-of-stream.
- longest
- Duration of longest input. (default)
- shortest
- Duration of shortest input.
- first
- Duration of first input.
- dropout_transition
- Transition time, in seconds, for volume renormalization when an input stream ends. The default value is 2 seconds.
anull¶
Pass the audio source unchanged to the output.asetpts¶
Change the PTS (presentation timestamp) of the input audio frames. This filter accepts the following options:- expr
- The expression which is evaluated for each frame to construct its timestamp.
- PTS
- the presentation timestamp in input
- PI
- Greek PI
- PHI
- golden ratio
- E
- Euler number
- N
- Number of the audio samples pass through the filter so far, starting at 0.
- S
- Number of the audio samples in the current frame.
- SR
- Audio sample rate.
- STARTPTS
- the PTS of the first frame
- PREV_INPTS
- previous input PTS
- PREV_OUTPTS
- previous output PTS
- RTCTIME
- wallclock (RTC) time in microseconds
- RTCSTART
- wallclock (RTC) time at the start of the movie in microseconds
# 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)"
ashowinfo¶
Show a line containing various information for each input audio frame. The input audio is not modified. The shown line contains a sequence of key/value pairs of the form key:value. A description of each shown parameter follows:- n
- sequential number of the input frame, starting from 0
- pts
- Presentation timestamp of the input frame, in time base units; the time base depends on the filter input pad, and is usually 1/ sample_rate.
- pts_time
- presentation timestamp of the input frame in seconds
- fmt
- sample format
- chlayout
- channel layout
- rate
- sample rate for the audio frame
- nb_samples
- number of samples (per channel) in the frame
- checksum
- 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. The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2. For exampleavconv -i INPUT -filter_complex asplit=5 OUTPUTwill 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. The filter accepts the following named parameters:- compensate
- Enable stretching/squeezing the data to make it match the timestamps. Disabled by default. When disabled, time gaps are covered with silence.
- min_delta
- Minimum difference between timestamps and audio data (in seconds) to trigger adding/dropping samples. Default value is 0.1. If you get non-perfect sync with this filter, try setting this parameter to 0.
- max_comp
- Maximum compensation in samples per second. Relevant only with compensate=1. Default value 500.
- first_pts
- Assume the first pts should be this value. The time base is 1 / sample rate. This allows for padding/trimming at the start of stream. By default, no assumption is made about the first frame's expected pts, so no padding or trimming is done. For example, this could be set to 0 to pad the beginning with silence if an audio stream starts after the video stream 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. This filter accepts the following options:- start
- Timestamp (in seconds) of the start of the kept section. 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
- Maximum duration of the output in seconds.
- start_sample
- Number of the first sample that should be passed to output.
- end_sample
- Number of the first sample that should be dropped.
- •
- 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
channelsplit¶
Split each channel in input audio stream into a separate output stream. This filter accepts the following named parameters:- channel_layout
- Channel layout of the input stream. Default is "stereo".
avconv -i in.mp3 -filter_complex channelsplit out.mkvwill create an output Matroska file with two audio streams, one containing only the left channel and the other the right channel. To split a 5.1 WAV file into per-channel files
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. This filter accepts the following named parameters:- channel_layout
- Channel layout of the output stream.
- map
- Map channels from input to output. The argument is a '|'-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.
avconv -i in.mov -filter 'channelmap=map=DL-FL|DR-FR' out.wavwill 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 audio dynamic range. A description of the accepted options follows.- attacks
- decays
- Set 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
- Set 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|...."
- soft-knee
- Set the curve radius in dB for all joints. Defaults to 0.01.
- gain
- Set additional gain in dB to be applied at all points on the transfer function. This allows easy adjustment of the overall gain. Defaults to 0.
- volume
- Set 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. Defaults to 0.
- delay
- Set 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. Defaults to 0.
- •
- Make music with both quiet and loud passages suitable for
listening in a noisy environment:
compand=.3|.3:1|1:-90/-60|-60/-40|-40/-30|-20/-20:6:0:-90:0.2
- •
- 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. The filter accepts the following named parameters:- inputs
- Number of input streams. Defaults to 2.
- channel_layout
- Desired output channel layout. Defaults to stereo.
