The generic syntax is:
avprobe [options] [‘input_file’] |
avprobe gathers information from multimedia streams and prints it in human- and machine-readable fashion.
For example it can be used to check the format of the container used by a multimedia stream and the format and type of each media stream contained in it.
If a filename is specified in input, avprobe will try to open and probe the file content. If the file cannot be opened or recognized as a multimedia file, a positive exit code is returned.
avprobe may be employed both as a standalone application or in combination with a textual filter, which may perform more sophisticated processing, e.g. statistical processing or plotting.
Options are used to list some of the formats supported by avprobe or for specifying which information to display, and for setting how avprobe will show it.
avprobe output is designed to be easily parsable by any INI or JSON parsers.
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".
Some options are applied per-stream, e.g. bitrate or codec. Stream specifiers are used to precisely specify which stream(s) does a given option belong to.
A stream specifier is a string generally appended to the option name and
separated from it by a colon. E.g. -codec:a:1 ac3 option contains
a:1 stream specifier, which matches the second audio stream. Therefore it
would select the ac3 codec for the second audio stream.
A stream specifier can match several stream, the option is then applied to all
of them. E.g. the stream specifier in -b:a 128k matches all audio
streams.
An empty stream specifier matches all streams, for example -codec copy
or -codec: copy would copy all the streams without reencoding.
Possible forms of stream specifiers are:
Matches the stream with this index. E.g. -threads:1 4 would set the
thread count for the second stream to 4.
stream_type is one of: ’v’ for video, ’a’ for audio, ’s’ for subtitle, ’d’ for data and ’t’ for attachments. If stream_index is given, then matches stream number stream_index of this type. Otherwise matches all streams of this type.
If stream_index is given, then matches stream number stream_index in program with id program_id. Otherwise matches all streams in this program.
Match the stream by stream id (e.g. PID in MPEG-TS container).
Matches streams with the metadata tag key having the specified value. If value is not given, matches streams that contain the given tag with any value.
Matches streams with usable configuration, the codec must be defined and the essential information such as video dimension or audio sample rate must be present.
Note that in avconv, matching by metadata will only work properly for
input files.
These options are shared amongst the av* tools.
Show license.
Show help. An optional parameter may be specified to print help about a specific item.
Possible values of arg are:
Print detailed information about the decoder named decoder_name. Use the ‘-decoders’ option to get a list of all decoders.
Print detailed information about the encoder named encoder_name. Use the ‘-encoders’ option to get a list of all encoders.
Print detailed information about the demuxer named demuxer_name. Use the ‘-formats’ option to get a list of all demuxers and muxers.
Print detailed information about the muxer named muxer_name. Use the ‘-formats’ option to get a list of all muxers and demuxers.
Print detailed information about the filter name filter_name. Use the ‘-filters’ option to get a list of all filters.
Show version.
Show available formats.
The fields preceding the format names have the following meanings:
Decoding available
Encoding available
Show all codecs known to libavcodec.
Note that the term ’codec’ is used throughout this documentation as a shortcut for what is more correctly called a media bitstream format.
Show available decoders.
Show all available encoders.
Show available bitstream filters.
Show available protocols.
Show available libavfilter filters.
Show available pixel formats.
Show available sample formats.
Set the logging level used by the library. loglevel is a number or a string containing one of the following values:
By default the program logs to stderr, if coloring is supported by the
terminal, colors are used to mark errors and warnings. Log coloring
can be disabled setting the environment variable
AV_LOG_FORCE_NOCOLOR or NO_COLOR, or can be forced setting
the environment variable AV_LOG_FORCE_COLOR.
The use of the environment variable NO_COLOR is deprecated and
will be dropped in a following Libav version.
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.
These options are provided directly by the libavformat, libavdevice and libavcodec libraries. To see the list of available AVOptions, use the ‘-help’ option. They are separated into two categories:
These options can be set for any container, codec or device. Generic options are listed under AVFormatContext options for containers/devices and under AVCodecContext options for codecs.
These options are specific to the given container, device or codec. Private options are listed under their corresponding containers/devices/codecs.
For example to write an ID3v2.3 header instead of a default ID3v2.4 to an MP3 file, use the ‘id3v2_version’ private option of the MP3 muxer:
avconv -i input.flac -id3v2_version 3 out.mp3 |
All codec AVOptions are obviously per-stream, so the chapter on stream specifiers applies to them
Note ‘-nooption’ syntax cannot be used for boolean AVOptions, use ‘-option 0’/‘-option 1’.
Note2 old undocumented way of specifying per-stream AVOptions by prepending v/a/s to the options name is now obsolete and will be removed soon.
set bitrate (in bits/s)
Set video bitrate tolerance (in bits/s). In 1-pass mode, bitrate tolerance specifies how far ratecontrol is willing to deviate from the target average bitrate value. This is not related to minimum/maximum bitrate. Lowering tolerance too much has an adverse effect on quality.
Possible values:
allow decoders to produce unaligned output
use four motion vectors per macroblock (MPEG-4)
use 1/4-pel motion compensation
use loop filter
use fixed qscale
use internal 2-pass ratecontrol in first pass mode
use internal 2-pass ratecontrol in second pass mode
only decode/encode grayscale
error[?] variables will be set during encoding
use interlaced DCT
force low delay
place global headers in extradata instead of every keyframe
use only bitexact functions (except (I)DCT)
H.263 advanced intra coding / MPEG-4 AC prediction
interlaced motion estimation
closed GOP
Output even potentially corrupted frames
set the group of picture (GOP) size
set audio sampling rate (in Hz)
set number of audio channels
set cutoff bandwidth
video quantizer scale compression (VBR). Constant of ratecontrol equation. Recommended range for default rc_eq: 0.0-1.0
video quantizer scale blur (VBR)
minimum video quantizer scale (VBR)
maximum video quantizer scale (VBR)
maximum difference between the quantizer scales (VBR)
use ’frames’ B-frames
QP factor between P- and B-frames
strategy to choose between I/P/B-frames
RTP payload size in bytes
work around not autodetected encoder bugs
Possible values:
Xvid interlacing bug (autodetected if FOURCC == XVIX)
(autodetected if FOURCC == UMP4)
padding bug (autodetected)
old standard qpel (autodetected per FOURCC/version)
direct-qpel-blocksize bug (autodetected per FOURCC/version)
edge padding bug (autodetected per FOURCC/version)
work around various bugs in Microsoft’s broken decoders
truncated frames
how strictly to follow the standards
Possible values:
strictly conform to a older more strict version of the spec or reference software
strictly conform to all the things in the spec no matter what the consequences
allow unofficial extensions
allow non-standardized experimental things
QP offset between P- and B-frames
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
use MPEG quantizers instead of H.263
Set maximum bitrate tolerance (in bits/s). Requires bufsize to be set.