- map
- Map channels from inputs to output. The argument is a '|'-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.
avconv -i INPUT1 -i INPUT2 -i INPUT3 -filter_complex join=inputs=3 OUTPUTTo 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. This filter 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. The filter accepts the following named parameters:- volume
- Expresses how the audio volume will be increased or
decreased.
<output_volume> = <volume> * <input_volume>
- precision
- Mathematical precision.
- fixed
- 8-bit fixed-point; limits input sample format to U8, S16, and S32.
- float
- 32-bit floating-point; limits input sample format to FLT. (default)
- double
- 64-bit floating-point; limits input sample format to DBL.
- •
- 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
AUDIO SOURCES¶
Below is a description of the currently available audio sources.anullsrc¶
Null audio source, never return audio frames. It is mainly useful as a template and to be employed in analysis / debugging tools. It accepts as optional parameter a string of the form sample_rate:channel_layout. sample_rate specify the sample rate, and defaults to 44100. channel_layout specify 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. Follow some examples:# set the sample rate to 48000 Hz and the channel layout to CH_LAYOUT_MONO. anullsrc=48000:4 # same as 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 but for insertion by calling programs through the interface defined in libavfilter/buffersrc.h. It accepts the following named parameters:- time_base
- Timebase which will be used for timestamps of submitted frames. It must be either a floating-point number or in numerator/denominator form.
- sample_rate
- Audio sample rate.
- sample_fmt
- Name of the sample format, as returned by "av_get_sample_fmt_name()".
- channel_layout
- Channel layout of the audio data, in the form that can be accepted by "av_get_channel_layout()".
AUDIO SINKS¶
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 to be employed in analysis / debugging tools.abuffersink¶
This sink is intended for programmatic use. Frames that arrive on this sink can be retrieved by the calling program using the interface defined in libavfilter/buffersink.h. This filter accepts no parameters.VIDEO FILTERS¶
When you configure your 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. The filter accepts the following options:- amount
- The percentage of the pixels that have to be below the threshold, defaults to 98.
- threshold
- Threshold below which a pixel value is considered black, defaults to 32.
boxblur¶
Apply boxblur algorithm to the input video. This filter accepts the following options:- luma_radius
- luma_power
- chroma_radius
- chroma_power
- alpha_radius
- alpha_power
- w, h
- the input width and height in pixels
- cw, ch
- the input chroma image width and height in pixels
- hsub, vsub
- horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
- •
- Apply a boxblur filter with luma, chroma, and alpha radius
set to 2:
boxblur=luma_radius=2:luma_power=1
- •
- Set luma radius to 2, alpha and chroma radius to 0
boxblur=2:1:0:0:0:0
- •
- Set luma and chroma radius to a fraction of the video
dimension
boxblur=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. Mainly useful for testing purposes.crop¶
Crop the input video to given dimensions. This filter accepts the following options:- out_w
- Width of the output video.
- out_h
- Height of the output video.
- x
- Horizontal position, in the input video, of the left edge of the output video.
- y
- Vertical position, in the input video, of the top edge of the output video.
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), PHI (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
- same as in_w and in_h
- out_w, out_h
- the output (cropped) width and height
- ow, oh
- same as out_w and out_h
- n
- the number of input frame, starting from 0
- t
- timestamp expressed in seconds, NAN if the input timestamp is unknown
# 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 crop size. Calculate necessary cropping parameters and prints the recommended parameters through the logging system. The detected dimensions correspond to the non-black area of the input video. This filter accepts the following options:- limit
- Threshold, which can be optionally specified from nothing (0) to everything (255), defaults to 24.
- round
- Value which the width/height should be divisible by, defaults to 16. The offset is automatically adjusted to center the video. Use 2 to get only even dimensions (needed for 4:2:2 video). 16 is best when encoding to most video codecs.
- reset
- Counter that determines after how many frames cropdetect
will reset the previously detected largest video area and start over to
detect the current optimal crop area. Defaults to 0.
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). This filter accepts the following options:- 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.
- •
- Set a rectangle covering the area with top left corner
coordinates 0,0 and size 100x77, setting 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. This filter accepts the following options:- x, y
- Specify the top left corner coordinates of the box. Default to 0.
- width, height
- Specify the width and height of the box, if 0 they are interpreted as the input width and height. Default to 0.
- color
- Specify the color of the box to write, it can be the name of a color (case insensitive match) or a 0xRRGGBB[AA] sequence.
# 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 text string or text from specified file on top of video using the libfreetype library. To enable compilation of this filter you need to configure Libav with "--enable-libfreetype". The filter also recognizes strftime() sequences in the provided text and expands them accordingly. Check the documentation of strftime(). The description of the accepted parameters follows.- fontfile
- The font file to be used for drawing text. Path must be included. This parameter is mandatory.
- 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.
- x, y
- The offsets where text will be drawn within the video frame. Relative to the top/left border of the output image. They accept expressions similar to the overlay filter:
- x, y
- the computed values for x and y. They are evaluated for each new frame.
- main_w, main_h
- main input width and height
- W, H
- same as main_w and main_h
- text_w, text_h
- rendered text width and height
- w, h
- same as text_w and text_h
- n
- the number of frames processed, starting from 0
- t
- timestamp expressed in seconds, NAN if the input timestamp is unknown
- 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. 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. 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 background color. Value should 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. 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
- Flags to be used for loading the fonts.
- default
- no_scale
- no_hinting
- render
- no_bitmap
- vertical_layout
- force_autohint
- crop_bitmap
- pedantic
- ignore_global_advance_width
- no_recurse
- ignore_transform
- monochrome
- linear_design
- no_autohint
- end table
- 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.
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 fade-in/out effect to input video. This filter accepts the following options:- type
- The effect type -- can be either "in" for fade-in, or "out" for a fade-out effect.
- start_frame
- The number of the start frame for starting to apply the fade effect.
- nb_frames
- The number of frames for which the fade effect has to last. At the end of the fade-in effect the output video will have the same intensity as the input video, at the end of the fade-out transition the output video will be completely black.
# fade in first 30 frames of video fade=type=in:nb_frames=30 # fade out last 45 frames of a 200-frame video fade=type=out:start_frame=155:nb_frames=45 # fade in first 25 frames and fade out 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 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. This filter accepts the following options:- order
- Output field order. Valid values are tff for top field first or bff for bottom field first.
./avconv -i in.vob -vf "fieldorder=order=bff" out.dv
fifo¶
Buffer input images and send them when they are requested. This filter is mainly useful when auto-inserted by the libavfilter framework. The filter does not take parameters.format¶
Convert the input video to one of the specified pixel formats. Libavfilter will try to pick one that is supported for the input to the next filter. This filter accepts the following parameters:- pix_fmts
- A '|'-separated list of pixel format names, for example "pix_fmts=yuv420p|monow|rgb24".
# convert the input video to the format "yuv420p" 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. This filter accepts the following named parameters:- fps
- 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. This filter accepts the following named parameters:- format
- Desired packing format. Supported values are:
- sbs
- Views are next to each other (default).
- tab
- Views are on top of each other.
- lines
- Views are packed by line.
- columns
- Views are eacked by column.
- frameseq
- Views are temporally interleaved.
# 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 compilation of this filter you need to install the frei0r header and configure Libav with --enable-frei0r. This filter accepts the following options:- filter_name
- The name to the frei0r effect to load. If the environment variable FREI0R_PATH is defined, the frei0r effect is searched in each one of the directories specified by the colon separated list in FREIOR_PATH, otherwise in the standard frei0r paths, which are in this order: HOME/.frei0r-1/lib/, /usr/local/lib/frei0r-1/, /usr/lib/frei0r-1/.
- filter_params
- A '|'-separated list of parameters to pass to the frei0r effect.
# apply the distort0r effect, set the first two double parameters frei0r=filter_name=distort0r:filter_params=0.5|0.01 # apply the colordistance effect, takes a color as first parameter frei0r=colordistance:0.2/0.3/0.4 frei0r=colordistance:violet frei0r=colordistance:0x112233 # apply the perspective effect, specify the top left and top right # image positions frei0r=perspective:0.2/0.2|0.8/0.2For 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. This filter is designed for playback only. Do not use it prior to lossy compression, because compression tends to lose the dither and bring back the bands. This filter accepts the following options:- strength
- The maximum amount by which the filter will change any one pixel. Also the threshold for detecting nearly flat regions. Acceptable values range from .51 to 64, 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, default value is 16, out-of-range values will be clipped to the valid range.