Set minimum bitrate tolerance (in bits/s). Most useful in setting up a CBR encode. It is of little use otherwise.
set ratecontrol buffer size (in bits)
QP factor between P- and I-frames
QP offset between P- and I-frames
DCT algorithm
Possible values:
autoselect a good one (default)
fast integer
accurate integer
floating point AAN DCT
compresses bright areas stronger than medium ones
temporal complexity masking
spatial complexity masking
inter masking
compresses dark areas stronger than medium ones
select IDCT implementation
Possible values:
floating point AAN IDCT
set error concealment strategy
Possible values:
iterative motion vector (MV) search (slow)
use strong deblock filter for damaged MBs
prediction method
Possible values:
sample aspect ratio
print specific debug info
Possible values:
picture info
rate control
macroblock (MB) type
per-block quantization parameter (QP)
error recognition
memory management control operations (H.264)
picture buffer allocations
threading operations
full-pel ME compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
sub-pel ME compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
macroblock compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
interlaced DCT compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation
amount of motion predictors from the previous frame
pre motion estimation
pre motion estimation compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
diamond type & size for motion estimation pre-pass
sub-pel motion estimation quality
limit motion vectors range (1023 for DivX player)
Possible values:
variable length coder / Huffman coder
arithmetic coder
raw (no encoding)
run-length coder
context model
macroblock decision algorithm (high quality mode)
Possible values:
use mbcmp (default)
use fewest bits
use best rate distortion
scene change threshold
noise reduction
number of bits which should be loaded into the rc buffer before decoding starts
Possible values:
allow non-spec-compliant speedup tricks
skip bitstream encoding
ignore cropping information from sps
place global headers at every keyframe instead of in extradata
Possible values:
autodetect a suitable number of threads to use
intra_dc_precision
nsse weight
number of macroblock rows at the top which are skipped
number of macroblock rows at the bottom which are skipped
Possible values:
Possible values:
frame skip threshold
frame skip factor
frame skip exponent
frame skip compare function
Possible values:
sum of absolute differences, fast (default)
sum of squared errors
sum of absolute Hadamard transformed differences
sum of absolute DCT transformed differences
sum of squared quantization errors (avoid, low quality)
number of bits needed for the block
rate distortion optimal, slow
0
sum of absolute vertical differences
sum of squared vertical differences
noise preserving sum of squared differences
minimum macroblock Lagrange factor (VBR)
maximum macroblock Lagrange factor (VBR)
motion estimation bitrate penalty compensation (1.0 = 256)
Possible values:
Possible values:
Possible values:
refine the two motion vectors used in bidirectional macroblocks
downscale frames for dynamic B-frame decision
minimum interval between IDR-frames (x264)
reference frames to consider for motion compensation
chroma QP offset from luma
rate-distortion optimal quantization
adjust sensitivity of b_frame_strategy 1
GOP timecode frame start number, in non-drop-frame format
Possible values:
Possible values:
color primaries
Possible values:
BT.709
Unspecified
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Film
BT.2020
SMPTE 428-1
SMPTE 431-2
SMPTE 422-1
JEDEC P22
Unspecified
SMPTE 428-1
color transfer characteristics
Possible values:
BT.709
Unspecified
BT.470 M
BT.470 BG
SMPTE 170 M
SMPTE 240 M
Linear
Log
Log square root
IEC 61966-2-4
BT.1361
IEC 61966-2-1
BT.2020 - 10 bit
BT.2020 - 12 bit
SMPTE 2084
SMPTE 428-1
ARIB STD-B67
Unspecified
Log
Log square root
IEC 61966-2-4
BT.1361
IEC 61966-2-1
BT.2020 - 10 bit
BT.2020 - 12 bit
SMPTE 2084
SMPTE 428-1
color space
Possible values:
RGB
BT.709
Unspecified
FCC
BT.470 BG
SMPTE 170 M
SMPTE 240 M
YCGCO
BT.2020 NCL
BT.2020 CL
SMPTE 2085
Unspecified
YCGCO
BT.2020 NCL
BT.2020 CL
color range
Possible values:
Unspecified
MPEG (219*2^(n-8))
JPEG (2^n-1)
Unspecified
MPEG (219*2^(n-8))
JPEG (2^n-1)
chroma sample location
Possible values:
Unspecified
Left
Center
Top-left
Top
Bottom-left
Bottom
Unspecified
number of slices, used in parallelized encoding
select multithreading type
Possible values:
audio service type
Possible values:
Main Audio Service
Effects
Visually Impaired
Hearing Impaired
Dialogue
Commentary
Emergency
Voice Over
Karaoke
Possible values:
8-bit unsigned integer
16-bit signed integer
32-bit signed integer
32-bit float
64-bit double
8-bit unsigned integer planar
16-bit signed integer planar
32-bit signed integer planar
32-bit float planar
64-bit double planar
Number of extra hardware frames to allocate for the user
set probing size
set packet size
Possible values:
reduce the latency by flushing out packets immediately
ignore index
generate pts
do not fill in missing values that can be exactly calculated
disable AVParsers, this needs nofillin too
ignore dts
discard corrupted frames
reduce the latency introduced by optional buffering
do not write random/volatile data
how many microseconds are analyzed to estimate duration
decryption key
max memory used for timestamp index (per stream)
max memory used for buffering real-time frames
print specific debug info
Possible values:
maximum muxing or demuxing delay in microseconds
number of frames used to probe fps
set error detection flags (deprecated; use err_detect, save via avconv)
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
set error detection flags
Possible values:
verify embedded CRCs
detect bitstream specification deviations
detect improper bitstream length
abort decoding on minor error detection
maximum buffering duration for interleaving
how strictly to follow the standards (deprecated; use strict, save via avconv)
Possible values:
strictly conform to all the things in the spec no matter what the consequences
allow non-standardized experimental variants
how strictly to follow the standards
Possible values:
strictly conform to all the things in the spec no matter what the consequences
allow non-standardized experimental variants
maximum number of packets to read while waiting for the first timestamp
shift timestamps so they start at 0
Possible values:
enabled when required by target format
shift timestamps so they are non negative
shift timestamps so they start at 0
A comma-separated list of blacklisted protocols used for opening files internally by lavf
A comma-separated list of whitelisted protocols used for opening files internally by lavf
Force format to use.
Use a specific formatter to output the document. The following formatters are available
Pseudo-INI format that used to be the only one available in old avprobe versions.
Show the unit of the displayed values.
Use SI prefixes for the displayed values. Unless the "-byte_binary_prefix" option is used all the prefixes are decimal.
Force the use of binary prefixes for byte values.
Use sexagesimal format HH:MM:SS.MICROSECONDS for time values.
Prettify the format of the displayed values, it corresponds to the options "-unit -prefix -byte_binary_prefix -sexagesimal".
Show information about the container format of the input multimedia stream.
All the container format information is printed within a section with name "FORMAT".
Like ‘-show_format’, but only prints the specified entry of the container format information, rather than all. This option may be given more than once, then all specified entries will be shown.
Show information about each packet contained in the input multimedia stream.
The information for each single packet is printed within a dedicated section with name "PACKET".
Show information about each media stream contained in the input multimedia stream.
Each media stream information is printed within a dedicated section with name "STREAM".
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.
Image file demuxer.
This demuxer reads from a list of image files specified by a pattern.