# default parameters gradfun=strength=1.2:radius=16 # omitting 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¶
High precision/quality 3d denoise filter. This filter aims to reduce image noise producing smooth images and making still images really still. It should enhance compressibility. It accepts the following optional parameters:- luma_spatial
- a non-negative float number which specifies spatial luma strength, defaults to 4.0
- chroma_spatial
- a non-negative float number which specifies spatial chroma strength, defaults to 3.0* luma_spatial/4.0
- luma_tmp
- a float number which specifies luma temporal strength, defaults to 6.0* luma_spatial/4.0
- chroma_tmp
- a float number which specifies chroma temporal strength, defaults to luma_tmp*chroma_spatial/luma_spatial
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 onIt accepts the following optional parameters:
- scan
- 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 input video. lutyuv applies a lookup table to a YUV input video, lutrgb to an RGB input video. These filters accept the following options:- 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)
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), PHI (golden ratio)
- w, h
- the input width and height
- val
- input value for the pixel component
- clipval
- the input value clipped in the minval-maxval range
- maxval
- maximum value for the pixel component
- minval
- minimum value for the pixel component
- negval
- the negated value for the pixel component value clipped in the minval-maxval range , it corresponds to the expression "maxval-clipval+minval"
- clip(val)
- the computed value in val clipped in the minval- maxval range
- gammaval(gamma)
- the computed gamma correction value of the pixel component value clipped in the minval-maxval range, corresponds to the expression "pow((clipval-minval)/(maxval-minval), gamma)*(maxval-minval)+minval"
# 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, turns the video into a graytone image lutyuv="u=128:v=128" # apply a luma burning effect lutyuv="y=2*val" # remove green and blue components lutrgb="g=0:b=0" # set a constant alpha channel value on input format=rgba,lutrgb=a="maxval-minval/2" # correct luminance gamma by a 0.5 factor lutyuv=y=gammaval(0.5)
negate¶
Negate input video. This filter accepts an integer in input, if non-zero it negates the alpha component (if available). The default value in input is 0.noformat¶
Force libavfilter not to use any of the specified pixel formats for the input to the next filter. This filter accepts the following parameters:- pix_fmts
- A '|'-separated list of pixel format names, for example "pix_fmts=yuv420p|monow|rgb24".
# 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 video transform using libopencv. To enable this filter install libopencv library and headers and configure Libav with --enable-libopencv. This filter 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.
# use the default values ocv=dilate # dilate using a structuring element with a 5x5 cross, iterate two times ocv=filter_name=dilate:filter_params=5x5+2x2/cross|2 # read the shape from the file diamond.shape, iterate two times # the file diamond.shape may contain a pattern of characters like this: # * # *** # ***** # *** # * # the specified cols and rows are ignored (but not the anchor point coordinates) ocv=dilate:0x0+2x2/custom=diamond.shape|2erode Erode an image by using a specific structuring element. This filter corresponds to the libopencv function "cvErode". The filter accepts the parameters: struct_el:nb_iterations, with the same syntax and semantics as the dilate filter. smooth Smooth the input video. The filter takes the following parameters: type|param1|param2| param3|param4. type is the type of smooth filter to apply, and can be one of the following values: "blur", "blur_no_scale", "median", "gaussian", "bilateral". The default value is "gaussian". param1, param2, param3, and param4 are parameters whose meanings depend on smooth type. param1 and param2 accept integer positive values or 0, param3 and param4 accept float values. The default value for param1 is 3, the default value for the other parameters is 0. These parameters correspond to the parameters assigned to the libopencv function "cvSmooth".
overlay¶
Overlay one video on top of another. It takes two inputs and one output, the first input is the "main" video on which the second input is overlayed. This filter 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.
- main_w, main_h
- main input width and height
- W, H
- same as main_w and main_h
- overlay_w, overlay_h
- overlay input width and height
- w, h
- same as overlay_w and overlay_h
- eof_action
- The action to take when EOF is encountered on the secondary input, accepts one of the following values:
- repeat
- repeat the last frame (the default)
- endall
- end both streams
- pass
- pass through the main input
# 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.aviYou can chain together more overlays but the efficiency of such approach is yet to be tested.
pad¶
Add paddings to the input image, and places the original input at the given coordinates x, y. This filter 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.