The pattern may contain the string "%d" or "%0Nd", which specifies the position of the characters representing a sequential number in each filename matched by the pattern. If the form "%d0Nd" is used, the string representing the number in each filename is 0-padded and N is the total number of 0-padded digits representing the number. The literal character ’%’ can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified by the pattern must contain a number inclusively contained between 0 and 4, all the following numbers must be sequential. This limitation may be hopefully fixed.
The pattern may contain a suffix which is used to automatically determine the format of the images contained in the files.
For example the pattern "img-%03d.bmp" will match a sequence of filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ..., ‘img-010.bmp’, etc.; the pattern "i%%m%%g-%d.jpg" will match a sequence of filenames of the form ‘i%m%g-1.jpg’, ‘i%m%g-2.jpg’, ..., ‘i%m%g-10.jpg’, etc.
The size, the pixel format, and the format of each image must be the same for all the files in the sequence.
The following example shows how to use avconv for creating a
video from the images in the file sequence ‘img-001.jpeg’,
‘img-002.jpeg’, ..., assuming an input framerate of 10 frames per
second:
avconv -i 'img-%03d.jpeg' -r 10 out.mkv |
Note that the pattern must not necessarily contain "%d" or "%0Nd", for example to convert a single image file ‘img.jpeg’ you can employ the command:
avconv -i img.jpeg img.png |
Set the pixel format (for raw image)
Set the frame size (for raw image)
Set the frame rate
Loop over the images
Specify the first number in the sequence
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".
Adobe Flash Video Format demuxer.
This demuxer is used to demux FLV files and RTMP network streams.
Allocate the streams according to the onMetaData array content.
Advanced Systems Format demuxer.
This demuxer is used to demux ASF files and MMS network streams.
Do not try to resynchronize by looking for a certain optional start code.
Muxers are configured elements in Libav which allow writing multimedia streams to a particular type of file.
When you configure your Libav build, all the supported muxers
are enabled by default. You can list all available muxers using the
configure option --list-muxers.
You can disable all the muxers with the configure option
--disable-muxers and selectively enable / disable single muxers
with the options --enable-muxer=MUXER /
--disable-muxer=MUXER.
The option -formats of the av* tools will display the list of
enabled muxers.
A description of some of the currently available muxers follows.
CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC of all the input audio and video frames. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a single line of the form: CRC=0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC for all the decoded input frames.
For example to compute the CRC of the input, and store it in the file ‘out.crc’:
avconv -i INPUT -f crc out.crc |
You can print the CRC to stdout with the command:
avconv -i INPUT -f crc - |
You can select the output format of each frame with avconv by
specifying the audio and video codec and format. For example to
compute the CRC of the input audio converted to PCM unsigned 8-bit
and the input video converted to MPEG-2 video, use the command:
avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f crc - |
See also the framecrc muxer.
Dynamic Adaptive Streaming over HTTP (DASH) muxer that creates segments and manifest files according to the MPEG-DASH standard ISO/IEC 23009-1:2014.
For more information see:
It creates a MPD manifest file and segment files for each stream.
The segment filename might contain pre-defined identifiers used with SegmentTemplate as defined in section 5.3.9.4.4 of the standard. Available identifiers are "$RepresentationID$", "$Number$", "$Bandwidth$" and "$Time$".
avconv -re -i <input> -map 0 -map 0 -c:a libfdk_aac -c:v libx264 -b:v:0 800k -b:v:1 300k -s:v:1 320x170 -profile:v:1 baseline -profile:v:0 main -bf 1 -keyint_min 120 -g 120 -sc_threshold 0 -b_strategy 0 -ar:a:1 22050 -use_timeline 1 -use_template 1 -window_size 5 -adaptation_sets "id=0,streams=v id=1,streams=a" -f dash /path/to/out.mpd |
Set the segment length in microseconds.
Set the maximum number of segments kept in the manifest.
Set the maximum number of segments kept outside of the manifest before removing from disk.
Enable (1) or disable (0) removal of all segments when finished.
Enable (1) or disable (0) use of SegmentTemplate instead of SegmentList.
Enable (1) or disable (0) use of SegmentTimeline in SegmentTemplate.
Enable (1) or disable (0) storing all segments in one file, accessed using byte ranges.
DASH-templated name to be used for baseURL. Implies single_file set to "1".
DASH-templated name to used for the initialization segment. Default is "init-stream$RepresentationID$.m4s"
DASH-templated name to used for the media segments. Default is "chunk-stream$RepresentationID$-$Number%05d$.m4s"
URL of the page that will return the UTC timestamp in ISO format. Example: "https://time.akamai.com/?iso"
Assign streams to AdaptationSets. Syntax is "id=x,streams=a,b,c id=y,streams=d,e" with x and y being the IDs of the adaptation sets and a,b,c,d and e are the indices of the mapped streams.
To map all video (or audio) streams to an AdaptationSet, "v" (or "a") can be used as stream identifier instead of IDs.
When no assignment is defined, this defaults to an AdaptationSet for each stream.
Per-frame CRC (Cyclic Redundancy Check) testing format.
This muxer computes and prints the Adler-32 CRC for each decoded audio and video frame. By default audio frames are converted to signed 16-bit raw audio and video frames to raw video before computing the CRC.
The output of the muxer consists of a line for each audio and video frame of the form: stream_index, frame_dts, frame_size, 0xCRC, where CRC is a hexadecimal number 0-padded to 8 digits containing the CRC of the decoded frame.
For example to compute the CRC of each decoded frame in the input, and store it in the file ‘out.crc’:
avconv -i INPUT -f framecrc out.crc |
You can print the CRC of each decoded frame to stdout with the command:
avconv -i INPUT -f framecrc - |
You can select the output format of each frame with avconv by
specifying the audio and video codec and format. For example, to
compute the CRC of each decoded input audio frame converted to PCM
unsigned 8-bit and of each decoded input video frame converted to
MPEG-2 video, use the command:
avconv -i INPUT -c:a pcm_u8 -c:v mpeg2video -f framecrc - |
See also the crc muxer.
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.
Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.
avconv -i in.mkv -c:v h264 -flags +cgop -g 30 -hls_time 1 out.m3u8 |
Set the segment length in seconds.
Set the maximum number of playlist entries.
Set the number after which index wraps.
Start the sequence from number.
Append baseurl to every entry in the playlist. Useful to generate playlists with absolute paths.
Explicitly set whether the client MAY (1) or MUST NOT (0) cache media segments
Set the protocol version. Enables or disables version-specific features such as the integer (version 2) or decimal EXTINF values (version 3).
Enable (1) or disable (0) the AES128 encryption. When enabled every segment generated is encrypted and the encryption key is saved as playlist name.key.
Use the specified hex-coded 16byte key to encrypt the segments, by default it is randomly generated.
If set, keyurl is prepended instead of baseurl to the key filename in the playlist.
Use a specified hex-coded 16byte initialization vector for every segment instead of the autogenerated ones.
Image file muxer.
The image file muxer writes video frames to image files.
The output filenames are specified by a pattern, which can be used to produce sequentially numbered series of files. The pattern may contain the string "%d" or "%0Nd", this string specifies the position of the characters representing a numbering in the filenames. If the form "%0Nd" is used, the string representing the number in each filename is 0-padded to N digits. The literal character ’%’ can be specified in the pattern with the string "%%".