- x, y
- Specify the offsets where to place the input image in the
padded area with respect to the top/left border of the output image.
- color
- Specify the color of the padded area, it can be the name of
a color (case insensitive match) or a 0xRRGGBB[AA] sequence.
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), phi (golden ratio)
- in_w, in_h
- the input video width and height
- iw, ih
- same as in_w and in_h
- out_w, out_h
- the output width and height, that is the size of the padded area as specified by the width and height expressions
- ow, oh
- same as out_w and out_h
- x, y
- x and y offsets as specified by the x and y expressions, or NAN if not yet specified
- a
- input display aspect ratio, same as iw / ih
- hsub, vsub
- horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
# Add paddings with color "violet" to the input video. Output video # size is 640x480, the top-left corner of the input video is placed at # column 0, row 40. pad=width=640:height=480:x=0:y=40:color=violet # pad the input to get an output with dimensions increased bt 3/2, # and put the input video at the center of the padded area pad="3/2*iw:3/2*ih:(ow-iw)/2:(oh-ih)/2" # pad the input to get a squared output with size equal to the maximum # value between the input width and height, and put the input video at # the center of the padded area pad="max(iw,ih):ow:(ow-iw)/2:(oh-ih)/2" # pad the input to get a final w/h ratio of 16:9 pad="ih*16/9:ih:(ow-iw)/2:(oh-ih)/2" # double output size and put the input video in the bottom-right # corner of the output padded area pad="2*iw:2*ih:ow-iw:oh-ih"
pixdesctest¶
Pixel format descriptor test filter, mainly useful for internal testing. The output video should be equal to the input video. For example:format=monow, pixdesctestcan be used to test the monowhite pixel format descriptor definition.
scale¶
Scale the input video and/or convert the image format. This filter accepts the following options:- w
- Output video width.
- h
- Output video height.
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), phi (golden ratio)
- in_w, in_h
- the input width and height
- iw, ih
- same as in_w and in_h
- out_w, out_h
- the output (cropped) width and height
- ow, oh
- same as out_w and out_h
- a
- same as iw / ih
- sar
- input sample aspect ratio
- dar
- input display aspect ratio, it is the same as (iw / ih) * sar
- hsub, vsub
- horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
# 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 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 for Greek harmony scale=iw:1/PHI*iw scale=ih*PHI:ih # increase the height, and set the width to 3/2 of the height scale=w=3/2*oh:h=3/5*ih # increase the size, but make the size a multiple of the chroma scale="trunc(3/2*iw/hsub)*hsub:trunc(3/2*ih/vsub)*vsub" # increase the width to a maximum of 500 pixels, keep the same input aspect ratio scale=w='min(500, iw*3/2):h=-1'
select¶
Select frames to pass in output. This filter accepts the following options:- 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.
- PI
- Greek PI
- PHI
- golden ratio
- E
- Euler number
- 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, NAN if undefined
- TB
- timebase of the input timestamps
- pts
- the PTS (Presentation TimeStamp) of the filtered video frame, expressed in TB units, NAN if undefined
- t
- the PTS (Presentation TimeStamp) of the filtered video frame, expressed in seconds, NAN if undefined
- prev_pts
- the PTS of the previously filtered video frame, NAN if undefined
- prev_selected_pts
- the PTS of the last previously filtered video frame, NAN if undefined
- prev_selected_t
- the PTS of the last previously selected video frame, NAN if undefined
- start_pts
- the PTS of the first video frame in the video, NAN if undefined
- start_t
- the time of the first video frame in the video, NAN if undefined
- pict_type
- the type of the filtered frame, can assume one of the following values:
- I
- P
- B
- S
- SI
- SP
- BI
- interlace_type
- the frame interlace type, can assume one of the following values:
- PROGRESSIVE
- the frame is progressive (not interlaced)
- TOPFIRST
- the frame is top-field-first
- BOTTOMFIRST
- the frame is bottom-field-first
- key
- 1 if the filtered frame is a key-frame, 0 otherwise
# select all 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 every 100 select='not(mod(n,100))' # select only frames contained in the 10-20 time interval select='gte(t,10)*lte(t,20)' # select only I frames contained in the 10-20 time interval select='gte(t,10)*lte(t,20)*eq(pict_type,I)' # select frames with a minimum distance of 10 seconds select='isnan(prev_selected_t)+gte(t-prev_selected_t,10)'
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. This filter accepts the following options:- dar
- Output display aspect ratio.
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), phi (golden ratio)
- w, h
- the input width and height
- a
- same as w / h
- sar
- input sample aspect ratio
- dar
- input display aspect ratio, it is the same as (w / h) * sar
- hsub, vsub
- horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
setdar=dar=16/9 # the above is equivalent to setdar=dar=1.77777See also the setsar filter documentation.
setpts¶
Change the PTS (presentation timestamp) of the input video frames. This filter accepts the following options:- expr
- The expression which is evaluated for each frame to construct its timestamp.
- PTS
- the presentation timestamp in input
- PI
- Greek PI
- PHI
- golden ratio
- E
- Euler number
- N
- the count of the input frame, starting from 0.
- STARTPTS
- the PTS of the first video frame
- INTERLACED
- tell if the current frame is interlaced
- PREV_INPTS
- previous input PTS
- PREV_OUTPTS
- previous output PTS
- RTCTIME
- wallclock (RTC) time in microseconds
- RTCSTART
- wallclock (RTC) time at the start of the movie in microseconds
- TB
- timebase of the input timestamps
# start counting 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. This filter accepts the following options:- sar
- Output sample aspect ratio.
- E, PI, PHI
- the corresponding mathematical approximated values for e (euler number), pi (greek PI), phi (golden ratio)
- w, h
- the input width and height
- a
- same as w / h
- sar
- input sample aspect ratio
- dar
- input display aspect ratio, it is the same as (w / h) * sar
- hsub, vsub
- horizontal and vertical chroma subsample values. For example for the pixel format "yuv422p" hsub is 2 and vsub is 1.
setsar=sar=10/11
settb¶
Set the timebase to use for the output frames timestamps. It is mainly useful for testing timebase configuration. This filter accepts the following options:- expr
- The expression which is evaluated into the output timebase.
# 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. A description of each shown parameter follows:- n
- sequential number of the input frame, starting from 0
- pts
- Presentation TimeStamp of the input frame, expressed as a number of time base units. The time base unit depends on the filter input pad.
- pts_time
- Presentation TimeStamp of the input frame, expressed as a number of seconds
- pos
- position of the frame in the input stream, -1 if this information in unavailable and/or meaningless (for example in case of synthetic video)
- fmt
- pixel format name
- sar
- sample aspect ratio of the input frame, expressed in the form num/den
- s
- size of the input frame, expressed in the form widthx height
- i
- interlaced mode ("P" for "progressive", "T" for top field first, "B" for bottom field first)
- iskey
- 1 if the frame is a key frame, 0 otherwise
- type
- picture type of the input frame ("I" for an I-frame, "P" for a P-frame, "B" for a B-frame, "?" for unknown type). Check also the documentation of the "AVPictureType" enum and of the "av_get_picture_type_char" function defined in libavutil/avutil.h.
- checksum
- Adler-32 checksum of all the planes of the input frame
- plane_checksum
- Adler-32 checksum of each plane of the input frame, expressed in the form "[ c0 c1 c2 c3]"
split¶
Split input video into several identical outputs. The filter accepts a single parameter which specifies the number of outputs. If unspecified, it defaults to 2. For exampleavconv -i INPUT -filter_complex split=5 OUTPUTwill create 5 copies of the input video.
transpose¶
Transpose rows with columns in the input video and optionally flip it. This filter accepts the following options:- dir
- The direction of the transpose.
- 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. This filter accepts the following options:- start
- Timestamp (in seconds) of the start of the kept section. I.e. the frame with the timestamp start will be the first frame in the output.
- end
- Timestamp (in seconds) of the first frame that will be dropped. I.e. the frame immediately preceding the one with the timestamp end will be the last frame in the output.
- start_pts
- Same as start, except this option sets the start timestamp in timebase units instead of seconds.
- end_pts
- Same as end, except this option sets the end timestamp in timebase units instead of seconds.
- duration
- Maximum duration of the output in seconds.
- start_frame
- Number of the first frame that should be passed to output.
- end_frame
- Number of the first frame that should be dropped.
- •
- 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 can be an integer between 3 and 13, default value is 5.