If the pattern contains "%d" or "%0Nd", the first filename of the file list specified will contain the number 1, all the following numbers will be sequential.
The pattern may contain a suffix which is used to automatically determine the format of the image files to write.
For example the pattern "img-%03d.bmp" will specify a sequence of filenames of the form ‘img-001.bmp’, ‘img-002.bmp’, ..., ‘img-010.bmp’, etc. The pattern "img%%-%d.jpg" will specify a sequence of filenames of the form ‘img%-1.jpg’, ‘img%-2.jpg’, ..., ‘img%-10.jpg’, etc.
The following example shows how to use 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 "%0Nd", 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 the sequence from 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 container muxer.
This muxer implements the matroska and webm container specs.
The recognized metadata settings in this muxer are:
Name provided to a single track
Specifies the language of the track in the Matroska languages form
Stereo 3D video layout of two views in a single video track
video is not stereo
Both views are arranged side by side, Left-eye view is on the left
Both views are arranged in top-bottom orientation, Left-eye view is at bottom
Both views are arranged in top-bottom orientation, Left-eye view is on top
Each view is arranged in a checkerboard interleaved pattern, Left-eye view being first
Each view is arranged in a checkerboard interleaved pattern, Right-eye view being first
Each view is constituted by a row based interleaving, Right-eye view is first row
Each view is constituted by a row based interleaving, Left-eye view is first row
Both views are arranged in a column based interleaving manner, Right-eye view is first column
Both views are arranged in a column based interleaving manner, Left-eye view is first column
All frames are in anaglyph format viewable through red-cyan filters
Both views are arranged side by side, Right-eye view is on the left
All frames are in anaglyph format viewable through green-magenta filters
Both eyes laced in one Block, Left-eye view is first
Both eyes laced in one Block, Right-eye view is first
For example a 3D WebM clip can be created using the following command line:
avconv -i sample_left_right_clip.mpg -an -c:v libvpx -metadata STEREO_MODE=left_right -y stereo_clip.webm |
This muxer supports the following options:
By default, this muxer writes the index for seeking (called cues in Matroska terms) at the end of the file, because it cannot know in advance how much space to leave for the index at the beginning of the file. However for some use cases – e.g. streaming where seeking is possible but slow – it is useful to put the index at the beginning of the file.
If this option is set to a non-zero value, the muxer will reserve a given amount of space in the file header and then try to write the cues there when the muxing finishes. If the available space does not suffice, muxing will fail. A safe size for most use cases should be about 50kB per hour of video.
Note that cues are only written if the output is seekable and this option will have no effect if it is not.
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:
Start a new fragment at each video keyframe.
Create fragments that are duration microseconds long.
Create fragments that contain up to size bytes of payload data.
Allow the caller to manually choose when to cut fragments, by
calling av_write_frame(ctx, NULL) to write a fragment with
the packets written so far. (This is only useful with other
applications integrating libavformat, not from avconv.)
Don’t create fragments that are shorter than duration microseconds long.
If more than one condition is specified, fragments are cut when
one of the specified conditions is fulfilled. The exception to this is
-min_frag_duration, which has to be fulfilled for any of the other
conditions to apply.
Additionally, the way the output file is written can be adjusted through a few other options:
Write an initial moov atom directly at the start of the file, without describing any samples in it. Generally, an mdat/moov pair is written at the start of the file, as a normal MOV/MP4 file, containing only a short portion of the file. With this option set, there is no initial mdat atom, and the moov atom only describes the tracks but has a zero duration.
This option is implicitly set when writing ismv (Smooth Streaming) files.
Write a separate moof (movie fragment) atom for each track. Normally, packets for all tracks are written in a moof atom (which is slightly more efficient), but with this option set, the muxer writes one moof/mdat pair for each track, making it easier to separate tracks.
This option is implicitly set when writing ismv (Smooth Streaming) files.
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.
Disable Nero chapter markers (chpl atom). Normally, both Nero chapters and a QuickTime chapter track are written to the file. With this option set, only the QuickTime chapter track will be written. Nero chapters can cause failures when the file is reprocessed with certain tagging programs.
Do not write any absolute base_data_offset in tfhd atoms. This avoids tying fragments to absolute byte positions in the file/streams.
Similarly to the omit_tfhd_offset, this flag avoids writing the absolute base_data_offset field in tfhd atoms, but does so by using the new default-base-is-moof flag instead. This flag is new from 14496-12:2012. This may make the fragments easier to parse in certain circumstances (avoiding basing track fragment location calculations on the implicit end of the previous track fragment).
Smooth Streaming content can be pushed in real time to a publishing point on IIS with this muxer. Example:
avconv -re <normal input/transcoding options> -movflags isml+frag_keyframe -f ismv http://server/publishingpoint.isml/Streams(Encoder1) |
The MP3 muxer writes a raw MP3 stream with the following optional features:
id3v2_version private option controls which one is
used (3 or 4). Setting id3v2_version to 0 disables the ID3v2 header
completely.
The muxer supports writing attached pictures (APIC frames) to the ID3v2 header. 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.
write_xing private option can be used to disable it. The frame contains
various information that may be useful to the decoder, like the audio duration
or encoder delay.
write_id3v1 private option, but as its capabilities are
very limited, its usage is not recommended.
Examples:
Write an mp3 with an ID3v2.3 header and an ID3v1 footer:
avconv -i INPUT -id3v2_version 3 -write_id3v1 1 out.mp3 |
Attach a picture to an mp3:
avconv -i input.mp3 -i cover.png -c copy -metadata:s:v title="Album cover" -metadata:s:v comment="Cover (Front)" out.mp3 |
Write a "clean" MP3 without any extra features:
avconv -i input.wav -write_xing 0 -id3v2_version 0 out.mp3 |
MPEG transport stream muxer.
This muxer implements ISO 13818-1 and part of ETSI EN 300 468.
The muxer options are:
Set the original_network_id (default 0x0001). This is unique identifier of a network in DVB. Its main use is in the unique identification of a service through the path Original_Network_ID, Transport_Stream_ID.
Set the transport_stream_id (default 0x0001). This identifies a transponder in DVB.
Set the service_id (default 0x0001) also known as program in DVB.
Set the first PID for PMT (default 0x1000, max 0x1f00).
Set the first PID for data packets (default 0x0100, max 0x0f00).
Set a constant muxrate (default VBR).
Override the default PCR retransmission time (default 20ms), ignored if variable muxrate is selected.
The recognized metadata settings in mpegts muxer are service_provider
and service_name. If they are not set the default for
service_provider is "Libav" and the default for
service_name is "Service01".
avconv -i file.mpg -c copy \
-mpegts_original_network_id 0x1122 \
-mpegts_transport_stream_id 0x3344 \
-mpegts_service_id 0x5566 \
-mpegts_pmt_start_pid 0x1500 \
-mpegts_start_pid 0x150 \
-metadata service_provider="Some provider" \
-metadata service_name="Some Channel" \
-y out.ts
|
Null muxer.
This muxer does not generate any output file, it is mainly useful for testing or benchmarking purposes.