- luma_msize_y
- Set the luma matrix vertical size. It can be an integer between 3 and 13, default value is 5.
- luma_amount
- Set the luma effect strength. It can be a float number between -2.0 and 5.0, default value is 1.0.
- chroma_msize_x
- Set the chroma matrix horizontal size. It can be an integer between 3 and 13, default value is 5.
- chroma_msize_y
- Set the chroma matrix vertical size. It can be an integer between 3 and 13, default value is 5.
- chroma_amount
- Set the chroma effect strength. It can be a float number between -2.0 and 5.0, default value is 0.0.
# Strong luma sharpen effect parameters unsharp=luma_msize_x=7:luma_msize_y=7:luma_amount=2.5 # 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"). This filter accepts the following options:- mode
- The interlacing mode to adopt, accepts one of the following values:
- 0
- output 1 frame for each frame
- 1
- output 1 frame for each field
- 2
- like 0 but skips spatial interlacing check
- 3
- like 1 but skips spatial interlacing check
- parity
- The picture field parity assumed for the input interlaced video, accepts one of the following values:
- 0
- assume top field first
- 1
- assume bottom field first
- -1
- enable automatic detection
- auto
- Whether deinterlacer should trust the interlaced flag and only deinterlace frames marked as interlaced
- 0
- deinterlace all frames
- 1
- only deinterlace frames marked as interlaced
VIDEO SOURCES¶
Below is a description of the currently available video sources.buffer¶
Buffer video frames, and make them available to the filter chain. This source is mainly intended for a programmatic use, in particular through the interface defined in libavfilter/vsrc_buffer.h. This filter accepts the following parameters:- width
- Input video width.
- height
- Input video height.
- pix_fmt
- Name of the input video pixel format.
- time_base
- The time base used for input timestamps.
- sar
- Sample (pixel) aspect ratio of the input video.
buffer=width=320:height=240:pix_fmt=yuv410p:time_base=1/24:sar=1will 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 widthxheight, 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 float number or a valid video frame rate abbreviation. The default value is "25".
"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. Specifically in avconv this filter should never be used, the -filter_complex option fully replaces it. This filter accepts the following options:- filename
- The name of the resource to read (not necessarily a file but also 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. Default value is "0".
- stream_index, si
- Specifies the index of the video stream to read. If the value is -1, the best suited video stream will be automatically selected. Default value is "-1".
input -----------> deltapts0 --> overlay --> output ^ | movie --> scale--> deltapts1 -------+Some examples follow:
# skip 3.2 seconds from the start of the avi file in.avi, and overlay it # on top of the input labelled as "in". movie=in.avi:seek_point=3.2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out] # read from a video4linux2 device, and overlay it on top of the input # labelled as "in" movie=/dev/video0:f=video4linux2, scale=180:-1, setpts=PTS-STARTPTS [movie]; [in] setpts=PTS-STARTPTS, [movie] overlay=16:16 [out]
nullsrc¶
Null video source, never return images. It is mainly useful as a template and to be employed in analysis / debugging tools. It accepts as optional parameter a string of the form width:height: timebase. 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", "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 options:- size
- The size of the video to generate, may be a string of the form widthxheight or a frame size abbreviation.
- framerate
- Framerate of the generated video, 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 section frei0r in the description of the video filters.
- filter_params
- A '|'-separated list of parameters to pass to the frei0r source.
# 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 options:- size, s
- Specify the size of the sourced video, it may be a string of the form widthxheight, or the name of a size abbreviation. The default value is "320x240".
- rate, r
- Specify the frame rate of the sourced video, as the number of frames generated per second. It has to be a string in the format frame_rate_num/ frame_rate_den, an integer number, a float number or a valid video frame rate abbreviation. The default value is "25".
- 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...]
testsrc=duration=5.3:size=qcif:rate=10will generate a video with a duration of 5.3 seconds, with size 176x144 and a framerate of 10 frames per second.
VIDEO SINKS¶
Below is a description of the currently available video sinks.buffersink¶
Buffer video frames, and make them available to the end of the filter graph. This sink is intended for a 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 to be employed in analysis / debugging tools.SEE ALSO¶
avconv(1), avprobe(1) and the Libav HTML documentationAUTHORS¶
The Libav developers2014-06-01 |