For example to benchmark decoding with avconv you can use the
command:
avconv -benchmark -i INPUT -f null out.null |
Note that the above command does not read or write the ‘out.null’
file, but specifying the output file is required by the avconv
syntax.
Alternatively you can write the command as:
avconv -benchmark -i INPUT -f null - |
Change the syncpoint usage in nut:
The none and timestamped flags are experimental.
avconv -i INPUT -f_strict experimental -syncpoints none - | processor |
Ogg container muxer.
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.
Serial value from which to set the streams serial number. Setting it to different and sufficiently large values ensures that the produced ogg files can be safely chained.
Basic stream segmenter.
The segmenter muxer outputs streams to a number of separate files of nearly fixed duration. Output filename pattern can be set in a fashion similar to image2.
Every segment starts with a video keyframe, if a video stream is present. The segment muxer works best with a single constant frame rate video.
Optionally it can generate a flat list of the created segments, one segment per line.
Override the inner container format, by default it is guessed by the filename extension.
Set segment duration to t seconds.
Generate also a listfile named name.
Select the listing format.
Overwrite the listfile once it reaches size entries.
Prepend prefix to each entry. Useful to generate absolute paths.
Wrap around segment index once it reaches limit.
Make sure to require a closed GOP when encoding and to set the GOP size to fit your segment time constraint.
avconv -i in.mkv -c hevc -flags +cgop -g 60 -map 0 -f segment -list out.list out%03d.nut |
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.
All protocols accept the following options:
Maximum time to wait for (network) read/write operations to complete, in microseconds.
A description of the currently available protocols follows.
Physical concatenation protocol.
Allow to read and seek from many resource in sequence as if they were a unique resource.
A URL accepted by this protocol has the syntax:
concat:URL1|URL2|...|URLN |
where URL1, URL2, ..., URLN are the urls of the resource to be concatenated, each one possibly specifying a distinct protocol.
For example to read a sequence of files ‘split1.mpeg’,
‘split2.mpeg’, ‘split3.mpeg’ with avplay use the
command:
avplay concat:split1.mpeg\|split2.mpeg\|split3.mpeg |
Note that you may need to escape the character "|" which is special for many shells.
File access protocol.
Allow to read from or read to a file.
For example to read from a file ‘input.mpeg’ with avconv
use the command:
avconv -i file:input.mpeg output.mpeg |
The av* tools default to the file protocol, that is a resource specified with the name "FILE.mpeg" is interpreted as the URL "file:FILE.mpeg".
This protocol accepts the following options:
If set to 1, the protocol will retry reading at the end of the file, allowing reading files that still are being written. In order for this to terminate, you either need to use the rw_timeout option, or use the interrupt callback (for API users).
Gopher protocol.
Read Apple HTTP Live Streaming compliant segmented stream as a uniform one. The M3U8 playlists describing the segments can be remote HTTP resources or local files, accessed using the standard file protocol. The nested protocol is declared by specifying "+proto" after the hls URI scheme name, where proto is either "file" or "http".
hls+http://host/path/to/remote/resource.m3u8 hls+file://path/to/local/resource.m3u8 |
Using this protocol is discouraged - the hls demuxer should work just as well (if not, please report the issues) and is more complete. To use the hls demuxer instead, simply use the direct URLs to the m3u8 files.
HTTP (Hyper Text Transfer Protocol).
This protocol accepts the following options:
If set to 1 use chunked Transfer-Encoding for posts, default is 1.
Set a specific content type for the POST messages.
Set custom HTTP headers, can override built in default headers. The value must be a string encoding the headers.
Use persistent connections if set to 1, default is 0.
Set custom HTTP post data.
Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.
Export the MIME type.
If set to 1 request ICY (SHOUTcast) metadata from the server. If the server supports this, the metadata has to be retrieved by the application by reading the ‘icy_metadata_headers’ and ‘icy_metadata_packet’ options. The default is 1.
If the server supports ICY metadata, this contains the ICY-specific HTTP reply headers, separated by newline characters.
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.
Set initial byte offset.
Try to limit the request to bytes preceding this offset.
Icecast (stream to Icecast servers)
This protocol accepts the following options:
Set the stream genre.
Set the stream name.
Set the stream description.
Set the stream website URL.
Set if the stream should be public or not. The default is 0 (not public).
Override the User-Agent header. If not specified a string of the form "Lavf/<version>" will be used.
Set the Icecast mountpoint password.
Set the stream content type. This must be set if it is different from audio/mpeg.
This enables support for Icecast versions < 2.4.0, that do not support the HTTP PUT method but the SOURCE method.
MMS (Microsoft Media Server) protocol over TCP.
MMS (Microsoft Media Server) protocol over HTTP.
The required syntax is:
mmsh://server[:port][/app][/playpath] |
MD5 output protocol.
Computes the MD5 hash of the data to be written, and on close writes this to the designated output or stdout if none is specified. It can be used to test muxers without writing an actual file.
Some examples follow.
# Write the MD5 hash of the encoded AVI file to the file output.avi.md5. 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.
UNIX pipe access protocol.
Allow to read and write from UNIX pipes.
The accepted syntax is:
pipe:[number] |
number is the number corresponding to the file descriptor of the pipe (e.g. 0 for stdin, 1 for stdout, 2 for stderr). If number is not specified, by default the stdout file descriptor will be used for writing, stdin for reading.
For example to read from stdin with avconv:
cat test.wav | avconv -i pipe:0 # ...this is the same as... cat test.wav | avconv -i pipe: |
For writing to stdout with avconv:
avconv -i test.wav -f avi pipe:1 | cat > test.avi # ...this is the same as... avconv -i test.wav -f avi pipe: | cat > test.avi |
Note that some formats (typically MOV), require the output protocol to be seekable, so they will fail with the pipe output protocol.
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:
An optional username (mostly for publishing).
An optional password (mostly for publishing).
The address of the RTMP server.
The number of the TCP port to use (by default is 1935).
It is the name of the application to access. It usually corresponds to
the path where the application is installed on the RTMP server
(e.g. ‘/ondemand/’, ‘/flash/live/’, etc.). You can override
the value parsed from the URI through the rtmp_app option, too.
It is the path or name of the resource to play with reference to the
application specified in app, may be prefixed by "mp4:". You
can override the value parsed from the URI through the rtmp_playpath
option, too.
Act as a server, listening for an incoming connection.
Maximum time to wait for the incoming connection. Implies listen.
Additionally, the following parameters can be set via command line options
(or in code via AVOptions):
Name of application to connect on the RTMP server. This option overrides the parameter specified in the URI.
Set the client buffer time in milliseconds. The default is 3000.
Extra arbitrary AMF connection parameters, parsed from a string,
e.g. like B:1 S:authMe O:1 NN:code:1.23 NS:flag:ok O:0.
Each value is prefixed by a single character denoting the type,
B for Boolean, N for number, S for string, O for object, or Z for null,
followed by a colon. For Booleans the data must be either 0 or 1 for
FALSE or TRUE, respectively. Likewise for Objects the data must be 0 or
1 to end or begin an object, respectively. Data items in subobjects may
be named, by prefixing the type with ’N’ and specifying the name before
the value (i.e. NB:myFlag:1). This option may be used multiple
times to construct arbitrary AMF sequences.
Version of the Flash plugin used to run the SWF player. The default is LNX 9,0,124,2. (When publishing, the default is FMLE/3.0 (compatible; <libavformat version>).)
Number of packets flushed in the same request (RTMPT only). The default is 10.
Specify that the media is a live stream. No resuming or seeking in
live streams is possible. The default value is any, which means the
subscriber first tries to play the live stream specified in the
playpath. If a live stream of that name is not found, it plays the
recorded stream. The other possible values are live and
recorded.
URL of the web page in which the media was embedded. By default no value will be sent.
Stream identifier to play or to publish. This option overrides the parameter specified in the URI.
Name of live stream to subscribe to. By default no value will be sent. It is only sent if the option is specified or if rtmp_live is set to live.
SHA256 hash of the decompressed SWF file (32 bytes).
Size of the decompressed SWF file, required for SWFVerification.
URL of the SWF player for the media. By default no value will be sent.
URL to player swf file, compute hash/size automatically.
URL of the target stream. Defaults to proto://host[:port]/app.
For example to read with avplay a multimedia resource named
"sample" from the application "vod" from an RTMP server "myserver":
avplay rtmp://myserver/vod/sample |
To publish to a password protected server, passing the playpath and app names separately:
avconv -re -i <input> -f flv -rtmp_playpath some/long/path -rtmp_app long/app/name rtmp://username:password@myserver/ |
Encrypted Real-Time Messaging Protocol.
The Encrypted Real-Time Messaging Protocol (RTMPE) is used for streaming multimedia content within standard cryptographic primitives, consisting of Diffie-Hellman key exchange and HMACSHA256, generating a pair of RC4 keys.
Real-Time Messaging Protocol over a secure SSL connection.
The Real-Time Messaging Protocol (RTMPS) is used for streaming multimedia content across an encrypted connection.
Real-Time Messaging Protocol tunneled through HTTP.
The Real-Time Messaging Protocol tunneled through HTTP (RTMPT) is used for streaming multimedia content within HTTP requests to traverse firewalls.
Encrypted Real-Time Messaging Protocol tunneled through HTTP.
The Encrypted Real-Time Messaging Protocol tunneled through HTTP (RTMPTE) is used for streaming multimedia content within HTTP requests to traverse firewalls.
Real-Time Messaging Protocol tunneled through HTTPS.
The Real-Time Messaging Protocol tunneled through HTTPS (RTMPTS) is used for streaming multimedia content within HTTPS requests to traverse firewalls.
Real-Time Messaging Protocol and its variants supported through librtmp.
Requires the presence of the librtmp headers and library during configuration. You need to explicitly configure the build with "–enable-librtmp". If enabled this will replace the native RTMP protocol.
This protocol provides most client functions and a few server functions needed to support RTMP, RTMP tunneled in HTTP (RTMPT), encrypted RTMP (RTMPE), RTMP over SSL/TLS (RTMPS) and tunneled variants of these encrypted types (RTMPTE, RTMPTS).
The required syntax is:
rtmp_proto://server[:port][/app][/playpath] options |
where rtmp_proto is one of the strings "rtmp", "rtmpt", "rtmpe", "rtmps", "rtmpte", "rtmpts" corresponding to each RTMP variant, and server, port, app and playpath have the same meaning as specified for the RTMP native protocol. options contains a list of space-separated options of the form key=val.
See the librtmp manual page (man 3 librtmp) for more information.
For example, to stream a file in real-time to an RTMP server using
avconv:
avconv -re -i myfile -f flv rtmp://myserver/live/mystream |
To play the same stream using avplay:
avplay "rtmp://myserver/live/mystream live=1" |
Real-Time Protocol.
RTSP is not technically a protocol handler in libavformat, it is a demuxer and muxer. The demuxer supports both normal RTSP (with data transferred over RTP; this is used by e.g. Apple and Microsoft) and Real-RTSP (with data transferred over RDT).
The muxer can be used to send a stream using RTSP ANNOUNCE to a server supporting it (currently Darwin Streaming Server and Mischa Spiegelmock’s RTSP server).
The required syntax for a RTSP url is:
rtsp://hostname[:port]/path |
The following options (set on the avconv/avplay command
line, or set in code via AVOptions or in avformat_open_input),
are supported:
Flags for rtsp_transport:
Use UDP as lower transport protocol.
Use TCP (interleaving within the RTSP control channel) as lower transport protocol.
Use UDP multicast as lower transport protocol.
Use HTTP tunneling as lower transport protocol, which is useful for passing proxies.
Multiple lower transport protocols may be specified, in that case they are
tried one at a time (if the setup of one fails, the next one is tried).
For the muxer, only the tcp and udp options are supported.
Flags for rtsp_flags:
Accept packets only from negotiated peer address and port.
Act as a server, listening for an incoming connection.
When receiving data over UDP, the demuxer tries to reorder received packets
(since they may arrive out of order, or packets may get lost totally). This
can be disabled by setting the maximum demuxing delay to zero (via
the max_delay field of AVFormatContext).
When watching multi-bitrate Real-RTSP streams with avplay, the
streams to display can be chosen with -vst n and
-ast n for video and audio respectively, and can be switched
on the fly by pressing v and a.
Example command lines:
To watch a stream over UDP, with a max reordering delay of 0.5 seconds:
avplay -max_delay 500000 -rtsp_transport udp rtsp://server/video.mp4 |
To watch a stream tunneled over HTTP:
avplay -rtsp_transport http rtsp://server/video.mp4 |
To send a stream in realtime to a RTSP server, for others to watch:
avconv -re -i input -f rtsp -muxdelay 0.1 rtsp://server/live.sdp |
To receive a stream in realtime:
avconv -rtsp_flags listen -i rtsp://ownaddress/live.sdp output |
Session Announcement Protocol (RFC 2974). This is not technically a protocol handler in libavformat, it is a muxer and demuxer. It is used for signalling of RTP streams, by announcing the SDP for the streams regularly on a separate port.
The syntax for a SAP url given to the muxer is:
sap://destination[:port][?options] |
The RTP packets are sent to destination on port port,
or to port 5004 if no port is specified.
options is a &-separated list. The following options
are supported:
Specify the destination IP address for sending the announcements to. If omitted, the announcements are sent to the commonly used SAP announcement multicast address 224.2.127.254 (sap.mcast.net), or ff0e::2:7ffe if destination is an IPv6 address.
Specify the port to send the announcements on, defaults to 9875 if not specified.
Specify the time to live value for the announcements and RTP packets, defaults to 255.
If set to 1, send all RTP streams on the same port pair. If zero (the default), all streams are sent on unique ports, with each stream on a port 2 numbers higher than the previous. VLC/Live555 requires this to be set to 1, to be able to receive the stream. The RTP stack in libavformat for receiving requires all streams to be sent on unique ports.
Example command lines follow.
To broadcast a stream on the local subnet, for watching in VLC:
avconv -re -i input -f sap sap://224.0.0.255?same_port=1 |
Similarly, for watching in avplay:
avconv -re -i input -f sap sap://224.0.0.255 |
And for watching in avplay, over IPv6:
avconv -re -i input -f sap sap://[ff0e::1:2:3:4] |
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] |
Haivision Secure Reliable Transport Protocol via libsrt.
The supported syntax for a SRT URL is:
srt://hostname:port[?options] |
options contains a list of &-separated options of the form key=val.
or
options srt://hostname:port |
options contains a list of ’-key val’ options.
This protocol accepts the following options.
Connection timeout; SRT cannot connect for RTT > 1500 msec (2 handshake exchanges) with the default connect timeout of 3 seconds. This option applies to the caller and rendezvous connection modes. The connect timeout is 10 times the value set for the rendezvous mode (which can be used as a workaround for this connection problem with earlier versions).
Flight Flag Size (Window Size), in bytes. FFS is actually an internal parameter and you should set it to not less than ‘recv_buffer_size’ and ‘mss’. The default value is relatively large, therefore unless you set a very large receiver buffer, you do not need to change this option. Default value is 25600.
Sender nominal input rate, in bytes per seconds. Used along with ‘oheadbw’, when ‘maxbw’ is set to relative (0), to calculate maximum sending rate when recovery packets are sent along with the main media stream: ‘inputbw’ * (100 + ‘oheadbw’) / 100 if ‘inputbw’ is not set while ‘maxbw’ is set to relative (0), the actual input rate is evaluated inside the library. Default value is 0.
IP Type of Service. Applies to sender only. Default value is 0xB8.
IP Time To Live. Applies to sender only. Default value is 64.
Timestamp-based Packet Delivery Delay. Used to absorb bursts of missed packet retransmissions. This flag sets both ‘rcvlatency’ and ‘peerlatency’ to the same value. Note that prior to version 1.3.0 this is the only flag to set the latency, however this is effectively equivalent to setting ‘peerlatency’, when side is sender and ‘rcvlatency’ when side is receiver, and the bidirectional stream sending is not supported.
Set socket listen timeout.
Maximum sending bandwidth, in bytes per seconds. -1 infinite (CSRTCC limit is 30mbps) 0 relative to input rate (see ‘inputbw’) >0 absolute limit value Default value is 0 (relative)
Connection mode. ‘caller’ opens client connection. ‘listener’ starts server to listen for incoming connections. ‘rendezvous’ use Rendez-Vous connection mode. Default value is caller.
Maximum Segment Size, in bytes. Used for buffer allocation and rate calculation using a packet counter assuming fully filled packets. The smallest MSS between the peers is used. This is 1500 by default in the overall internet. This is the maximum size of the UDP packet and can be only decreased, unless you have some unusual dedicated network settings. Default value is 1500.
If set to 1, Receiver will send ‘UMSG_LOSSREPORT‘ messages periodically until a lost packet is retransmitted or intentionally dropped. Default value is 1.
Recovery bandwidth overhead above input rate, in percents. See ‘inputbw’. Default value is 25%.
HaiCrypt Encryption/Decryption Passphrase string, length from 10 to 79 characters. The passphrase is the shared secret between the sender and the receiver. It is used to generate the Key Encrypting Key using PBKDF2 (Password-Based Key Derivation Function). It is used only if ‘pbkeylen’ is non-zero. It is used on the receiver only if the received data is encrypted. The configured passphrase cannot be recovered (write-only).
Sets the maximum declared size of a packet transferred during the single call to the sending function in Live mode. Use 0 if this value isn’t used (which is default in file mode). Default value is for MPEG-TS; if you are going to use SRT to send any different kind of payload, such as, for example, wrapping a live stream in very small frames, then you can use a bigger maximum frame size, though not greater than 1456 bytes.
The latency value (as described in ‘rcvlatency’) that is set by the sender side as a minimum value for the receiver.
Sender encryption key length, in bytes. Only can be set to 0, 16, 24 and 32. Enable sender encryption if not 0. Not required on receiver (set to 0), key size obtained from sender in HaiCrypt handshake. Default value is 0.
The time that should elapse since the moment when the packet was sent and the moment when it’s delivered to the receiver application in the receiving function. This time should be a buffer time large enough to cover the time spent for sending, unexpectedly extended RTT time, and the time needed to retransmit the lost UDP packet. The effective latency value will be the maximum of this options’ value and the value of ‘peerlatency’ set by the peer side. Before version 1.3.0 this option is only available as ‘latency’.
Set receive buffer size, expressed in bytes.
Set send buffer size, expressed in bytes.
Set raise error timeout for read/write optations.
This option is only relevant in read mode: if no data arrived in more than this time interval, raise error.
Too-late Packet Drop. When enabled on receiver, it skips missing packets that have not been delivered in time and delivers the following packets to the application when their time-to-play has come. It also sends a fake ACK to the sender. When enabled on sender and enabled on the receiving peer, the sender drops the older packets that have no chance of being delivered in time. It was automatically enabled in the sender if the receiver supports it.
For more information see: https://github.com/Haivision/srt.
Transmission Control Protocol.
The required syntax for a TCP url is:
tcp://hostname:port[?options] |
Listen for an incoming connection
avconv -i input -f format tcp://hostname:port?listen avplay tcp://hostname:port |
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 AVOptions):
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.
If enabled, try to verify the peer that we are communicating with. Note, if using OpenSSL, this currently only makes sure that the peer certificate is signed by one of the root certificates in the CA database, but it does not validate that the certificate actually matches the host name we are trying to connect to. (With GnuTLS, the host name is validated as well.)
This is disabled by default since it requires a CA database to be provided by the caller in many cases.
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.)
A file containing the private key for the certificate.
If enabled, listen for connections on the provided port, and assume the server role in the handshake instead of the client role.
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.
set the UDP buffer size in bytes
override the local UDP port to bind with
Choose the local IP address. This is useful e.g. if sending multicast and the host has multiple interfaces, where the user can choose which interface to send on by specifying the IP address of that interface.
set the size in bytes of UDP packets
explicitly allow or disallow reusing UDP sockets
set the time to live value (for multicast only)
Initialize the UDP socket with connect(). In this case, the
destination address can’t be changed with ff_udp_set_remote_url later.
If the destination address isn’t known at the start, this option can
be specified in ff_udp_set_remote_url, too.
This allows finding out the source address for the packets with getsockname,
and makes writes return with AVERROR(ECONNREFUSED) if "destination
unreachable" is received.
For receiving, this gives the benefit of only receiving packets from
the specified peer address/port.
Only receive packets sent to the multicast group from one of the specified sender IP addresses.
Ignore packets sent to the multicast group from the specified sender IP addresses.
Some usage examples of the udp protocol with avconv follow.
To stream over UDP to a remote endpoint:
avconv -i input -f format udp://hostname:port |
To stream in mpegts format over UDP using 188 sized UDP packets, using a large input buffer:
avconv -i input -f mpegts udp://hostname:port?pkt_size=188&buffer_size=65535 |
To receive over UDP from a remote endpoint:
avconv -i udp://[multicast-address]:port |
Unix 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 AVOptions):
Timeout in ms.
Create the Unix socket in listening mode.
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 (Advanced Linux Sound Architecture) input device.
To enable this input device during configuration you need libasound installed on your system.
This device allows capturing from an ALSA device. The name of the device to capture has to be an ALSA card identifier.
An ALSA identifier has the syntax:
hw:CARD[,DEV[,SUBDEV]] |
where the DEV and SUBDEV components are optional.
The three arguments (in order: CARD,DEV,SUBDEV) specify card number or identifier, device number and subdevice number (-1 means any).
To see the list of cards currently recognized by your system check the files ‘/proc/asound/cards’ and ‘/proc/asound/devices’.
For example to capture with avconv from an ALSA device with
card id 0, you may run the command:
avconv -f alsa -i hw:0 alsaout.wav |
For more information see: http://www.alsa-project.org/alsa-doc/alsa-lib/pcm.html
BSD video input device.
Linux framebuffer input device.
The Linux framebuffer is a graphic hardware-independent abstraction layer to show graphics on a computer monitor, typically on the console. It is accessed through a file device node, usually ‘/dev/fb0’.
For more detailed information read the file Documentation/fb/framebuffer.txt included in the Linux source tree.
To record from the framebuffer device ‘/dev/fb0’ with
avconv:
avconv -f fbdev -r 10 -i /dev/fb0 out.avi |
You can take a single screenshot image with the command:
avconv -f fbdev -frames:v 1 -r 1 -i /dev/fb0 screenshot.jpeg |
See also http://linux-fbdev.sourceforge.net/, and fbset(1).
JACK input device.
To enable this input device during configuration you need libjack installed on your system.
A JACK input device creates one or more JACK writable clients, one for each audio channel, with name client_name:input_N, where client_name is the name provided by the application, and N is a number which identifies the channel. Each writable client will send the acquired data to the Libav input device.
Once you have created one or more JACK readable clients, you need to connect them to one or more JACK writable clients.
To connect or disconnect JACK clients you can use the ‘jack_connect’ and ‘jack_disconnect’ programs, or do it through a graphical interface, for example with ‘qjackctl’.
To list the JACK clients and their properties you can invoke the command ‘jack_lsp’.
Follows an example which shows how to capture a JACK readable client
with avconv.
# Create a JACK writable client with name "libav". $ avconv -f jack -i libav -y out.wav # Start the sample jack_metro readable client. $ jack_metro -b 120 -d 0.2 -f 4000 # List the current JACK clients. $ jack_lsp -c system:capture_1 system:capture_2 system:playback_1 system:playback_2 libav:input_1 metro:120_bpm # Connect metro to the avconv writable client. $ jack_connect metro:120_bpm libav:input_1 |
For more information read: http://jackaudio.org/
IIDC1394 input device, based on libdc1394 and libraw1394.
Open Sound System input device.
The filename to provide to the input device is the device node representing the OSS input device, and is usually set to ‘/dev/dsp’.
For example to grab from ‘/dev/dsp’ using avconv use the
command:
avconv -f oss -i /dev/dsp /tmp/oss.wav |
For more information about OSS see: http://manuals.opensound.com/usersguide/dsp.html
pulseaudio input device.
To enable this input device during configuration you need libpulse-simple installed in your system.
The filename to provide to the input device is a source device or the string "default"
To list the pulse source devices and their properties you can invoke the command ‘pactl list sources’.
avconv -f pulse -i default /tmp/pulse.wav |
The syntax is:
-server server name |
Connects to a specific server.
The syntax is:
-name application name |
Specify the application name pulse will use when showing active clients, by default it is "libav"
The syntax is:
-stream_name stream name |
Specify the stream name pulse will use when showing active streams, by default it is "record"
The syntax is:
-sample_rate samplerate |
Specify the samplerate in Hz, by default 48kHz is used.
The syntax is:
-channels N |
Specify the channels in use, by default 2 (stereo) is set.
The syntax is:
-frame_size bytes |
Specify the number of byte per frame, by default it is set to 1024.
The syntax is:
-fragment_size bytes |
Specify the minimal buffering fragment in pulseaudio, it will affect the audio latency. By default it is unset.
sndio input device.
To enable this input device during configuration you need libsndio installed on your system.
The filename to provide to the input device is the device node representing the sndio input device, and is usually set to ‘/dev/audio0’.
For example to grab from ‘/dev/audio0’ using avconv use the
command:
avconv -f sndio -i /dev/audio0 /tmp/oss.wav |
Video4Linux2 input video device.
The name of the device to grab is a file device node, usually Linux systems tend to automatically create such nodes when the device (e.g. an USB webcam) is plugged into the system, and has a name of the kind ‘/dev/videoN’, where N is a number associated to the device.
Video4Linux2 devices usually support a limited set of
widthxheight sizes and framerates. You can check which are
supported using -list_formats all for Video4Linux2 devices.
Some usage examples of the video4linux2 devices with avconv and avplay:
# List supported formats for a video4linux2 device. avplay -f video4linux2 -list_formats all /dev/video0 # 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 |
VfW (Video for Windows) capture input device.
The filename passed as input is the capture driver number, ranging from 0 to 9. You may use "list" as filename to print a list of drivers. Any other filename will be interpreted as device number 0.
X11 video input device.
This device allows to capture a region of an X11 display.
The filename passed as input has the syntax:
[hostname]:display_number.screen_number[+x_offset,y_offset] |
hostname:display_number.screen_number specifies the
X11 display name of the screen to grab from. hostname can be
omitted, and defaults to "localhost". The environment variable
DISPLAY contains the default display name.
x_offset and y_offset specify the offsets of the grabbed area with respect to the top-left border of the X11 screen. They default to 0.
Check the X11 documentation (e.g. man X) for more detailed information.
Use the ‘dpyinfo’ program for getting basic information about the properties of your X11 display (e.g. grep for "name" or "dimensions").
For example to grab from ‘:0.0’ using avconv:
avconv -f x11grab -r 25 -s cif -i :0.0 out.mpg # Grab at position 10,20. avconv -f x11grab -r 25 -s cif -i :0.0+10,20 out.mpg |
The syntax is:
-follow_mouse centered|PIXELS |
When it is specified with "centered", the grabbing region follows the mouse pointer and keeps the pointer at the center of region; otherwise, the region follows only when the mouse pointer reaches within PIXELS (greater than zero) to the edge of region.
For example:
avconv -f x11grab -follow_mouse centered -r 25 -s cif -i :0.0 out.mpg # Follows only when the mouse pointer reaches within 100 pixels to edge avconv -f x11grab -follow_mouse 100 -r 25 -s cif -i :0.0 out.mpg |
The syntax is:
-show_region 1 |
If show_region AVOption is specified with 1, then the grabbing region will be indicated on screen. With this option, it’s easy to know what is being grabbed if only a portion of the screen is grabbed.
For example:
avconv -f x11grab -show_region 1 -r 25 -s cif -i :0.0+10,20 out.mpg # With follow_mouse avconv -f x11grab -follow_mouse centered -show_region 1 -r 25 -s cif -i :0.0 out.mpg |
The syntax is:
-grab_x x_offset -grab_y y_offset |
Set the grabbing region coordinates. The are expressed as offset from the top left corner of the X11 window. The default value is 0.